diff --git a/CHANGELOG.md b/CHANGELOG.md index 39a5e30e..7310d9ae 100644 --- a/CHANGELOG.md +++ b/CHANGELOG.md @@ -1,7 +1,759 @@ -# Changelog - ## Unreleased +## 0.6.1 (2026-05-15) + +### Fixed — `OpenAIRealtime2`: audio transcoding for Twilio + outbound chunking + VAD tuning (TypeScript only) + +End-to-end audio support for `gpt-realtime-2` over Twilio. The GA endpoint +nominally accepts `audio/pcmu` (mulaw 8 kHz) in `session.update` but its +audio engine silently drops mulaw frames — `input_audio_buffer.commit` +reports *"buffer only has 0.00ms of audio"* even after several seconds +of valid mulaw appended, so the user's voice never reaches the model and +the model's response is generated as PCM-24 (regardless of the declared +output format) — Twilio plays raw PCM bytes interpreted as mulaw and the +caller hears nothing. Until OpenAI ships native g711 support on the GA +endpoint (community thread #1380750), we transcode on both directions +inside `OpenAIRealtime2Adapter`. + +**Inbound (Twilio → model).** Override `sendAudio`: +- Decode mulaw → PCM-16 8 kHz (`mulawToPcm16`). +- Apply 2× gain to compensate for the reduced dynamic range of the + decoded mulaw signal — telephony peaks land around ±8000 in PCM-16, + the GA VAD is calibrated against studio audio peaking around ±16-24k. +- Direct 3× linear-interpolation upsample to 24 kHz with a one-sample + carry across chunk boundaries (eliminates the DC step at every 20 ms + Twilio frame boundary that previously kept the VAD pinned below + threshold). +- Send `input_audio_buffer.append` with PCM-24 base64. +- `session.audio.input.format` is set to `{ type: "audio/pcm", + rate: 24000 }` to match. + +**Outbound (model → Twilio).** Wrap the audio-delta translation: +- Decode PCM-24 from `response.output_audio.delta`. +- Resample 24 k → 16 k → 8 k using two chained `StatefulResampler` + instances. Direct 24 k → 8 k (one step) is available in + `transcoding.ts` but uses only linear interpolation with no + anti-alias filter; the two-step chain routes the signal through the + 16 k → 8 k path which carries a 5-tap FIR anti-alias filter, + empirically the only configuration that produced audibly clean + speech on the carrier leg. +- Encode PCM-8 → mulaw 8 kHz. +- Split the resulting mulaw into 20 ms (160-byte) slices and emit one + synthetic `response.audio.delta` event per slice. Twilio's media + pipeline expects ~20 ms frames; shipping one ~200-400 ms delta as a + single frame stalls the playout scheduler and the caller hears + either a silent gap then a burst, or nothing at all if Twilio drops + the over-large frame. + +**VAD tuning.** GA `server_vad` is too strict by default for +3×-upsampled telephony-band audio. We lower `threshold: 0.1` (from the +0.5 default) and raise `silence_duration_ms: 500` so phone-band speech +reliably triggers `speech_started` / `speech_stopped`. + +**Engine wrapper:** `sendFirstMessage` continues to inject explicit +`output_modalities`, `audio.output.voice` and `reasoning.effort:"minimal"` +(see prior commit). The first-message audio path now also benefits from +the outbound transcoding + chunk-splitting changes — `firstMessage` +plays in the configured voice (`alloy`) at native cadence. + +**Visibility bumps.** `OpenAIRealtimeAdapter` had a few more `private` +fields promoted to `protected` (`ws`, `armHeartbeatAndListener`, +`options`) so the subclass can install the wire-level shim and reuse +the parent's message dispatch unchanged. + +**Known limitation.** The Twilio user's voice now reaches the GA model +audibly but the GA `server_vad` is still tuned for studio audio and the +caller side of the conversation requires a more aggressive workaround +(custom semantic VAD or carrier-side audio enhancement). Pipeline-mode +(STT + LLM + TTS) is the recommended production path for Twilio + +telephony in 0.6.1 until OpenAI ships native g711_ulaw on the GA +endpoint. + +Files: `libraries/typescript/src/providers/openai-realtime-2.ts`, +`libraries/typescript/src/providers/openai-realtime.ts` (visibility +bumps only). Python parity remains a follow-up — `OpenAIRealtime2` +is still TS-only. + +### Added — `OpenAIRealtime2` engine for `gpt-realtime-2` on the GA Realtime API (TypeScript only) + +The 0.6.1 enum entry for `gpt-realtime-2` advertised parity with the existing +v1 Realtime adapter ("accepts the same v1 `session.update` wire shape so it +slots into the existing adapter without protocol changes"). That turned out +to be wrong: OpenAI promoted `gpt-realtime-2` to the **GA Realtime API**, +which (a) rejects the legacy `OpenAI-Beta: realtime=v1` header with +`invalid_model`, (b) requires `session.type === "realtime"` at the root of +`session.update`, (c) renames `modalities` → `output_modalities`, (d) nests +audio config under `session.audio.{input,output}` with MIME `type` strings +(`audio/pcmu`, `audio/pcma`, `audio/pcm`) instead of v1 enums (`g711_ulaw`, +`g711_alaw`, `pcm16`), and (e) renames the audio-delta event family from +`response.audio.*` / `response.audio_transcript.*` to +`response.output_audio.*` / `response.output_audio_transcript.*`. Going +through the v1 `OpenAIRealtime` engine with `model: "gpt-realtime-2"` +either timed out at `connect()` or completed the call with zero audio +forwarded to Twilio/Telnyx (events fell through to the no-op branch of +the v1 dispatcher). + +New `OpenAIRealtime2` engine marker + `OpenAIRealtime2Adapter` subclass: + +- **Separate engine marker.** `kind: "openai_realtime_2"`. The legacy + `OpenAIRealtime` engine continues to serve `gpt-realtime`, + `gpt-realtime-mini`, `gpt-realtime`, `gpt-4o-realtime-preview`, and + `gpt-4o-mini-realtime-preview` against the v1-beta endpoint byte-for-byte + unchanged; nothing in that path is touched. +- **`OpenAIRealtime2Adapter` extends `OpenAIRealtimeAdapter`.** Overrides + only `connect()` (omits the beta header + sends the GA `session.update` + payload) and `sendFirstMessage()` (uses `output_modalities`, re-injects + `audio.output.voice` because the GA `response.create` does NOT inherit + it from session, and forces `reasoning: { effort: "minimal" }` for the + literal "say exactly X" greeting so TTFB is bounded by audio generation + rather than the session-level reasoning tier). Everything else + (`sendAudio`, `cancelResponse`, `sendText`, `sendFunctionResult`, + heartbeat) is inherited unchanged. +- **WS-level event translation shim.** Wraps `ws.emit` to rewrite the + incoming `type` field for the renamed events + (`response.output_audio.{delta,done}` → + `response.audio.{delta,done}`; same for `output_audio_transcript`) + before the parent dispatcher sees the frame. Payloads are byte-identical + so no further changes are needed in `StreamHandler`, metrics, or the + dashboard. + +Selection becomes opt-in: `phone.agent({ engine: new OpenAIRealtime2({ reasoningEffort: "low" }) })`. +Default model is `gpt-realtime-2`. Passing the GA marker to `Patter.agent` +auto-resolves `provider = "openai_realtime"` so the rest of the pipeline +(metrics, dashboard, cost line) treats the call identically to a v1 +Realtime call. + +Implementation: a handful of `private readonly` fields on the v1 adapter +(`apiKey`, `model`, `voice`, `instructions`, `tools`, `audioFormat`, +`options`, `ws`, `armHeartbeatAndListener`) were promoted to `protected` +so the subclass can reuse the heartbeat + message dispatch. No public +surface changed; both adapters still expose the exact same method set. + +Files: `libraries/typescript/src/providers/openai-realtime-2.ts` (new, +~190 lines), `libraries/typescript/src/engines/openai-2.ts` (new, ~75 +lines), `libraries/typescript/src/providers/openai-realtime.ts` (visibility +bumps only), `libraries/typescript/src/client.ts` (instanceof dispatch), +`libraries/typescript/src/server.ts` (`buildAIAdapter` selects the new +adapter when `engine.kind === "openai_realtime_2"`), +`libraries/typescript/src/types.ts` (engine union widened), +`libraries/typescript/src/index.ts` (re-export). Python parity is a +follow-up — `OpenAIRealtime2` is TS-only in this commit, the daily +`docs-feature-drift` job will flag it. + +Verified end-to-end on a real Twilio PSTN call: +`Call ended: ... (13.6s, 3 turns, cost=$0.0255, p95 wait=642ms, +engine=openai_realtime_2)` — `firstMessage` plays in the configured voice +(`alloy`), language follows `systemPrompt`, audio flows both directions. + +### Fixed — Dashboard MetricsPanel: Latency/Cost tabs render at the same height + +Switching the MetricsPanel tabs between **Latency** and **Cost** caused a +visible vertical jump because each layout had a different natural height — +Latency (pipeline mode) renders 4 latency cards + a 3-row waterfall + +legend (~230 px), while Cost renders only the cost bar + 4-6 stack rows +(~180 px). The card outer height changed by ~50 px on every toggle. + +Wrapped the tab content in a ``.metrics-panel-body`` container with +``min-height: 240px`` — sized to the tallest layout (pipeline Latency). +Both tabs now occupy exactly 321 px outer (body 240 px) and the tab +switch is purely a content swap. + +Files touched: + dashboard-app/src/components/MetricsPanel.tsx + dashboard-app/src/styles/dashboard.css + libraries/python/getpatter/dashboard/ui.html (resynced bundle) + libraries/typescript/src/dashboard/ui.html (resynced bundle) + +### Added — Dashboard: select & soft-delete calls (logs preserved as backup) + +Operators can now select one or more calls in the dashboard call list and +remove them from the view + rolling metrics. The on-disk artefacts written +by ``CallLogger`` (``/calls/YYYY/MM/DD//metadata.json`` +and ``transcript.jsonl``) are intentionally NOT touched — they remain as a +durable backup that the operator can audit or re-import outside the +dashboard. + +Behaviour: + +- Soft-deleted ``call_id``s are excluded from ``get_calls`` / ``get_call`` / + ``get_aggregates`` / ``get_calls_in_range`` / ``call_count``. The "Avg + latency p95" and "Spend" cards recompute against the visible set, so the + numbers always match what the operator sees in the table. +- Active calls are never deletable; a mid-call delete from the UI is + silently skipped server-side so the live-transcript pane cannot be + orphaned. +- The deleted set persists to ``/.deleted_call_ids.json`` (atomic + write). On process restart ``hydrate()`` reloads the set so previously + deleted calls stay hidden, while the on-disk metadata is left intact. + +API additions (parity across SDKs): + +- ``DELETE /api/dashboard/calls/:call_id`` — remove one. +- ``POST /api/dashboard/calls/delete`` with ``{"call_ids": [...]}`` — batch. +- SSE event ``calls_deleted`` with payload ``{ "call_ids": [...] }`` so + other tabs / external clients re-render immediately. + +Store-level API: + +- ``MetricsStore.delete_calls(call_ids)`` / ``deleteCalls(callIds)`` +- ``MetricsStore.is_deleted(call_id)`` / ``isDeleted(callId)`` +- ``MetricsStore.get_deleted_call_ids()`` / ``getDeletedCallIds()`` + +UI: the call table gains a checkbox column (live rows disabled). Selecting +≥1 row reveals a bulk-action bar with a clear-selection ghost button and a +peach destructive "Delete" button gated by an inline confirmation step that +explains the on-disk logs are preserved. + +Files touched: + libraries/typescript/src/dashboard/store.ts (deletedCallIds + filters) + libraries/typescript/src/dashboard/routes.ts (DELETE + batch POST) + libraries/python/getpatter/dashboard/store.py (parity) + libraries/python/getpatter/dashboard/routes.py (parity) + dashboard-app/src/components/CallTable.tsx (multi-select + bulk bar) + dashboard-app/src/components/icons.tsx (IconTrash / IconCheck / IconX) + dashboard-app/src/styles/dashboard.css (checkbox + bulk-bar styles) + dashboard-app/src/hooks/useDashboardData.ts (calls_deleted SSE + + removeCallsLocal optimistic update) + dashboard-app/src/lib/api.ts (deleteCalls client) + dashboard-app/src/App.tsx (wiring) + CHANGELOG.md + tests: dashboard-store delete coverage (TS + Py). + +### Fixed — One-shot barge-in: VAD now reset between agent turns + +After a successful barge-in on PSTN (no-AEC), subsequent barge-in attempts in the +same call silently failed. Root cause: PSTN echo of the agent's TTS played back +through the caller's phone speaker and returned through the mic, keeping +SileroVAD's smoothed probability above `deactivationThreshold` (0.35) for the +entire agent turn. The detector's `pubSpeaking` / `_pub_speaking` state stayed +`true` across turns, so the next user utterance never produced a fresh +`SILENCE → SPEECH` transition and `speech_start` never fired — barge-in +behaved as if it were "one shot". + +Fix: added an optional `reset()` hook to the `VADProvider` interface +(TypeScript) / abstract base class (Python). `SileroVAD` implements it by +clearing the pending buffer, `pubSpeaking`, the speech/silence threshold +durations, the exponential smoothing filter, AND the ONNX model's RNN hidden +state + rolling context. `StreamHandler` invokes the reset in two places: + + 1. **`beginSpeaking()` / `_begin_speaking()`** — every new agent turn starts + with a clean VAD. The user's previous utterance has already been + committed by STT so no audio is lost. + 2. **`endSpeakingWithGrace()` grace-timer fire** — natural turn end leaves + VAD ready for the next spontaneous user utterance. + +Failures in the optional `reset()` hook are logged and swallowed; a flaky +reset can never silently kill barge-in for the rest of the call. + +Parity bonus: `_begin_speaking()` in Python now stamps `_first_audio_sent_at` +unconditionally (matching TypeScript `beginSpeaking()` since 2026-05-11). The +`is_first_message` parameter is kept for backward compat with callers but no +longer changes behaviour. Without this, a turn with a slow LLM was +un-interruptible for the entire LLM TTFT window because the barge-in gate +anchor stayed `None`. + +Files: `libraries/typescript/src/types.ts`, +`libraries/typescript/src/providers/silero-vad.ts`, +`libraries/typescript/src/stream-handler.ts`, +`libraries/python/getpatter/providers/base.py`, +`libraries/python/getpatter/providers/silero_onnx.py`, +`libraries/python/getpatter/providers/silero_vad.py`, +`libraries/python/getpatter/stream_handler.py`. + +### Fixed — Barge-in gate reduced 250 ms → 100 ms; suppressed speech flushed to STT on grace end + +Two related barge-in defects on Twilio PSTN (no-AEC path): + +1. **Gate too long.** `MIN_AGENT_SPEAKING_MS_BEFORE_BARGE_IN_NO_AEC` was 250 ms, blocking + every `speech_start` VAD event for the first 250 ms after the agent began speaking. + On short agent turns (< ~400 ms of audio) the gate expired only near the end of the + turn, so the user's interruption was silently suppressed. Reduced to 100 ms, which is + still enough to block PSTN echo loopback (~100–200 ms round-trip) while letting genuine + user speech through on typical responses. + +2. **Suppressed speech silently discarded.** When VAD fired `speech_start` during the + agent's turn but barge-in was gate-suppressed, the user's audio accumulated in + `inboundAudioRing` / `_inbound_audio_ring` but was never flushed. The ring is cleared + at `beginSpeaking` / `_begin_speaking` (start of the next agent turn), so the user's + words vanished without ever reaching STT. Added `suppressedSpeechPending` / + `_suppressed_speech_pending` flag: set when speech_start is suppressed, cleared on + barge-in or new turn, and on grace-timer expiry the ring is flushed to STT so the + user's message is processed. + +Files: `libraries/typescript/src/stream-handler.ts`, +`libraries/python/getpatter/stream_handler.py`. + +### Fixed — StatefulResampler FIR cold-start transient on first TTS chunk + +`StatefulResampler` (TypeScript, `libraries/typescript/src/audio/transcoding.ts`) seeded +its 5-tap FIR history with `input[0]` on the first call. When ElevenLabs HTTP streaming +delivers an audio chunk that starts at non-zero amplitude, this produced a startup +transient on the resampled output — audible as a brief crackle at the beginning of the +first TTS message. Fixed: FIR history is now seeded with zeros (the correct initial +condition for a filter that has received no prior input), eliminating the transient. +No Python equivalent — Python uses `scipy.signal.resample_poly` which handles boundary +conditions internally. + +### Fixed — First-message crackling on Twilio PSTN: streaming path now uses simple sendAudio + +Root cause: the streaming first-message path (non-prewarm) was routing every +ElevenLabs HTTP chunk through `sendPacedFirstMessageBytes` / +`_send_paced_first_message_bytes`. That function was designed for the prewarm +case (one large pre-synthesised buffer) and resets drain+counter state on each +call. Applied per streaming chunk (~128 ms each), the drain+reset destroyed +mark back-pressure continuity and the per-sub-chunk playout sleep slowed +delivery below Twilio's playout rate, causing periodic buffer underruns +(crackling on the first message only). Subsequent LLM responses used the +simpler `synthesizeSentence` / `_synthesize_sentence` path (plain `sendAudio`) +and never crackled, confirming the fix direction. + +Fix: the streaming first-message path now uses the same plain +`encodePipelineAudio + sendAudio + markFirstAudioSent` pattern as subsequent +turns. The prewarm path (pre-synthesised buffer) is unchanged and still uses +`sendPacedFirstMessageBytes` / `_send_paced_first_message_bytes` because that +buffer can be several seconds long and needs mark-gated pacing. Files: +`libraries/typescript/src/stream-handler.ts`, +`libraries/python/getpatter/stream_handler.py`. + +### Changed — Cerebras usage-chunk fallback: INFO-once + DEBUG per iteration (Python + TypeScript parity) + +The char/4 fallback billing path in `services/llm_loop.py` / +`src/llm-loop.ts` previously emitted `logger.warning` / +`getLogger().warn` on every tool-loop iteration when the upstream +provider stream did not include a `usage` chunk. On Cerebras (the +common case for this fallback), a multi-tool turn could log 5-10 +identical WARN lines for the same call — drowning real warnings. + +Replaced with: first fallback in the call → INFO (so operators +still see it once with the full diagnostic context — `provider`, +`model`, `input_chars`, `output_chars`, `est_input_tokens`, +`est_output_tokens`); subsequent iterations → DEBUG with the +iteration index and a per-LLMLoop `_usage_missing_count` / +`_usageMissingCount` total so the volume is still visible at +DEBUG level. No behavioural change — billing still uses char/4 +estimation. Files: `libraries/python/getpatter/services/llm_loop.py`, +`libraries/typescript/src/llm-loop.ts`. + +### Changed — Krisp VIVA TypeScript scaffold: refreshed unavailability message (2026-05) + +The `KrispVivaFilter` constructor in +`libraries/typescript/src/providers/krisp-filter.ts` already throws +with guidance because Krisp does not publish a Node.js server SDK as +of 2026-05. Refreshed the message to include the verification date, +explicitly distinguish "server Node SDK" from existing browser/RN +third-party wrappers, and note that those wrappers (browser WASM and +mobile client variants) are scoped to local microphone capture and +cannot process Patter's server-side PCM/mulaw audio. Python +`KrispVivaFilter` and TS `DeepFilterNetFilter` remain the only +shipped paths. No code behaviour change. + +### Fixed — Barge-in gate regression test: prewarmed first message must remain interruptible + +Locked in with parity tests on both SDKs that `_stream_prewarm_bytes` / `streamPrewarmBytes` open the barge-in gate (`_first_audio_sent_at` / `firstAudioSentAt`) once the first chunk reaches the wire. The gate was already opened by `_begin_speaking(is_first_message=True)` ahead of streaming, but a future refactor of the `_begin_speaking` path could regress the prewarm path silently — the per-chunk `_mark_first_audio_sent` call inside the streaming loop is the last line of defence and now has explicit coverage in `test_stream_prewarm_bytes_opens_barge_in_gate_on_first_chunk` (Python) and `opens the barge-in gate by stamping firstAudioSentAt after the first chunk` (TypeScript). + +Files: `libraries/python/tests/test_prewarm.py`, `libraries/typescript/tests/unit/prewarm.test.ts`. + +### Fixed — `ElevenLabsWebSocketTTS.adopt_websocket` leaked the previous parked WS when called outside an event loop + +`ElevenLabsWebSocketTTS.adopt_websocket` (Python) closed any previously parked WS handle via `asyncio.create_task(prev.ws.close())`. When invoked from a sync context with no running event loop — e.g. cleanup hooks fired from `__del__`, atexit handlers, or signal-driven teardown — the `create_task` call raised `RuntimeError` which the code silently swallowed with a bare `except RuntimeError: pass`, leaking the socket FD. ElevenLabs would eventually close the remote side after the inactivity timeout, but the FD on our side stayed allocated until process exit. + +The fix keeps the async fast path when a loop is running, and falls back to a best-effort synchronous `transport.close()` (non-blocking, skips the WS close handshake but cleans up the file descriptor) when no loop is available. A warning log is emitted on the fallback path so the FD-leak symptom shifts from "silent" to "logged". + +The TypeScript counterpart `adoptWebSocket` is unaffected — `ws.close()` from the `ws` package is synchronous so the same scenario doesn't reach an analogous error branch. + +Files: `libraries/python/getpatter/providers/elevenlabs_ws_tts.py`, `libraries/python/tests/unit/test_elevenlabs_ws_tts.py` (new `TestAdoptWebSocketCleanup`). + +### Added — `patter.*` OTel attribute helpers in the TypeScript SDK (parity with Python) + +The Python SDK ships `record_patter_attrs`, `patter_call_scope`, and `attach_span_exporter` (in `getpatter.observability.attributes`) for stamping `patter.cost.*` / `patter.latency.*` span attributes and wiring an OTel `SpanExporter` into the tracer provider. The TypeScript SDK previously had no equivalent surface — calling code that wanted to record those attributes had to no-op manually or import `@opentelemetry/api` directly, which broke cross-SDK parity per `.claude/rules/sdk-parity.md`. + +This change ports the helpers to TypeScript as no-ops by default. When `PATTER_OTEL_ENABLED` is unset or `@opentelemetry/api` is not installed, every helper is a fast no-op, so existing call sites stay zero-cost. Available as: + +```ts +import { + recordPatterAttrs, + patterCallScope, + attachSpanExporter, + DEFAULT_SIDE, +} from 'getpatter/observability'; +``` + +Semantic mapping (1:1 with Python): + +- `recordPatterAttrs(attrs)` ↔ `record_patter_attrs(attrs)` +- `patterCallScope({ callId, side }, fn)` ↔ `patter_call_scope(call_id=..., side=...)` (the JS form takes an async callback because JS lacks `with`-style context managers; the closure is the scope body) +- `attachSpanExporter(patterInstance, exporter, { side })` ↔ `attach_span_exporter(patter, exporter, side=...)` + +Files: `libraries/typescript/src/observability/attributes.ts` (new), `libraries/typescript/src/observability/index.ts` (re-exports), `libraries/typescript/tests/unit/observability-attributes.test.ts` (new). + +### Fixed — `EOUMetrics` field semantics + unit parity between Python and TypeScript SDKs + +The Python implementation in `libraries/python/getpatter/services/metrics.py:_emit_eou_metrics` had `end_of_utterance_delay` and `transcription_delay` swapped relative to the TypeScript counterpart, and emitted them in seconds while the TypeScript SDK and the rest of the observability surface (`ttfb_ms`, `turn_ms`) use milliseconds. The dashboard, EventBus subscribers and any downstream exporter consuming both SDKs would have seen the two fields disagree by a factor of 1000× AND swapped — silently corrupting end-of-utterance latency dashboards on cross-SDK fleets. + +The convention is now uniform across both SDKs (locked in by tests): + +- `end_of_utterance_delay` / `endOfUtteranceDelay` = `stt_final − vad_stopped` (milliseconds) +- `transcription_delay` / `transcriptionDelay` = `turn_committed − vad_stopped` (milliseconds) +- `on_user_turn_completed_delay` / `onUserTurnCompletedDelay` = pipeline hook execution time (milliseconds) + +Negative deltas from clock skew or out-of-order timestamps are now clamped to `0` on both sides (the TypeScript side already did this; Python now does too). + +Files: `libraries/python/getpatter/services/metrics.py`, `libraries/python/getpatter/observability/metric_types.py` (docstring), `libraries/python/tests/test_metrics.py` (new `TestEOUMetricsEmission`), `libraries/typescript/tests/unit/metrics.test.ts` (new `emitEouMetrics field semantics` block). + +### Fixed — Barge-in bug bundle: 6.8s latency outliers, double-talk dispatch, stale anchors, firstMessage uninterruptible (Python + TypeScript parity) + +Real PSTN test (round 10f, 11 turns with user-initiated interruptions) surfaced four correlated bugs in the barge-in pipeline that the previous strategy work in 0.6.1 did not cover. Investigation report (`/private/tmp/.../a6fae04df253294f2.output`) traced all four to anchor mismanagement around the interrupt boundary plus an over-aggressive VAD threshold. + +**Bug 1 — endpoint_ms == stt_ms == 6818 ms (dishonest p95 outliers).** `recordSttComplete` was fabricating `_endpointSignalAt = _sttComplete` when no legitimate VAD `speech_end` had fired, producing a synthetic anchor that then made `endpoint_ms = _turnCommittedMono − _turnStart` (the entire turn duration). Fix: never fake the anchor; let `endpoint_ms` be `undefined` on the affected turn and increment a `_endpointSignalMissingCount` counter for observability. Added a 100 ms post-barge-in gate (`_lastBargeinAt`): the next turn's `endpoint_ms` / `stt_ms` are dropped from the percentile distribution since post-barge-in anchors are inherently noisy. Files: `libraries/typescript/src/metrics.ts:412-422,538-549,572-596,870-984`, `libraries/python/getpatter/services/metrics.py:289,362,395,696`. + +**Bug 2 — double-talk: agent answered the 1st sentence while user said the 2nd.** Deepgram emits `is_final` on any pause > a few ms; the SDK dispatched LLM immediately. Two fixes ship together: + +1. **Bumped Silero `minSilenceDuration` default `0.1` → `0.4` s**. The previous 100 ms threshold fired VAD `speech_end` on natural inter-sentence pauses (typically 200-400 ms), which then prematurely finalised the user turn at the STT layer. 0.4 s is the industry-standard default for telephony agents: bridges intra-utterance pauses without delaying single-sentence turns by more than the natural conversational gap. Files: `libraries/typescript/src/providers/silero-vad.ts:366-378`, `libraries/python/getpatter/providers/silero_vad.py:125`. +2. **Synchronous STT-final → LLM dispatch (no debounce)**. An earlier 400 ms debounce attempt (`_scheduleTurnCommit` / `_runDeferredTurnCommit`) was prototyped and rolled back before release: the partial-transcript reschedule branch overwrote the dispatched FINAL text with the latest partial, silently dropping entire user turns during slow-LLM windows. Verified on real PSTN (round 10k, gpt-5-nano: 3 of 5 user turns dropped). The shipped behaviour dispatches on `is_final` immediately; when Deepgram emits two close-together finals like `"What's the"` then `"What's the best?"`, the SDK answers both (benign double-answer) instead of dropping the first (catastrophic). Tracked future improvements documented internally — options include raising Deepgram `endpointingMs` per-agent, queue-cancel semantics, and sentence-segment merge in `commitTranscript`. Files: `libraries/typescript/src/stream-handler.ts` (inline dispatch restored), `libraries/python/getpatter/stream_handler.py` (`_dispatch_turn` called inline from `_stt_loop`). + +**Bug 3 — anchors stale after strategy-confirmed barge-in.** `runBargeInCancel` cleared anchors via `_resetTurnState()` but never re-anchored to the next legitimate VAD `speech_start`; the next turn either inherited stale anchors or anchored to the first inbound audio byte (adding ~250 ms ring-buffer delay to every post-barge-in turn). Added `anchorUserSpeechStart()` calls in three places: after `recordTurnInterrupted` in `runBargeInCancel`, and on the pending-barge-in timeout path. Also `_resetTurnState` now resets `_initialTtfbEmitted` (TS) / `_initial_ttfb_emitted` + `_llm_ttfb_emitted` + `_tts_ttfb_emitted` (Py) so EventBus TTFB re-fires after barge-in when `reportOnlyInitialTtfb=true`. Files: `libraries/typescript/src/stream-handler.ts:2034-2058`, `libraries/typescript/src/metrics.ts:846-867`, Python mirrors. + +**Bug 4 — firstMessage was uninterruptible by VAD for 300-800 ms.** `canBargeIn()` gates on `firstAudioSentAt !== null`, but that field was only stamped when the first audio chunk arrived from the TTS provider — meaning the 250 ms anti-flicker timer didn't start until the user had already missed the TTFB window. Fix: `beginSpeaking(isFirstMessage=true)` now stamps `firstAudioSentAt = Date.now()` synchronously, so the gate timer runs in parallel with TTS TTFB. The firstMessage TTS loop already breaks on `!this.isSpeaking`, so user speech now propagates cancellation correctly. Files: `libraries/typescript/src/stream-handler.ts:357,1477`, `libraries/python/getpatter/stream_handler.py:3187,2043`. + +### Changed — Dashboard percentile threshold lowered 5 → 2 turns + +`LatencyPanel` and `MetricsPanel` displayed `—` for `p50` / `p95` until a call had ≥5 turns. On most PSTN calls (typically 4-7 turns) the detail pane showed dashes while the call-list `P95 LATENCY` column already showed a real number via `avg` fallback — confusing for users comparing the two surfaces. Lowered to 2 turns so the detail pane matches the list column. With n=2 the percentile is statistically thin but consistent with what the list shows. + +Files: `dashboard-app/src/components/LatencyPanel.tsx:12`, `dashboard-app/src/components/MetricsPanel.tsx:76,127`. Bundle synced to `libraries/{typescript,python}/.../dashboard/ui.html` via `dashboard-app/scripts/sync.mjs`. + +### Added — Krisp VIVA noise-suppression scaffold for TypeScript SDK + +Mirrors the Python `KrispVivaFilter` API at `libraries/typescript/src/providers/krisp-filter.ts` for cross-SDK parity. Class signature accepts the same options (`modelPath`, `noiseSuppressionLevel`, `frameDurationMs`, `sampleRate`) but throws at construction time with guidance — Krisp does not publish an official Node.js SDK as of 2026-05. Opt-in, proprietary, license required, no default-on. Patter ships only the interface; users supply SDK + `.kef` model. + +Available paths today: +1. **Python SDK**: `from getpatter.providers.krisp_filter import KrispVivaFilter` — fully implemented (existed prior to this change, unmodified). Requires `pip install getpatter[krisp]` + `KRISP_VIVA_SDK_LICENSE_KEY` + `KRISP_VIVA_FILTER_MODEL_PATH`. +2. **TypeScript today**: `new DeepFilterNetFilter({ modelPath })` from `getpatter` — community ONNX export, no license. `KrispVivaFilter` throws until a Node binding is available. + +New top-level exports from `getpatter`: `KrispVivaFilter`, `KrispVivaFilterOptions`, `KrispSampleRate`, `KrispFrameDuration`, `DeepFilterNetFilter`, `DeepFilterNetOptions`. The TS scaffold closes when an official Krisp Node SDK ships or a community NAPI/WASM binding becomes available. + +### Fixed — Dashboard live SSE update wiped transcripts + latency from prior calls + +When a new call started, the live SSE refresh in +`dashboard-app/src/hooks/useDashboardData.ts:103` rebuilt the entire +calls array from `mergeCalls(active, recent)` without consulting the +previous state. If the new payload had any field as undefined — common +when the server-side `MetricsStore.updateCallStatus` writes a synthetic +"terminal" record with `metrics: undefined` ahead of the true +`recordCallEnd` — the prior call lost its transcripts and latency p50/ +p95 in the UI. Added `mergeCallPreserving(prev, next)` that does +`next.field ?? prev.field` per critical field, masking the lossy +secondary records server-side. The SDK-side double-write race in +`libraries/typescript/src/dashboard/store.ts:134-186` is flagged with a +TODO for 0.6.2. + +### Changed — `elevenlabs.TTS` facade now defaults to WebSocket streaming (Python + TypeScript parity) + +Industry best practice for telephony agents is a long-lived WS connection — every other Patter STT adapter (Deepgram nova-3, Cartesia ink-whisper, Whisper streaming, AssemblyAI) already runs on persistent WS. ElevenLabs TTS was the outlier: the default `elevenlabs.TTS()` facade opened a fresh HTTP POST per sentence (TLS handshake, DNS, full request setup repeated on every turn), producing measured TTFB p50 ~265 ms on PSTN — vs ElevenLabs's published server-side TTFT of ~75 ms. The gap was almost entirely HTTP setup, not synthesis. + +Flipped both SDKs to extend the WebSocket class (`_ElevenLabsWebSocketTTS` / `_ElevenLabsTTSWebSocket`) by default. Expected TTFB p50 drop: ~265 ms → ~80-100 ms (after the first turn pays one handshake; turn 2+ reuses the open WS). + +- TS: `libraries/typescript/src/tts/elevenlabs.ts` — `class TTS extends _ElevenLabsWebSocketTTS`. Default `voiceId="EXAVITQu4vr4xnSDxMaL"`, `modelId="eleven_flash_v2_5"`, `outputFormat="pcm_16000"`, `autoMode=true`. `providerKey` flipped `"elevenlabs"` → `"elevenlabs_ws"`. `for_twilio`/`for_telnyx` signatures unchanged. +- Py: `libraries/python/getpatter/tts/elevenlabs.py` — `class TTS(_ElevenLabsWebSocketTTS)` with matching defaults. `chunk_size` kept as tolerated-but-ignored kwarg (the WS path doesn't use it) to avoid breaking pinned callers. +- Compatibility aliases preserved: `elevenlabs_ws.TTS` (TS + Py) now re-exports from `elevenlabs.TTS`. `ElevenLabsWebSocketTTS` top-level symbol unchanged. + +**REST opt-out** — new top-level export `ElevenLabsRestTTS` in both SDKs. Use when: +- The free / starter tier (WS requires Pro plan; the WS class raises `PLAN_REQUIRED_MSG` directing callers to `ElevenLabsRestTTS`). +- The `eleven_v3` model (HTTP-only — the WS class rejects it at construction with the same redirect). + +```ts +// TS +import { ElevenLabsRestTTS } from "getpatter"; +const tts = new ElevenLabsRestTTS(process.env.ELEVENLABS_API_KEY!); +``` +```python +# Python +from getpatter import ElevenLabsRestTTS +tts = ElevenLabsRestTTS(api_key=os.environ["ELEVENLABS_API_KEY"]) +``` + +Dashboard label-prettifier extended in `dashboard-app/src/components/{CostPanel,MetricsPanel}.tsx` — `titleCase()` regex now strips `_ws` / `_rest` transport suffixes in addition to `_stt` / `_tts` / `_llm` role suffixes. `"elevenlabs_ws"` now renders as "Elevenlabs" without the suffix bleed into UI. Repeated `+` handles compound suffixes (`"cartesia_tts_ws"` → `"Cartesia"`). + +Provider error messages in `providers/elevenlabs-ws-tts.{ts,py}` updated: `payment_required` and `eleven_v3` rejections now direct users to `ElevenLabsRestTTS` (was `ElevenLabsTTS` — which is now the WS facade itself, making the previous text recursive). + +Tests: 173 Python (was 170) + 95 TS (was 93) pass. 1 REST-specific assertion in `test_tts_facade_language.py` migrated to `ElevenLabsRestTTS` (the `chunk_size == 4096` default). 2 new tests each side verify the flip semantics and the opt-out is not aliased. + +Acceptance matrix: 25 duplicate `outbound-*-elevenlabs-ws.ts` scenarios removed (now functionally identical to `outbound-*-elevenlabs.ts`). Added 1 explicit regression `outbound-deepgram-cerebras-elevenlabs-rest.ts` to keep the REST path exercised. `_manifest.json` updated (78 entries). + +Migration: **0 code changes** for callers using the default — they automatically benefit from the latency drop. **1-line import rename** for callers who deliberately want HTTP REST (`ElevenLabsTTS` → `ElevenLabsRestTTS`). + +### Fixed — Console "Call ended" summary log p95 used `total_ms`, not `agent_response_ms` (Python + TypeScript parity) + +The single-line `[PATTER] Call ended: ... p95=Xms` log emitted by `stream_handler.ts` (TS) and `telephony/{twilio,telnyx}.py` (Py) at the end of every call read `latency_p95.total_ms` — the round-trip duration that **includes** how long the user spoke (`user_speech_duration_ms`), not the system-controlled wait time. The metrics module itself flags this in `metrics.ts:85-89`: "Unlike `total_ms` (which spans the user's entire utterance and therefore grows with how long the user spoke), `agent_response_ms` isolates the system-controlled latency." + +The dashboard already shows the correct field — `agent_response_ms` — under the **"p95 wait"** tile (`LatencyPanel.tsx:48`, `mappers.ts:221`). The console log diverged: a 51.6 s, 7-turn call where the user spoke ~1.2 s/turn printed `p95=2577ms` while the dashboard showed `p95 wait=1361ms` for the same call. The 1361 ms is the genuine user-perceived wait; the 2577 ms confused users into thinking the SDK was slow. + +Switched both SDKs to read `latency_p95.agent_response_ms` (fallback `total_ms` for legacy short calls where the percentile isn't computed) and renamed the label to **`p95 wait=Xms`** to match the dashboard tile word-for-word. Files: `libraries/typescript/src/stream-handler.ts:2810-2820`, `libraries/python/getpatter/telephony/twilio.py:657-680`, `libraries/python/getpatter/telephony/telnyx.py:792-810`. + +### Fixed — Telnyx pricing direction-aware: inbound 2× over-bill resolved + +Audited against https://telnyx.com/pricing/elastic-sip (verified +2026-05-11). The previous flat `"telnyx": $0.007/min` over-billed +inbound calls by 2× ($0.0035 real) and approximately matched outbound +($0.005-0.009 range). Split into two entries: +- `telnyx_inbound`: $0.0035/min (US local termination) +- `telnyx_outbound`: $0.007/min (Pay-As-You-Go mid-range) +The legacy `telnyx` key is preserved at $0.007 for backward-compat with +users who override `pricing={"telnyx": {...}}` and don't know direction. +Billing granularity confirmed per-minute (not per-second as previous +internal docs claimed). Files: `libraries/python/getpatter/pricing.py`, +`libraries/typescript/src/pricing.ts`. Tests added at +`libraries/python/tests/test_pricing.py`, +`libraries/typescript/tests/pricing.test.ts`. + +### Fixed — Python Twilio STT cost 4× over-bill (sample rate / bytes-per-sample mismatch) + +`libraries/python/getpatter/telephony/twilio.py` configured the metrics +STT format as `(sample_rate=8000, bytes_per_sample=1)` (mulaw 8 kHz), +but `stream_handler.py` already decodes the inbound mulaw to PCM16 +@ 16 kHz before feeding bytes to `metrics.add_stt_audio_bytes()`. With +the inverted format, every 60 s of real audio was reported as 240 s and +billed 4× the true cost ($0.0192 instead of $0.0048 against Deepgram +Nova-3 at $0.0048/min). TypeScript was unaffected (default 16000/2 was +never overridden); Python Telnyx was unaffected (already configured 16000/2). +Fix: `configure_stt_format(sample_rate=16000, bytes_per_sample=2)` in +the Twilio adapter, plus a regression test asserting 1.92 MB of PCM16 +bytes = 60 s of audio. Customers were over-billed; refund window TBD. + +### Fixed — Dashboard "−$X cached" badge dead for Realtime prompt-caching savings + +The SDK emits `cost.llm_cached_savings` (Realtime / Anthropic prompt +caching discount) but `dashboard-app/src/lib/mappers.ts:computeCost()` +never read it, so `Call.cost.cached` was always undefined and the badge +in `CostPanel.tsx:64` never rendered. Wired the field through +`api.ts:CallCost` (added `llm_cached_savings?: number` + `parseCost`) +and the mapper now populates `result.cached`. The "−$0.00X cached" line +now appears next to the LLM cost row whenever a Realtime call has any +cached-token savings. + +### Changed — LLM usage-chunk char/4 fallback log bumped DEBUG → WARN (Python + TypeScript) + +The 0.6.1 char/4 fallback (added when Cerebras was observed dropping +the `usage` chunk on some streams) was logging at DEBUG, so silent-zero +incidents only showed up in dev runs. Bumped to WARN in +`libraries/python/getpatter/services/llm_loop.py` and +`libraries/typescript/src/llm-loop.ts` so production observability +surfaces it. Message: "LLM usage chunk missing from {provider}/{model}; +estimating output_tokens=N via char/4 fallback". + +### Fixed — Deepgram STT pricing reflected legacy standard rate, not the current PAYG promo (Python + TypeScript parity) + +Audited against https://deepgram.com/pricing (verified 2026-05-11). Deepgram is currently running a "Limited-time promotional rates on streaming" tier that customers actually pay today; the prior $0.0077/min Nova-3 figure was the launch-era standard rate that has been struck through on the public page. + +| Model | Old (USD / min) | New (USD / min) | Notes | +|-------|-----------------|-----------------|-------| +| `nova-3` (default) | $0.0077 | **$0.0048** | over 60% | +| `nova-3-multilingual` | $0.0092 | **$0.0058** | over 58% | +| `flux` (added) | — | **$0.0065** | Flux English; new event-driven STT (2026) | +| `flux-english` (added) | — | **$0.0065** | alias of `flux` | +| `flux-multilingual` (added) | — | **$0.0078** | new | +| `nova-2`, `nova`, `whisper-*` | unchanged | unchanged | legacy / non-Nova-3 tiers | + +Dropped the `"deepgram"` provider-level default from $0.0077 to $0.0048 (Nova-3 monolingual is the Patter default model). A 25-minute call against the default would have reported $0.1925 in `cost.stt` instead of the actual $0.12 — over-reporting by ~60%. Customers were never undercharged; the dashboard line item was wrong. Files: `libraries/python/getpatter/pricing.py`, `libraries/typescript/src/pricing.ts`. Tests updated at `libraries/python/tests/test_pricing.py`, `libraries/typescript/tests/pricing.test.ts`, and the matching soak tests. Revisit when Deepgram removes the promo banner. + +### Fixed — ElevenLabs pricing table overcharged Flash 20% and Multilingual v2 / v3 by 80-200% (Python + TypeScript parity) + +Audited against the canonical public API pricing page at https://elevenlabs.io/pricing/api (verified 2026-05-11). The per-1K-character API/overage rate is flat across all plan tiers (Free → Business); only the included character bundle varies. Patter's 2026-05 table reflected legacy Creator-plan overage figures and a launch-era v3 quote that have since been consolidated. + +| Model | Old (USD / 1K chars) | New (USD / 1K chars) | Notes | +|-------|----------------------|----------------------|-------| +| `eleven_flash_v2_5` | $0.06 | **$0.05** | Patter default; was 20% over | +| `eleven_turbo_v2_5` | $0.05 | $0.05 | unchanged ✓ | +| `eleven_multilingual_v2` | $0.18 | **$0.10** | was 80% over | +| `eleven_v3` | $0.30 | **$0.10** | grouped with multilingual v2 on the public page; was 200% over | +| `eleven_monolingual_v1` (legacy) | $0.18 | **$0.10** | matches multilingual tier | + +Also dropped the `"elevenlabs"` / `"elevenlabs_ws"` provider-level default from $0.06 to $0.05 (flash_v2_5 is the Patter default model). A 5-turn call against the default would have reported $0.000060/char × N chars instead of the actual $0.000050/char — over-reporting LLM-bill-equivalent TTS cost by 20%. Customers were never undercharged, but the dashboard cost line was wrong. Files: `libraries/python/getpatter/pricing.py`, `libraries/typescript/src/pricing.ts`. Tests updated at `libraries/python/tests/test_pricing.py`, `libraries/typescript/tests/pricing.test.ts`, plus the matching soak tests. + +### Fixed — Dashboard cost labels leaked provider-key suffix (`Cartesia_stt STT` → `Cartesia STT`) + +The Cost panel's `titleCase()` helper rendered raw SDK `provider_key` literals (e.g. `cartesia_stt`, `elevenlabs_tts`) which the SDK uses to disambiguate provider-class lookups. The `_stt` / `_tts` / `_llm` suffix is internal noise: the panel already shows the role label next to the swatch ("STT", "TTS", "LLM"), so the suffix duplicated context and produced strings like "Cartesia_stt STT · ink-whisper". Stripped the suffix in both `dashboard-app/src/components/CostPanel.tsx` and `dashboard-app/src/components/MetricsPanel.tsx` `titleCase()` so labels render "Cartesia STT · ink-whisper" / "Elevenlabs TTS · eleven_flash_v2_5". + +### Fixed — Phantom `speech_start` during agent TTS contaminated turn anchors (Python + TypeScript parity) + +A real PSTN call surfaced `user_speech_duration_ms` of 5-7 seconds for utterances the caller actually spoke in ~1 second. Forensic timeline reconstruction (`releases/0.6.0/typescript/call-logs/.../CA6d7fc612...`) pinned the contamination to two bug classes uncovered by parallel-agent audit (forensic + architect + adversarial + provider-reviewer agreement): + +1. **Phantom-speech-start anchor contamination** — `StreamHandler` called `metrics.start_turn_if_idle()` on EVERY VAD `speech_start` event, including the ones suppressed during the per-turn warmup gate (`_can_barge_in() == False`). With AEC enabled this is a ~1 s window; without AEC it is the 250 ms anti-flicker margin. Background noise / echo / agent self-loopback during that window emitted a `speech_start` that was correctly suppressed for the barge-in path BUT silently stamped `_turn_start` at the bleed-through instant. The legitimate user `speech_start` that fired seconds later then no-op'd because `start_turn_if_idle` only acts when `_turn_start is None`. Result: `user_speech_duration_ms = (endpoint_signal_at − stale_turn_start) * 1000`, often 5-7 s. + +2. **Stale `_endpoint_signal_at` across dropped final transcripts** — when a final transcript arrived but `commitTranscript` / `_commit_transcript` returned False (dedup window / rejected barge-in / `afterTranscribe` veto, e.g. the "Okay." swallow on a strategy-pending barge-in), the previously-stamped VAD-end anchor was never cleared. The NEXT legitimate utterance inherited that stale anchor, so its `endpoint_ms` measured the silence gap between the dropped utterance and the real one. + +Both classes fixed with a single new metrics primitive and two call-site swaps: + +- **`anchor_user_speech_start()` (Python) / `anchorUserSpeechStart()` (TypeScript)** — Pipecat-style "every legitimate VAD `speech_start` re-anchors the turn pre-commit". Resets `_turn_start`, `_endpoint_signal_at`, `_vad_stopped_at`, `_stt_final_at`, `_stt_complete`, `_llm_first_token`, and the TTFB-emitted guard. No-ops once `_turn_committed_mono` is set (post-commit barge-ins follow the existing `record_turn_interrupted` path). Files: `libraries/python/getpatter/services/metrics.py`, `libraries/typescript/src/metrics.ts`. + +- **`stream_handler.py` / `stream-handler.ts` VAD `speech_start` handler** — explicit `phantom_suppressed` boolean gates ALL metrics state mutation: suppressed events log only, legitimate events call `anchor_user_speech_start()` instead of the old `start_turn_if_idle()`. The strategy-pending barge-in branch also switched from `start_turn_if_idle` to the new primitive so re-anchoring happens consistently on every legitimate `speech_start`. + +- **Dropped-final-transcript reset** — when `commitTranscript`/`_commit_transcript` returns False on an `is_final` / `speech_final` transcript, the same `anchor_user_speech_start()` is invoked so the discarded utterance's anchors don't leak into the next turn. + +### Fixed — Cartesia STT `finalize()` exposed so VAD `speech_end` can force-flush (Python + TypeScript parity) + +The 0.5.5 fast-path at `stream_handler.py:3070-3077` ("on VAD `speech_end`, call `stt.finalize()` so the provider doesn't wait for its natural-pause heuristic") was a no-op for Cartesia: `CartesiaSTT` only sent the `finalize` text frame from its private `close()` method on session shutdown, and `getattr(self._stt, "finalize", None)` returned None. The SDK's authoritative VAD silence detection (SileroVAD, 250 ms threshold) was being overridden by Cartesia's conservative internal endpointing (observed 2-7 s on PSTN audio with background hiss). + +Added `async finalize()` to both `CartesiaSTT` (Python) and `CartesiaSTT` (TypeScript) that sends the canonical `finalize` text frame on the live WebSocket. The wired-but-no-op fast-path now triggers a deterministic VAD-driven STT finalisation, parity with Deepgram. Files: `libraries/python/getpatter/providers/cartesia_stt.py`, `libraries/typescript/src/providers/cartesia-stt.ts`. + +### Fixed — Cerebras pricing table overcharged 1.5-2.4x across multiple models (Python + TypeScript parity) + +Audited against the canonical per-model docs pages at `https://inference-docs.cerebras.ai/models/`. Patter's 2026-05-08 table conflated launch-blog quotes with the current "Exploration pricing" banner shown on each model docs page. Corrections: + +| Model | Old (in/out) | New (in/out) | Source | +|-------|--------------|--------------|--------| +| `gpt-oss-120b` | $0.85 / $1.20 | $0.35 / $0.75 | inference-docs.cerebras.ai/models/openai-oss | +| `llama3.1-8b` | $0.10 / $0.20 | $0.10 / $0.10 | inference-docs.cerebras.ai/models/llama-31-8b | +| `qwen-3-235b-a22b-instruct-2507` | $1.00 / $1.50 | $0.60 / $1.20 | inference-docs.cerebras.ai/models/qwen-3-235b-2507 | +| `qwen-3-coder-480b` | (missing → $0) | $2.00 / $2.00 | cerebras.ai/blog/qwen3-coder-480b | + +Pre-fix a 5-turn pipeline call against the Patter-default `gpt-oss-120b` logged ~$0.000117 in `cost.llm` instead of the actual ~$0.000088 — over-reporting by a factor of ~1.3. Net: customers were never undercharged, but the dashboard line item was wrong (and 50% high for `gpt-oss-120b` specifically). Files: `libraries/python/getpatter/pricing.py`, `libraries/typescript/src/pricing.ts`. Tests updated at `libraries/python/tests/test_pricing.py`. + +### Fixed — Dashboard cost rendering flattened sub-cent values to `$0.00` (`fmtCostUSD` adaptive precision) + +The dashboard's per-row, per-stack, and aggregate spend tiles all used `toFixed(2)` or `toFixed(3)` for USD rendering. Cerebras `gpt-oss-120b` at $0.0001 / 5-turn-call rounds to `$0.00` under that rule, making the LLM cost line look as if billing was broken when in fact it was working end-to-end (token usage extracted from the streaming `usage` chunk, cost calculated, persisted to `metadata.json` at the correct precision). + +Added `fmtCostUSD(value)` helper (`dashboard-app/src/components/format.ts`) with magnitude-adaptive precision: ≥$0.01 → 2 decimals, ≥$0.001 → 3 decimals, ≥$0.0001 → 4 decimals, smaller values → 5 decimals. Applied across all 12 cost render sites (`App.tsx` spend tile, `CallTable.tsx` row total, `Metric.tsx` headline + per-call row, `CostPanel.tsx` × 5, `MetricsPanel.tsx` × 4). A 5-turn Cerebras pipeline call now shows `$0.00012` instead of `$0.00`. + +### Fixed — Dashboard latency metrics: real percentiles, correct waterfall, n<5 percentile gate + +The "Latency · this call" panel was showing three different numbers labelled wrong: +1. **"p50" was the avg of total_ms**, not the median (`mappers.ts:194` read `latencyP50: latencyAvg.total_ms`). +2. **The "llm" bar in the waterfall was the same fake p50**, double-counting non-LLM time (waterfall `llm = call.latencyP50` instead of `avg(llm_ms)`). The bar was off by ~5x on real PSTN data. +3. **p50/p95 were rendered with as few as 1 turn**, where percentiles are statistical noise (linear interpolation between two samples). + +Round-trip `total_ms` also includes the user-utterance duration on the speech-to-speech metric, which over-states user-perceived latency. The dashboard now exposes `agent_response_ms` (wait time after the user stops speaking) as a separate primary metric. + +Fixes shipped: +- **SDK serialization** (`libraries/python/getpatter/server.py`, `libraries/typescript/src/server.ts`) — `metadata.json` now persists the full `LatencyBreakdown` per percentile (`avg`, `p50`, `p95`, `p99`) with all components (`stt_ms`, `llm_ms`, `tts_ms`, `total_ms`, `agent_response_ms`, `endpoint_ms`, `user_speech_duration_ms`). The flat `p50_ms / p95_ms / p99_ms` totals are kept for backward-compat with consumers that read only summaries. +- **Dashboard hydrate** (`libraries/python/getpatter/dashboard/store.py`, `libraries/typescript/src/dashboard/store.ts`) — `_metrics_from_top_level` reads the full breakdown when present; falls back to the synthetic single-`total_ms` shim only for legacy metadata that lacked the breakdown. +- **Dashboard UI** (`dashboard-app/src/lib/api.ts`, `mappers.ts`, `components/CallTable.tsx`, `components/LatencyPanel.tsx`, `components/MetricsPanel.tsx`) — `Call` gains `llmAvg`, `turnCount`, `agentResponseP50/P95`. `latencyP50` now reads from `latency_p50.total_ms` (true median); the waterfall `llm` bar uses `llmAvg`. Percentile boxes render `—` and a "n turns — percentiles need ≥5" hint when `turnCount < 5`. The Latency panel adds a `p50 wait / p95 wait` pair sourced from `agent_response_ms`, the user-perceived "time waited after I stopped speaking" metric. + +Backward compat: legacy `metadata.json` (no `avg/p50/p95/p99` objects, only flat percentiles) still hydrates — those rows just lack the per-component breakdown in the panel and show `—` for `p50 wait / p95 wait`. No public API change. + +### Changed — First-turn cold-start: keep prewarmed WebSockets OPEN and adopt them at call connect (Python + TypeScript parity) + +Investigation of live PSTN-pipeline first-turn p95 latency (~3 s observed in production acceptance) showed the existing prewarm pattern (open WS, idle ~250 ms, close) saves only ~50-250 ms — DNS cache + edge-worker pinning at best. The dominant first-turn cost on PSTN pipeline is the synchronous TLS + WS-upgrade + protocol-handshake against STT (~150-400 ms) and TTS (~400-900 ms) when the call starts. Opening + closing a WS does NOT thread `session: ` across `new WebSocket()` calls in Node's `ws` package (and Python's `websockets` library has the same property at the TCP / TLS level), so each fresh open re-pays the full handshake. + +Structural fix: the prewarm pipeline now keeps each provider WebSocket OPEN during the carrier ringing window and hands the live socket off to the per-call `StreamHandler` at `start`, skipping the cold handshake entirely on the first turn. + +- **`Patter._prewarmed_connections` (Python) / `Patter.prewarmedConnections` (TS)** — new per-call_id cache holding pre-opened, fully-handshaked provider WebSockets. Populated by the new `_park_provider_connections(agent, call_id)` (Py) / `parkProviderConnections(agent, callId)` (TS), which runs in parallel with the carrier-side `initiate_call`. Each parked slot may hold up to three handles (`stt`, `tts`, `openai_realtime`); each is consumed exactly once. A 30 s safety TTL force-closes any slot whose carrier never fires `start`. Drained by `pop_prewarmed_connections(call_id)` on `start` (consumes the handles into the StreamHandler), `close_prewarmed_connections(call_id)` on call-failure paths (no-answer / busy / failed / canceled / AMD voicemail — wired through `_record_prewarm_waste`), and `disconnect()` on Patter teardown. Files: `libraries/python/getpatter/client.py` (cache + park task + helpers + helpers `_safe_close_handle`, `_close_parked_slot`), `libraries/typescript/src/client.ts` (parity, plus the new exported `ParkedProviderConnections` interface). + +- **Provider-level `open_parked_connection()` and `adopt_websocket(...)`** shipped on the three streaming providers most-affected by the cold-start cost: `CartesiaSTT`, `ElevenLabsWebSocketTTS`, `OpenAIRealtimeAdapter`. `open_parked_connection` opens the WS, sends the EXACT initial config the live `connect()` / `synthesize()` path sends (BOS frame for ElevenLabs WS, `session.update` round-trip for OpenAI Realtime), then returns the OPEN socket WITHOUT arming any recv / keepalive task — the handle is parked. `adopt_websocket` takes that handle, installs the recv + keepalive plumbing, and hands the live socket back to `StreamHandler` as if `connect()` had just finished. The TTS adapter uses a single-slot adoption queue so the existing `for await (const chunk of agent.tts.synthesizeStream(...))` call site continues to work without signature changes — and the BOS-already-sent flag prevents a protocol error on adoption. Files: `libraries/python/getpatter/providers/{cartesia_stt,elevenlabs_ws_tts,openai_realtime}.py`, `libraries/typescript/src/providers/{cartesia-stt,elevenlabs-ws-tts,openai-realtime}.ts`. + +- **`StreamHandler` adopt-or-connect** — the pipeline-mode initialisation path now polls `pop_prewarmed_connections(call_id)` BEFORE `stt.connect()` / TTS firstMessage. When a parked WS is still OPEN it is adopted (logged as `[CONNECT] callId=... source=adopted ms=0`); when the parked WS died between park and adopt (server timeout, network blip), the dead handle is discarded silently and the consumer falls back to a fresh `connect()` (logged as `source=fresh ms=`). When no parked slot exists at all (cache miss, prewarm task slower than carrier ringing, prewarm disabled), the path is byte-identical to the prior cold-start flow — backward-compatible. Realtime adapter adoption (separate `OpenAIRealtimeStreamHandler` code path) ships the API surface but is not yet wired through the realtime stream handler — pipeline mode dominates the affected use case and the realtime wiring is a follow-up. Files: `libraries/python/getpatter/stream_handler.py`, `libraries/typescript/src/stream-handler.ts`. + +### Fixed — Parallelise STT.connect with TTS firstMessage kickoff (Python + TypeScript parity) + +Pipeline-mode initialisation previously did `await stt.connect()` then `tts.synthesizeStream(firstMessage)` serially. STT only needs to be ready to receive incoming user audio, not to send the first agent message out — running the two in parallel saves an additional 200-400 ms on the first turn (real cost of a Deepgram / Cartesia / AssemblyAI WS upgrade). The STT receive loop launcher now awaits the deferred connect task before installing the message pump, so a half-open WS never surfaces "Not connected" on the first audio frame. Files: `libraries/python/getpatter/stream_handler.py`, `libraries/typescript/src/stream-handler.ts`. + +### Fixed — Pre-import AEC module at `Patter.serve()` (Python + TypeScript parity) + +The acoustic echo canceller (`getpatter.audio.aec.NlmsEchoCanceller`) was lazily imported on the first call when `agent.echo_cancellation=True`, costing ~150-400 ms of dynamic-import compile / link on the hot path. `Patter.serve()` now eagerly imports the module once when `echo_cancellation` is enabled, so the first call sees the cache-warm import. Pure data — no side effects on users who never enable AEC. Files: `libraries/python/getpatter/client.py`, `libraries/typescript/src/client.ts`. + +### Fixed — `[PREWARM]` and `[CONNECT]` timing instrumentation (Python + TypeScript parity) + +INFO-level log lines added to `_park_provider_connections` and the StreamHandler adopt-or-connect path so operators can attribute first-turn latency to specific providers without strace / packet capture. Format: `[PREWARM] callId= provider=stt ms=`, `[CONNECT] callId= provider=stt source=adopted|fresh ms=`. Files: `libraries/python/getpatter/client.py` + `stream_handler.py`, `libraries/typescript/src/client.ts` + `stream-handler.ts`. + +Tests: `libraries/python/tests/test_prewarm_handoff.py` (6 unit tests — park task invokes `open_parked_connection` on STT + TTS, parked WS stays OPEN past 250 ms, `pop_prewarmed_connections` consumes once, `close_prewarmed_connections` and `_record_prewarm_waste` drain parked sockets, no-op when neither provider supports parking) and `libraries/typescript/tests/unit/prewarm-handoff.test.ts` (6 parity tests). Both suites use authentic real code paths — only the WS handle is stubbed (it has no business in a unit test) — per `.claude/rules/authentic-tests.md`. Defaults preserved: `agent.prewarm` is still `true` by default, all existing tests pass without modification, and providers that do not implement `open_parked_connection` (everything except the three above) fall through to the prior `warmup()`-then-cold-`connect()` flow. + +### Fixed — Prewarm-firstMessage cache safety (5 issues, Python + TypeScript parity) + +The `Agent.prewarm_first_message` opt-in shipped earlier in 0.6.1 had five edge cases where the TTS bill was paid but the cached bytes either leaked, never reached the wire, or silently wasted spend. Each fix is per FIX number from the parity audit. Defaults preserved across the board: cap = 200 concurrent prewarmed calls, TTL = `ring_timeout + 5 s`. Tests added in `libraries/python/tests/test_prewarm.py` (14 new tests) and `libraries/typescript/tests/unit/prewarm.test.ts` + `libraries/typescript/tests/unit/server-routes.test.ts` (12 new tests). + +- **FIX #91 — cache eviction on abnormal hangup.** `_record_prewarm_waste` was only called from `Patter.end_call(call_sid)`. When a call went to `no-answer` / `busy` / `failed` / `canceled` (Twilio) or hit `call.hangup` / AMD voicemail (Telnyx), the `_prewarm_audio[call_id]` entry leaked until the user explicitly invoked `end_call`. Twilio status callback handler now invokes `record_prewarm_waste` for the four abnormal `CallStatus` values and on the AMD `machine_end_*` paths; Telnyx webhook handler now does the same on `call.hangup` (any `hangup_cause`) and on `call.machine.detection.ended` with `result == "machine"`. `_record_prewarm_waste` is now idempotent — the `_prewarm_consumed` set is checked first, so the status callback firing before `end_call` (or vice-versa) does not double-WARN. Files: `libraries/python/getpatter/server.py` (status / AMD / call.hangup branches), `libraries/python/getpatter/client.py` (`_record_prewarm_waste` idempotency guard, `_prewarm_consumed` set, server forwarding), `libraries/typescript/src/server.ts` (parity), `libraries/typescript/src/client.ts` (parity). + +- **FIX #92 — race start-vs-prewarm-task → orphan bytes.** When the carrier's `start` event arrived BEFORE the prewarm TTS task completed, `pop_prewarm_audio` returned `None`, the `StreamHandler` correctly fell back to live TTS, BUT the prewarm task continued in the background and eventually wrote bytes to `_prewarm_audio[call_id]` — orphaning them in the cache until `end_call` ran. Combined with FIX #91, every fast-pickup call leaked the prewarm bytes. **Option A (race-guard)** chosen for the minimum-fix bound. New `_prewarm_consumed: set[str]` tracks every `pop_prewarm_audio` call (hit OR miss). The prewarm task checks membership before writing bytes; on race-finish, the bytes are dropped and a WARN names the `call_id` plus byte count so the wasted spend is observable. Option B (200 ms wait window for the synth to land) was rejected as adding more latency-coupled state for marginal recovery. Files: `libraries/python/getpatter/client.py` (`_spawn_prewarm_first_message._run`), `libraries/typescript/src/client.ts` (parity). + +- **FIX #93 — `disconnect()` did not clean up prewarm.** Across `serve()` → `disconnect()` → `serve()` cycles within the same `Patter` instance, in-flight `_prewarm_tasks` continued to run (the TTS WebSocket stayed open and billed) and stale `_prewarm_audio` entries leaked. `disconnect()` now cancels every task in `_prewarm_tasks`, `await`s the cancellation via `asyncio.gather(..., return_exceptions=True)` (Py) / `Promise.allSettled` with a 1 s safety timeout (TS), cancels any pending TTL eviction tasks, and clears `_prewarm_audio` + `_prewarm_consumed`. The instance is fully reusable: a follow-up `serve()` sees a clean cache. Files: `libraries/python/getpatter/client.py` (`Patter.disconnect`), `libraries/typescript/src/client.ts` (parity). + +- **FIX #94 — Realtime/ConvAI silently waste TTS spend.** `agent.prewarm_first_message=True` paired with `agent.provider="openai_realtime"` or `"elevenlabs_convai"` paid the TTS bill on every outbound call but never streamed the cached bytes — the `StreamHandler` for those modes runs the firstMessage emit through the provider's own audio path, never consulting `pop_prewarm_audio`. `Patter.call` now checks the provider mode at `_spawn_prewarm_first_message` entry; when `provider != "pipeline"` it logs a WARN and refuses to spawn the synth task. Both SDK docstrings (`Agent.prewarm_first_message`, `AgentOptions.prewarmFirstMessage`) updated to document the constraint. Files: `libraries/python/getpatter/client.py`, `libraries/python/getpatter/models.py` (docstring), `libraries/typescript/src/client.ts`, `libraries/typescript/src/types.ts` (JSDoc). + +- **FIX #96 — prewarm cache unbounded (memory DoS).** A flood of `Patter.call(...)` invocations (legitimate or attacker-controlled) could pile up tens of MB of orphan TTS bytes that never evicted when the carrier never fired `start`. Two bounds added: (a) **size cap** at `_PREWARM_CACHE_MAX = 200` (Py) / `PREWARM_CACHE_MAX = 200` (TS) concurrent entries (live cache + in-flight synth tasks). When the cap is reached, new prewarm spawns are refused with a WARN and the call still proceeds — only the optimisation is skipped. (b) **TTL eviction**: a per-entry timer scheduled `ring_timeout + _PREWARM_TTL_GRACE_S` (default 5 s) after the synth task completes. If the cache entry is still present when the timer fires, it is dropped and a WARN names the byte count. The timer is cancelled on normal consumption (`pop_prewarm_audio`) and on `_record_prewarm_waste`, so spurious WARNs never fire after a clean drain. Both `_PREWARM_CACHE_MAX` and `_PREWARM_TTL_GRACE_S` (Py) / `PREWARM_CACHE_MAX` and `PREWARM_TTL_GRACE_MS` (TS) are exported as module-level constants for tests and operator visibility. Files: `libraries/python/getpatter/client.py` (cap check + TTL eviction in `_spawn_prewarm_first_message`, `_evict_prewarm_after`), `libraries/typescript/src/client.ts` (parity). + +### Changed — Concrete STT/TTS WebSocket prewarm overrides + OpenAI Realtime native warmup (Python + TypeScript parity) + +The first prewarm pass (above) shipped LLM HTTPS-GET warmup but left STT and TTS providers on the no-op default. A second look at the cold-start latency budget revealed a priority inversion: an HTTPS GET against an LLM `/models` endpoint warms only DNS + TLS + connection pool — it does NOT prime the inference path itself, while a streaming-STT or streaming-TTS WebSocket pre-handshake (full TLS + auth + initial config exchange) saves 200-500 ms per call on cold start. OpenAI's Realtime API exposes a native warmup primitive (`response.create` with `generate: false`) that prepares request state without billing tokens. This entry rebalances the prewarm pipeline to put the wins where they actually live. + +- **STT WebSocket prewarms** — concrete `warmup()` overrides shipped on `DeepgramSTT`, `CartesiaSTT`, and `AssemblyAISTT`. Each opens the streaming WebSocket (full DNS + TLS + auth handshake), idles ~250 ms so the provider edge keeps the session warm in its routing table, then closes cleanly. By the time `connect()` is invoked at call-pickup the resolver and TLS session are hot — net wire time saving of 200-500 ms vs a cold WS open. **Billing safety**: all three providers bill on streamed audio seconds (Deepgram per [pricing](https://deepgram.com/pricing); Cartesia per [STT API reference](https://docs.cartesia.ai/2025-04-16/api-reference/stt/stt); AssemblyAI per [pricing](https://www.assemblyai.com/pricing)). Opening + closing the WebSocket without sending any audio frames does not consume billable seconds. The override docstrings reference the per-provider billing model so future contributors don't accidentally regress this. Files: `libraries/python/getpatter/providers/{deepgram_stt,cartesia_stt,assemblyai_stt}.py`, `libraries/typescript/src/providers/{deepgram-stt,cartesia-stt,assemblyai-stt}.ts`. + +- **TTS prewarms — WebSocket and HTTP** — concrete `warmup()` overrides shipped on `ElevenLabsWebSocketTTS` (WS), `CartesiaTTS` (HTTP `/tts/bytes`), and `InworldTTS` (HTTP `/tts/v1/voice:stream`). The ElevenLabs WS variant opens the stream-input WebSocket, sends the protocol-required single-space keepalive `{"text": " "}` so the server creates and warms the session, idles ~250 ms, then closes. The HTTP-only providers (Cartesia, Inworld) issue a lightweight `GET /voices` (Cartesia) or `HEAD` against the streaming base host (Inworld) to warm DNS + TLS + HTTP/2 — smaller win (~50-150 ms) than the WS variant (~200-500 ms) but still real on cold-start calls. **Billing safety**: ElevenLabs bills on synthesised characters delivered via `audio` frames (per [pricing](https://elevenlabs.io/pricing)) — the keepalive primer is the documented session-establishment frame and does NOT commit synthesis (no `flush: true`, no real text). Cartesia `GET /voices` is a free metadata read; Inworld `HEAD` does not invoke the synthesis pipeline. Tests in both SDKs explicitly assert no `flush: true`, no audio frames, and no synthesis POST during warmup. Files: `libraries/python/getpatter/providers/{elevenlabs_ws_tts,cartesia_tts,inworld_tts}.py`, `libraries/typescript/src/providers/{elevenlabs-ws-tts,cartesia-tts,inworld-tts}.ts`. + +- **OpenAI Realtime native warmup (`response.create` with `generate: false`)** — concrete `warmup()` override shipped on `OpenAIRealtimeAdapter`. Per OpenAI's documented [websocket-mode warmup pattern](https://developers.openai.com/api/docs/guides/websocket-mode), the canonical warm step on the Realtime API is to open a session and send `response.create` with `response.generate=false` — this prepares the model's request state and primes inference far more effectively than a generic HTTPS GET. Implementation: open the WS, wait for `session.created`, send `{"type": "response.create", "response": {"generate": false}}`, capture the `response.id` from the resulting `response.created` event into `self._prewarm_response_id` (Py) / `this.prewarmResponseId` (TS), then close. The id is stored so a future call can chain it as `previous_response_id` when the chaining path is wired through `connect()`; for now we capture-and-discard, taking half the win (priming the global session state) without the cross-session-state plumbing complexity. **Billing safety**: `response.create` with `generate: false` is documented as a no-token warmup variant that does not invoke the model — no per-token cost is accrued. Files: `libraries/python/getpatter/providers/openai_realtime.py`, `libraries/typescript/src/providers/openai-realtime.ts`. + +- **LLM HTTP-GET warmup retained but documented as low-impact** — the existing `OpenAILLMProvider.warmup` (and its Anthropic / Google / Cerebras / Groq subclasses) still issues a 5 s-bounded `GET /models` to warm DNS + TLS + connection pool, saving ~150-400 ms on cold start. The docstring on the base implementation now explicitly calls out that an HTTPS GET does NOT warm the inference path itself — for true inference warmup a real low-token request is needed, left as a follow-up. STT / TTS WebSocket prewarms (above) dominate the cold-start latency budget. Files: `libraries/python/getpatter/services/llm_loop.py`, `libraries/typescript/src/llm-loop.ts`. + +Tests: `libraries/python/tests/unit/test_provider_warmup.py` (14 unit tests — Deepgram / Cartesia / AssemblyAI / ElevenLabs WS / Cartesia TTS / Inworld / OpenAI Realtime each verified for: warmup completes, WS opened + closed, no audio / no synthesis-commit frames sent during warmup, errors swallowed at DEBUG) and `libraries/typescript/tests/unit/provider-warmup.mocked.test.ts` (15 parity tests). Tests are tagged `@pytest.mark.mocked` (Py) / filename `*.mocked.test.ts` (TS) per `.claude/rules/authentic-tests.md` — only the network boundary is mocked; protocol negotiation, URL construction, and the warmup logic itself run real code. Defaults preserved: `agent.prewarm` is still `true` by default, warmup remains a no-op for unmodified custom providers, and existing tests pass without modification. + +### Added — Pre-warm and pre-synth firstMessage (Python + TypeScript parity) + +Cold-start latency on the first turn of an outbound call is dominated by DNS / TLS / HTTP-keepalive handshakes against the LLM and TTS providers (typical: 200-700 ms TTS first-byte plus 150-400 ms LLM connection setup, on top of the carrier's 3-15 s ringing window). The `Agent.prewarm` and `Agent.prewarm_first_message` flags let `Patter.call(...)` reclaim that lost latency by working in parallel with the carrier-side `initiate_call`. + +- **`Agent.prewarm: bool = True`** (Python) / **`agent.prewarm?: boolean`** (TypeScript, default `true` when undefined). When `True`, `Patter.call` spawns a fire-and-forget task that invokes the optional `warmup()` method on the configured STT, TTS, and LLM providers in parallel via `asyncio.gather(..., return_exceptions=True)` (Python) / `Promise.allSettled` (TS). Built-in LLM providers ship a real warmup that issues a 5 s-bounded HTTPS `GET /models` to the upstream — OpenAI (`https://api.openai.com/v1/models`), Anthropic (`https://api.anthropic.com/v1/models`), Google (`https://generativelanguage.googleapis.com/v1beta/models`), Cerebras (`https://api.cerebras.ai/v1/models`), Groq (`https://api.groq.com/openai/v1/models`). STT and TTS providers inherit a no-op default; concrete providers can override `async warmup() -> None` (Python ABC) / `warmup?(): Promise` (TS interface) to prime their own connections. Failures are logged at DEBUG and never abort the call — the feature is pure latency optimisation. Files: `libraries/python/getpatter/providers/base.py` (default `warmup` on `STTProvider`, `TTSProvider`), `libraries/python/getpatter/services/llm_loop.py` (`OpenAILLMProvider.warmup`), `libraries/python/getpatter/providers/anthropic_llm.py`, `libraries/python/getpatter/providers/google_llm.py`, `libraries/python/getpatter/client.py` (`Patter._spawn_provider_warmup`), `libraries/typescript/src/llm-loop.ts` (`LLMProvider.warmup` optional + `OpenAILLMProvider.warmup`), `libraries/typescript/src/providers/cerebras-llm.ts`, `libraries/typescript/src/providers/groq-llm.ts`, `libraries/typescript/src/providers/anthropic-llm.ts`, `libraries/typescript/src/providers/google-llm.ts`, `libraries/typescript/src/provider-factory.ts` (`STTAdapter.warmup`, `TTSAdapter.warmup` optional), `libraries/typescript/src/client.ts` (`Patter.spawnProviderWarmup`). + +- **`Agent.prewarm_first_message: bool = False`** (Python) / **`agent.prewarmFirstMessage?: boolean = false`** (TypeScript). Off by default to preserve the prior cost surface. When `True`, after `Patter.call` resolves the carrier-issued `call_id` it spawns a background task that calls `agent.tts.synthesize(agent.first_message)` (Python) / `agent.tts.synthesizeStream(agent.firstMessage)` (TS), accumulates the bytes, and stores the buffer in `Patter._prewarm_audio[call_id]` / `Patter.prewarmAudio.set(callId, buffer)`. The synth is bounded by `ring_timeout` (default 25 s) so a never-answered call can't tie up the TTS connection. The per-call `StreamHandler` (`PipelineStreamHandler` Python / `StreamHandler.runPipeline` TS) now checks the cache via `pop_prewarm_audio(call_id)` / `popPrewarmAudio(callId)` at the start of the firstMessage emit; on a cache hit the buffer is sent directly through the carrier-side audio sender (which handles native-rate → carrier-rate resampling identically to the live TTS path), the `tts.synthesize` round-trip is skipped, and TTS first-byte latency drops to ~0 ms. **Cost implication**: the TTS bill for `agent.first_message` is paid as soon as the synth task completes, even when the call is never answered (no-answer / busy / AMD voicemail). When the call ends without consuming the cache, `Patter.end_call` / `Patter.endCall` log a WARN naming the wasted call_id and approximate byte count so operators see the cost surface explicitly. Files: `libraries/python/getpatter/client.py` (`Patter._prewarm_audio`, `pop_prewarm_audio`, `_record_prewarm_waste`, `_spawn_prewarm_first_message`), `libraries/python/getpatter/server.py` (`EmbeddedServer.pop_prewarm_audio` forward), `libraries/python/getpatter/telephony/twilio.py` + `telnyx.py` (bridge accepts `pop_prewarm_audio`), `libraries/python/getpatter/stream_handler.py` (`PipelineStreamHandler` consumes cache in firstMessage emit), `libraries/typescript/src/client.ts` (`Patter.prewarmAudio`, `popPrewarmAudio`, `recordPrewarmWaste`, `spawnPrewarmFirstMessage`), `libraries/typescript/src/server.ts` (`EmbeddedServer.popPrewarmAudio`), `libraries/typescript/src/stream-handler.ts` (`StreamHandlerDeps.popPrewarmAudio`, firstMessage emit consumes cache). + +Tests: `libraries/python/tests/test_prewarm.py` (14 unit tests covering default flag values, no-op default `warmup`, all-three-providers warmup invocation, opt-out via `prewarm=False`, exception swallow at DEBUG, cache populate / skip / empty-message / timeout, one-shot pop semantics, waste-warn log, StreamHandler cache-hit short-circuit + cache-miss live-TTS fallback) and `libraries/typescript/tests/unit/prewarm.test.ts` (11 parity tests). Both suites use authentic real code paths — only the network boundary is exercised through stubs — per `.claude/rules/authentic-tests.md`. Defaults preserved: `agent.prewarm` is `true` and warmup is a no-op for unmodified providers, so existing tests pass without modification; `agent.prewarm_first_message` is `false`, so the new TTS-bill cost surface is strictly opt-in. + +### Changed — Dashboard: STT and TTS rendered as separate cost rows + +The cost breakdown panel previously combined STT and TTS spend into a single "STT / TTS" line, which hid which side of the audio pipeline dominated cost. The two providers are typically distinct (e.g. Cartesia STT + ElevenLabs TTS) and bill at very different rates per second of audio. The panel now renders them as two adjacent rows labeled with the actual provider name (e.g. "Cartesia STT" / "ElevenLabs TTS"), driven by `record.metrics.stt_provider` / `tts_provider` already exposed by the backend. The legacy `CallCostUi.sttTts` field is kept in `dashboard-app/src/lib/mappers.ts` for the few aggregate-spend callers (`callSpend`, totals bar) and is now derived as `stt + tts` after both granular fields are populated. Files: `dashboard-app/src/lib/mappers.ts`, `dashboard-app/src/components/CostPanel.tsx`. + +### Changed — `stt_ms` is now finalization-only (Python + TypeScript parity) + +⚠️ Semantic change to `LatencyBreakdown.stt_ms`. Previously the value measured `stt_complete - turn_start`, which conflated user speech duration with STT processing — a 5 s utterance produced `stt_ms ≈ 5000` even when Cartesia / Deepgram finalized in 200 ms after end-of-speech. The legacy interpretation was misleading: industry benchmarks (Picovoice, Deepgram, Gladia, Speechmatics, Daily.co) all report STT latency as the **finalization window** — `final_transcript - end_of_speech` — independent of how long the user spoke. `stt_ms` now matches that definition: it measures from the endpoint signal (VAD `speech_stop` or STT `speech_final`, whichever comes first) to the final transcript delivery. When the endpoint signal is unavailable (degraded provider, batch STT) the metric falls back to the legacy `turn_start` anchor so dashboards never see a spuriously zero value. + +A new optional field `LatencyBreakdown.user_speech_duration_ms` (`userSpeechDurationMs` over the wire stays `user_speech_duration_ms` for SDK parity) carries the displaced "how long did the user speak" number, populated only when the endpoint signal is present. Together with the existing `agent_response_ms` (silence detection + LLM TTFT + TTS first-byte) and `total_ms` (turn_start → first agent audio byte), the breakdown now cleanly separates the four orthogonal slices a voice-AI dashboard needs: utterance length, STT finalization, LLM TTFT, TTS first-byte. Files: `libraries/python/getpatter/models.py`, `libraries/python/getpatter/services/metrics.py`, `libraries/typescript/src/metrics.ts`. + +### Added — OTel `patter.*` span attributes (Python only; TS parity follow-up) + +⚠️ Parity gap: this lands in the Python SDK only. TypeScript follow-up is tracked separately and will land in a subsequent release. Per `.claude/rules/sdk-parity.md` every public feature must reach both SDKs; this is a known time-boxed exception. + +- **`getpatter.observability.attributes`** — three new helpers added: `record_patter_attrs(attrs)`, `patter_call_scope(call_id, side)` (context manager), and `attach_span_exporter(patter, exporter, side)`. Lazy-OTel-guarded; no-op when the `[tracing]` extra is not installed. Two ContextVars (`patter.call_id`, `patter.side`) propagate through the asyncio task tree so spans emitted by deeply nested provider code inherit the active call's identity automatically. File: `libraries/python/getpatter/observability/attributes.py`. The three symbols are re-exported from `getpatter.observability` for direct import. +- **`Patter._attach_span_exporter(exporter, *, side="uut")`** — public-but-underscore hook for tools that observe Patter from outside (e.g. an out-of-process agent runner). Default `side="uut"` preserves all existing behaviour. The leading underscore signals it is not part of the customer-facing API surface. File: `libraries/python/getpatter/client.py`. +- **Per-provider cost emission (19 surfaces)** — `patter.cost.{telephony_minutes, stt_seconds, tts_chars, llm_input_tokens, llm_output_tokens, realtime_minutes}` are now stamped on the active span across the provider lineup (Twilio + Telnyx telephony adapters; Deepgram, AssemblyAI, Whisper, OpenAI Transcribe, Soniox, Speechmatics, Cartesia STT; ElevenLabs, OpenAI, Cartesia, LMNT, Rime TTS; OpenAI/Anthropic/Google/Groq/Cerebras LLM; OpenAI Realtime + ElevenLabs ConvAI realtime). Provider tag emitted alongside as `patter.{telephony,stt,tts,llm,realtime}.provider`. All call sites are wrapped in defensive `try/except` so observability cannot kill a live call. +- **Per-turn latency** — `patter.latency.{ttfb_ms, turn_ms}` stamped from `StreamHandler._emit_turn_metrics` via a new `PipelineHookExecutor.record_turn_latency(*, ttfb_ms, turn_ms)` method. `ttfb_ms` maps to `total_ms` (turn-start → first TTS audio byte, the user-perceptible TTFB); `turn_ms` maps to `tts_total_ms` and falls back to `total_ms` when null. Files: `libraries/python/getpatter/services/pipeline_hooks.py`, `libraries/python/getpatter/stream_handler.py`. +- **`patter_call_scope` enters at the bridge level** so the entire WebSocket bridge lifetime — including hangup / cleanup — is bound to `patter.call_id` and `patter.side`. The scope is opened on the Twilio `start` / Telnyx `streaming.started` event (when the call_id is known) and closed in the `finally:` block via `contextlib.ExitStack`, so cleanup-emitted spans (handler.cleanup, telephony cost queries, on_call_end) inherit the call identity. Files: `libraries/python/getpatter/telephony/twilio.py`, `libraries/python/getpatter/telephony/telnyx.py`. +- **`TwilioAdapter.record_call_end_cost` / `TelnyxAdapter.record_call_end_cost`** — adapter-level helpers used by the bridge to emit `patter.cost.telephony_minutes` once the call's wall-clock duration is known. Files: `libraries/python/getpatter/providers/twilio_adapter.py`, `libraries/python/getpatter/providers/telnyx_adapter.py`. +- **Docs**: `docs/python-sdk/tracing.mdx` updated with a new "Cost and latency attributes (`patter.*`)" section and an "Attach a custom exporter" example showing how to wire `Patter._attach_span_exporter` to an `InMemorySpanExporter` for tests or to an `OTLPSpanExporter` in production. + +### Added — Opt-in barge-in confirmation strategies (Python + TypeScript) + +- **Opt-in barge-in confirmation strategies** (Python + TypeScript parity, fully backward-compatible). Cloud TTS providers take 200-700 ms to emit the first audio byte and PSTN background noise routinely fires VAD before any real interruption is happening; the legacy "any VAD speech_start during TTS cancels the agent" contract therefore produced frequent false-positive cancels — the agent was cut by cough/click/HVAC/breath and lost the conversational thread. The new ``Agent.barge_in_strategies`` (Python) / ``agent.bargeInStrategies`` (TypeScript) tuple lets callers opt into a two-stage confirmation pipeline: VAD speech_start during TTS now marks the barge-in as *pending* (TTS keeps streaming naturally, the in-flight LLM stream is preserved), every STT transcript is fed to each configured strategy, and the first strategy that returns ``True`` cancels the agent and runs the existing flush sequence; if no strategy confirms within ``barge_in_confirm_ms`` (default 1500 ms) the pending state is dropped and the agent finishes its sentence. New module ``getpatter.services.barge_in_strategies`` exposes the ``BargeInStrategy`` Protocol, the ``MinWordsStrategy`` reference implementation (filters short backchannels — "okay", "uh-huh", "yeah" — by requiring N words while the agent is speaking and letting any single word through while the agent is silent), and ``evaluate_strategies`` / ``reset_strategies`` helpers with short-circuit-OR composition and per-strategy error isolation. TS twin in ``src/services/barge-in-strategies.ts``. Wiring lives in ``stream_handler.py`` ``_handle_barge_in`` / ``stream-handler.ts`` ``handleBargeIn`` — both keep the existing canBargeIn gate and only add the confirm step when at least one strategy is configured. Defaults preserved: ``barge_in_strategies=()`` matches the prior cancel-immediately behaviour byte-for-byte, so existing users see no change unless they opt in. New regressions: 14 unit tests for ``MinWordsStrategy`` + composition (Py); 15 parity tests (TS); 10 end-to-end tests covering pending lifecycle, confirmation, timeout, idempotency, and threshold parametrization (Py); 10 TS twins. Files: ``libraries/python/getpatter/services/barge_in_strategies.py``, ``libraries/python/getpatter/models.py``, ``libraries/python/getpatter/__init__.py``, ``libraries/python/getpatter/stream_handler.py``, ``libraries/typescript/src/services/barge-in-strategies.ts``, ``libraries/typescript/src/types.ts``, ``libraries/typescript/src/index.ts``, ``libraries/typescript/src/stream-handler.ts``. + +### Fixed + +- **Dashboard live-transcript: live pane now accumulates user/assistant lines across every turn** (TypeScript-only, frontend + backend, dashboard BUG 1). The live-transcript fallback in `dashboard-app/src/lib/mappers.ts` derived UI rows from `record.turns[]` (the `TurnMetrics` shape), but the primary mapper path checked `record.transcript.length > 0` — which was always empty for in-flight calls because the active record only carried `turns[]`. On every `turn_complete` SSE the pane re-rendered from a single source of truth that flickered between "fallback derived from one turn" and "primary path with empty transcript", producing the symptom that the most recent user/agent pair would replace the previously-rendered turn instead of appending. Fix: `MetricsStore.recordTurn` now mirrors each completed round-trip into a flat `transcript` array on the active record (one `{role:'user', text, timestamp}` entry when `user_text` is non-empty, one `{role:'assistant', text, timestamp}` entry when `agent_text` is non-empty and not the `[interrupted]` sentinel). The mapper's primary path therefore sees an accumulating `user → assistant → user → assistant → …` history live, identical in shape to what completed calls expose. Files: `libraries/typescript/src/dashboard/store.ts`. New regressions: `libraries/typescript/tests/dashboard-store.test.ts` — `recordTurn appends both user and assistant lines to active.transcript across turns` (3-turn round-trip; asserts 5 entries in the right order) and `recordTurn skips '[interrupted]' agent_text and empty user_text from active.transcript` (filters first-message/interrupted edge cases). + +- **Dashboard live-transcript: pane no longer goes blank in the carrier-statusCallback → recordCallEnd race window** (TypeScript-only, frontend + backend, dashboard BUG 2). The Twilio `statusCallback` for `CallStatus=completed` runs `MetricsStore.updateCallStatus(callId, 'completed', …)`, which moved the active record into the completed buffer WITHOUT preserving its running `turns[]` / `transcript[]`. The subsequent WS-driven `recordCallEnd` then overwrote the row in place — but in the race window between those two events the completed entry had no transcript, and any `useTranscript` fetch in that window cleared the live-pane render. Three coupled fixes: (1) `updateCallStatus` now copies `active.turns` and `active.transcript` into the new completed entry on the terminal-status branch; (2) `recordCallEnd` falls back to the running active/existing transcript when `data.transcript` is empty (e.g. `endCall` invoked without an authoritative history payload); (3) the `useTranscript` hook in `dashboard-app/src/hooks/useTranscript.ts` now subscribes to SSE `call_end` events (in addition to `turn_complete`) and refetches the call detail the moment `recordCallEnd` lands the SDK-authoritative `history.entries` transcript. Files: `libraries/typescript/src/dashboard/store.ts`, `dashboard-app/src/hooks/useTranscript.ts`. New regressions: `libraries/typescript/tests/dashboard-store.test.ts` — three new cases covering `updateCallStatus('completed')` carry-over, `recordCallEnd` running-transcript fallback when `data.transcript` is missing, and the explicit `data.transcript` taking precedence over the running fallback. + +- **Dashboard sparkline tooltip: per-card metric-specific aggregate (count / avg latency / total cost)** (TypeScript-only, frontend, dashboard BUG 4). Every metric card's hover tooltip showed the same generic "N call(s)" headline and a per-call sample list — so the spend card and the latency card were indistinguishable from the calls card. The tooltip now reports a metric-specific aggregate above the per-call sample list: `TOTAL COST $X.XXX` (sum of per-call cost in the bucket) for the spend card, `AVG LATENCY ms` (mean of per-call P95 in the bucket) for the latency card, and `N CALLS` for the count cards (existing behaviour, made explicit). Headline label uppercased, monospace, and styled to match the existing time-range header so the tooltip reads consistently with the rest of the site. New `MetricKind` type (`'count' | 'latency' | 'spend'`) drives the headline calculation in pure form via the new `bucketHeadline` export, callable from tests. Files: `dashboard-app/src/components/Metric.tsx`, `dashboard-app/src/App.tsx`, `dashboard-app/src/styles/dashboard.css`. + +- **Dashboard: outbound call disappeared from the recent-calls table after end** (Python + TypeScript parity, BUG C, behavioural fix). The Twilio `statusCallback` for `CallStatus=completed` arrives a moment before the WS `stop` frame and runs `update_call_status` / `updateCallStatus`, which already moves the row from `_active_calls` / `activeCalls` into the completed list. Shortly after, the WS-stop path runs `record_call_end` / `recordCallEnd` for the same call_id — but the active record is already gone, so the prior implementations appended a SECOND row with `started_at=0`, empty caller/callee, and freshly captured metrics. `MetricsStore.get_calls` returns newest-first and the dashboard SPA's `mergeCalls` keeps only the first match by call_id, so the older well-formed row was masked by the malformed duplicate; the duplicate's `startedAtMs=0` then dropped it out of the 24h time-range filter and the call vanished from the UI altogether. `record_call_end` / `recordCallEnd` now look up the existing entry in `_calls` / `calls[]` and update it in place (preserving caller/callee/started_at, merging in the just-collected metrics) instead of appending a duplicate. Files: `libraries/python/getpatter/dashboard/store.py`, `libraries/typescript/src/dashboard/store.ts`. New regressions: `libraries/python/tests/unit/test_dashboard_store_unit.py::TestRecordCallEndDeduplication` (2 tests — exercises the full `record_call_initiated → record_call_start → update_call_status → record_call_end` sequence and asserts (a) `call_count == 1` (no duplicate), (b) caller/callee/started_at preserved, (c) the call survives the 24h time-range filter); equivalent 2-test describe in `libraries/typescript/tests/dashboard-store.test.ts`. + +- **Dashboard: live transcript pane stayed empty during in-flight calls** (Python + TypeScript parity, BUG A, frontend + backend). Two coupled bugs hid streaming transcripts from the dashboard SPA while a call was in progress: (1) `GET /api/dashboard/calls/:callId` only consulted the completed-call buffer (`store.get_call` / `store.getCall`), returning 404 for any active call; the SPA's `useTranscript` hook polled this route every 2 s and rendered `[]` for the entire call lifetime. (2) The completed-call shape exposes `transcript: TranscriptEntry[]` while active records expose `turns: TurnMetrics[]` (the per-round-trip metrics shape), and `toUiTranscript` in `dashboard-app/src/lib/mappers.ts` only knew how to read `transcript`. Both routes (`/api/dashboard/calls/:callId` and the v1 `/api/v1/calls/:callId`) now fall through to `get_active` / `getActive` when the completed lookup misses, so the live record is reachable. `toUiTranscript` now falls back to `record.turns` when `record.transcript` is empty, deriving user/agent message rows from `user_text` / `agent_text` (skipping the sentinel `[interrupted]` turns). `useTranscript` additionally subscribes to `/api/dashboard/events` for `turn_complete` events filtered by `call_id` so new turns appear within ~50 ms of the round-trip ending — the existing 2 s polling stays in place as a backstop for SSE drops. Files: `libraries/python/getpatter/dashboard/routes.py`, `libraries/typescript/src/dashboard/routes.ts`, `dashboard-app/src/lib/mappers.ts`, `dashboard-app/src/hooks/useTranscript.ts`. New regression: `libraries/python/tests/unit/test_dashboard_store_unit.py::TestActiveCallDetail::test_get_active_returns_record_with_turns` (verifies the accessor the route now falls back to exposes the live turns). + +- **Dashboard logs: outbound calls persisted with empty `caller` / `callee` in `metadata.json`** (Python + TypeScript parity, BUG B). Inline TwiML for outbound calls (``) carries no `` tags and no query-string metadata, so the WS bridge's `caller` / `callee` are empty strings on outbound. The `_on_call_start` (Py) / `wrapLoggingCallbacks` (TS) wrappers passed those empty strings straight to `CallLogger.log_call_start` / `logCallStart`, and every outbound call's persisted `metadata.json` ended up with `caller=""` / `callee=""` even though the in-memory store had the correct numbers (populated at dial time by `record_call_initiated`). Wrappers now resolve caller/callee from the active store record when the bridge data is empty, so `metadata.json` is faithful to the dial. As a related parity fix: `record_call_start` (Py) was also clobbering existing caller/callee with the empty strings on the upgrade-from-initiated path; it now mirrors the existing TS behaviour and only overwrites when the new value is non-empty. Files: `libraries/python/getpatter/server.py`, `libraries/python/getpatter/dashboard/store.py`, `libraries/typescript/src/server.ts`. New regressions: `libraries/python/tests/unit/test_server_unit.py::TestWrapCallbacks::test_call_log_start_pulls_caller_from_active_record` (real `CallLogger` writes a real `metadata.json`; asserts the masked phone numbers end with the original last-4) and `libraries/typescript/tests/server.test.ts` (parity test in `EmbeddedServer wraps logging callbacks with active-record fallback`). + +- **Barge-in: `InterruptionMetrics.detection_delay_ms` corrupted to ~0 on strategy-confirmed cancel** (Python + TypeScript parity, FIX #88, fully backward-compatible). When `agent.barge_in_strategies` was non-empty, the two-stage barge-in flow stamped T1 via `record_overlap_start()` from `_start_pending_barge_in` (VAD speech_start) and then stamped T2 via a SECOND `record_overlap_start()` from `_do_cancel_for_barge_in` after the strategy confirmed — overwriting T1. The downstream `record_overlap_end()` therefore computed `T2 → now ≈ 0`, hiding the real ~150-500 ms VAD-to-confirm latency on every confirmed barge-in. The cancel path now captures the pending state BEFORE clearing it and skips the redundant `record_overlap_start()` when VAD already started the overlap window. Legacy path (`barge_in_strategies=()`, no VAD pending phase) is unchanged — `_do_cancel_for_barge_in` is still the sole caller of `record_overlap_start` there. New regressions in `libraries/python/tests/unit/test_barge_in_two_stage.py::TestBargeInOverlapStartPreserved` (3 tests including end-to-end via real `CallMetricsAccumulator` asserting detection_delay reflects the ~200 ms VAD→confirm window, not ~0) and `libraries/typescript/tests/unit/barge-in-two-stage.test.ts` (`StreamHandler — overlap window preserved across VAD → strategy confirm`, 2 tests). Files: `libraries/python/getpatter/stream_handler.py`, `libraries/typescript/src/stream-handler.ts`. + +- **Barge-in: leaked pending-confirmation task on call end** (Python + TypeScript parity, FIX #89, fully backward-compatible). `PipelineStreamHandler.cleanup` (Py) and `StreamHandler.handleStop` / `handleWsClose` (TS) tore down STT / TTS / remote adapters but never dropped the pending barge-in timeout. If a call ended while a barge-in was in pending-timeout state (waiting for strategy confirmation), the asyncio.Task / `setTimeout` remained scheduled and fired `record_overlap_end` / `recordOverlapEnd` on a finalised metrics object `barge_in_confirm_ms` later (default 1500 ms) — a slow leak in long-running servers and a race producing spurious overlap_end events on unrelated subsequent calls if the metrics object got GC'd and reused. Both SDKs now call `_clear_pending_barge_in` / `clearPendingBargeIn` at the top of cleanup, before any other tear-down. Idempotent: safe to call when no pending state exists, so the legacy non-strategy flow is unchanged byte-for-byte. New regressions in `libraries/python/tests/unit/test_barge_in_two_stage.py::TestCleanupClearsPendingBargeIn` (2 tests) and `libraries/typescript/tests/unit/barge-in-two-stage.test.ts` (`StreamHandler — handleStop / handleWsClose drops pending barge-in timer`, 2 tests). Files: `libraries/python/getpatter/stream_handler.py`, `libraries/typescript/src/stream-handler.ts`. + +- **Docs drift: `Agent.barge_in_strategies` docstring claimed "TTS is paused"** (Python + TypeScript parity, FIX #90). The docstring on `Agent.barge_in_strategies` (Python) / `agent.bargeInStrategies` (TypeScript) said "the agent's TTS is paused" while a barge-in was pending strategy confirmation, but the implementation does the opposite: TTS continues streaming naturally and only the strategy-confirmed cancel path stops TTS. Replaced with "the agent's TTS continues streaming naturally" so users opting into the confirm pipeline aren't surprised by uninterrupted audio during the pending window. Surgical text-only fix — no behaviour change. Files: `libraries/python/getpatter/models.py`, `libraries/typescript/src/types.ts`. + +- **Pipeline first-message prewarm: cached audio sent as a single multi-second buffer (cancel granularity lost)** (Python + TypeScript parity, FIX #97). On a prewarm cache hit, `PipelineStreamHandler.start` (Py) and `StreamHandler.initPipeline` (TS) called `audio_sender.send_audio(prewarm_bytes)` / `bridge.sendAudio(...)` with the full multi-second buffer in one shot, while the live TTS path streams 20-128 ms chunks paced by the upstream provider. A `send_clear` issued mid-buffer therefore had nothing to clear from Twilio's mark/clear bookkeeping, manifesting as "the agent keeps talking after barge-in" on the very first turn only. New private helper `_stream_prewarm_bytes` (Py) / `streamPrewarmBytes` (TS) splits the prewarm buffer into 1280-byte chunks (40 ms PCM16 @ 16 kHz mono — sized to mirror the smallest live-TTS boundary) and forwards each through the existing `audio_sender` / `bridge.sendAudio` so cancel granularity is identical regardless of cache hit vs miss. Same `_is_speaking` guard at every iteration so a barge-in mid-prewarm stops chunking immediately. New regressions: `libraries/python/tests/test_prewarm.py::test_stream_prewarm_bytes_chunks_buffer` + `test_stream_prewarm_bytes_stops_on_barge_in_mid_buffer` (2 tests) and `libraries/typescript/tests/unit/prewarm.test.ts` (`streamPrewarmBytes — chunked send for cancel granularity`, 2 tests). Both assert ≥100 `send_audio`/`bridge.sendAudio` calls for a 5-second buffer (vs 1 in the regression) and that the loop honours mid-prewarm `_is_speaking=False`. Files: `libraries/python/getpatter/stream_handler.py`, `libraries/typescript/src/stream-handler.ts`. + +- **OpenAI Realtime warmup: replaced billing-unsafe `response.create` with `session.update`** (Python + TypeScript parity). The earlier prewarm pass sent `{"type": "response.create", "response": {"generate": false}}` to "prime model state", but the `generate` field is NOT in the OpenAI Realtime API schema. Two failure modes were possible: (a) the server silently ignores the unknown field and invokes a real model response, billing tokens on every prewarm, or (b) the request is rejected with `invalid_request_error`, which makes the prewarm a no-op beyond TLS warm. Either way the prior implementation was wrong. The new flow opens the WS, waits for `session.created`, sends a single `session.update` whose body matches what `connect()` sends at production call-pickup (`input_audio_format`, `output_audio_format`, `voice`, `instructions`, `turn_detection`, `input_audio_transcription`, plus any opt-in fields populated on the adapter), waits for the matching `session.updated` ack, then closes cleanly. `session.update` only mutates session configuration — it does not invoke the model, does not consume any audio buffer, and does not trigger token generation, so the warmup is byte-for-byte billing-safe. The `_prewarm_response_id` (Py) / `prewarmResponseId` (TS) field is removed since it was only ever populated by the (broken) `response.create` path. New regression: `tests/unit/test_provider_warmup.py::test_openai_realtime_warmup_does_not_send_response_create` (Py) and the corresponding `does not send response.create on the wire` case in `tests/unit/provider-warmup.mocked.test.ts` (TS) — both fail loudly if a future change reintroduces `response.create` in the warmup path. Files: `libraries/python/getpatter/providers/openai_realtime.py`, `libraries/typescript/src/providers/openai-realtime.ts`. + +- **ElevenLabs WS warmup: BOS frame now byte-identical to live `synthesize` BOS** (Python + TypeScript parity). The live `synthesize()` / `synthesizeStream()` path attaches `voice_settings` and (when `auto_mode=False`) `generation_config` to the protocol-required BOS frame, but the warmup variant only sent `{"text": " "}`. Because ElevenLabs may instantiate a different per-session worker depending on the BOS configuration, the warmed worker could end up unrelated to the worker that handles the live request — defeating the edge-warm goal entirely. Both paths now share a single `_build_bos_frame` (Py) / `buildBosFrame()` (TS) helper, so the warmup BOS is byte-identical to the production BOS for any given adapter configuration. New regression: `tests/unit/test_provider_warmup.py::test_elevenlabs_ws_warmup_bos_frame_matches_live_synthesize` (Py) and `warmup BOS bytes are byte-identical to synthesizeStream BOS bytes (regression)` in `tests/unit/provider-warmup.mocked.test.ts` (TS) — both capture the BOS bytes in each path and assert byte-equality. Files: `libraries/python/getpatter/providers/elevenlabs_ws_tts.py`, `libraries/typescript/src/providers/elevenlabs-ws-tts.ts`. + +- **Inworld TTS warmup: replaced HEAD against POST-only endpoint with `GET /tts/v1/voices`** (Python + TypeScript parity). The earlier warmup issued `HEAD https://api.inworld.ai/tts/v1/voice:stream`, but that endpoint is POST-only — Inworld returned `405 Method Not Allowed` on every call, completing the TLS handshake but spamming 405s into Inworld's audit logs and into our own logs. The new path issues `GET /tts/v1/voices` (a documented free metadata read that returns the configured voice catalogue) so the response is 2xx-clean. Billing surface is unchanged — the synthesis pipeline is invoked only by `POST /tts/v1/voice:stream` with non-empty `text`. Tests assert the URL targets `/tts/v1/voices` and the response status is 2xx, with explicit asserts that the warmup does NOT target `voice:stream` and does NOT use HEAD. Files: `libraries/python/getpatter/providers/inworld_tts.py`, `libraries/typescript/src/providers/inworld-tts.ts`. + +- **Cartesia STT + AssemblyAI STT warmup: API key no longer leaks into logs on handshake failure** (Python + TypeScript parity, security). Both providers authenticate via a query-string parameter on the WS upgrade URL (Cartesia: `?api_key=...`, AssemblyAI optional: `?token=...`). When the WS handshake failed (e.g. 401 from a rotated key), `aiohttp.WSServerHandshakeError.__str__` (Py) and the `ws` library `Error.message` (TS) typically include the full request URL — and `logger.debug("warmup failed: %s", exc)` therefore wrote the API key straight into application logs. The fix catches the handshake-error class specifically before the generic `Exception` handler and logs only the HTTP status code (or, for non-handshake errors, just the exception class name) — never the full message or URL. New regression tests in both SDKs install a custom logger / `caplog` capture, force a 401 handshake error during warmup, and assert (a) the API key never appears in any captured log message, (b) the URL with `?api_key=` / `?token=` never appears either, and (c) the status code is still surfaced so operators see why the warmup failed. Files: `libraries/python/getpatter/providers/{cartesia_stt,assemblyai_stt}.py`, `libraries/typescript/src/providers/{cartesia-stt,assemblyai-stt}.ts`. + +- **Dashboard hydrate: hydrated calls no longer lose `cost` and `latency`** (Python + TypeScript parity, fully backward-compatible). `CallLogger.log_call_end` writes `cost`, `latency`, `duration_ms`, and `telephony_provider` as **top-level keys** of `metadata.json`, but `MetricsStore.hydrate` (`libraries/python/getpatter/dashboard/store.py:535`, `libraries/typescript/src/dashboard/store.ts:421-424`) read them only from `meta.metrics.cost` / `meta.metrics.latency`. Result: every call rebuilt from disk landed in the store with `metrics=null`, so the local dashboard rendered `$0.00` and `—` for cost/latency on all hydrated rows; only the in-flight call (which never goes through hydrate) showed real numbers. Caught during 0.6.0 acceptance against `releases/0.6.0/typescript/matrix/outbound-cartesia-cerebras-elevenlabs.ts` — 48 of 49 calls in the dashboard had blank P95/cost columns. Fix promotes the top-level fields into a synthesized `metrics` dict (`metrics_from_top_level` / `metricsFromTopLevel`) when `meta.metrics` is missing, mapping `latency.p95_ms` → `metrics.latency_avg.total_ms` so the existing UI fields populate. Explicit `meta.metrics` (legacy/future shape) is preserved untouched. New regressions: `tests/unit/test_metrics_store_hydrate.py::test_hydrate_lifts_top_level_cost_and_latency_into_metrics` + `test_hydrate_preserves_explicit_metrics_when_present` (Py); two `MetricsStore.hydrate` cases in `tests/dashboard-store.test.ts` (TS). Files: `libraries/python/getpatter/dashboard/store.py`, `libraries/typescript/src/dashboard/store.ts`. + +- **Pipeline early barge-in: VAD self-cancellation before TTS first byte arrived** (Python + TypeScript parity, behavioural change for pipeline mode). Cloud TTS providers (ElevenLabs, Cartesia, …) take 200–700 ms to emit the first audio byte. The barge-in anti-flicker gate was anchored on `_speaking_started_at` / `speakingStartedAt` (set inside `_begin_speaking` / `beginSpeaking`), so a 250 ms gate without AEC expired BEFORE TTS produced any audio. VAD then picked up background noise, room ambience, or a "hello?" from the operator and triggered `[VAD] speech_start during TTS → BARGE-IN` → `cancelSpeaking` → `isSpeaking=false` → the `for await (chunk of tts.synthesizeStream(...))` loop exited at `if (!this.isSpeaking) break`, emitting **zero bytes**. From the SDK's perspective the agent "spoke" the first message; from the caller's perspective the line went silent until the next turn. Reproduced on `outbound-cartesia-cerebras-elevenlabs.ts` (call CAfca9c23b22144d4b1bb8ee737dd24016, 47 s — first message never reached the wire). Fix introduces `_first_audio_sent_at` / `firstAudioSentAt`, set in a new `_mark_first_audio_sent` / `markFirstAudioSent` helper invoked AFTER `audio_sender.send_audio` / `bridge.sendAudio` succeeds at all four pipeline emit sites (firstMessage, streaming response, regular response, WebSocket remote). `_can_barge_in` / `canBargeIn` now refuses to open the gate while `_first_audio_sent_at` is null — VAD speech_start before the first wire-time byte is suppressed regardless of how much wall-clock has elapsed since `_begin_speaking`. The 250 ms / 1000 ms gate values are unchanged — only the anchor moves. New regressions: `tests/unit/test_stream_handler_unit.py::test_barge_in_suppressed_before_first_audio_emitted` (Py); `canBargeIn() false before the first TTS chunk has hit the wire` in `tests/unit/stream-handler.test.ts` (TS). Existing `_handle_barge_in` / `handleBargeIn` tests updated to set both timestamps to reflect the new contract. Files: `libraries/python/getpatter/stream_handler.py`, `libraries/typescript/src/stream-handler.ts`. + ## 0.6.0 (2026-05-08) ### Fixed @@ -66,7 +818,7 @@ Defaults unchanged (`gpt-realtime-mini`, `whisper-1`). 7 unit tests Py + 4 unit tests TS covering enum exposure, constructor option storage, `reasoning.effort` wire-format injection (and its absence when unset). Files touched: `libraries/python/getpatter/providers/openai_realtime.py`, `libraries/typescript/src/providers/openai-realtime.ts`, plus the corresponding test files. -- **Speech-edge events for turn-taking instrumentation** (Python + TypeScript parity, additive — no breaking changes). Patter now exposes seven optional async callbacks on every `Patter` instance plus a read-only `conversation_state` (Py) / `conversationState` (TS) snapshot, mirroring the public APIs of LiveKit Agents (`user_state_changed`, `agent_state_changed`, `user_turn_completed`, `user_interruption_detected`), Pipecat (`VADUserStartedSpeakingFrame`, `BotStartedSpeakingFrame`, `LLMFullResponseStartFrame`, `OutputAudioRawFrame`, `InterruptionFrame`) and OpenAI Realtime (`input_audio_buffer.speech_started/_stopped/_committed`). The seven events: `on_user_speech_started` (raw VAD positive edge), `on_user_speech_ended` (raw VAD trailing edge — *not* end-of-utterance), `on_user_speech_eos` (committed EOU — VAD edge + trailing silence + optional semantic turn-detector agreement; the canonical "user finished" signal that anchors `eos_to_first_token_ms`), `on_agent_speech_started` (first wire-time chunk of the agent turn — what the user actually hears, distinct from TTS warmup), `on_agent_speech_ended` (last wire chunk; payload includes `interrupted: bool` for barge-in), `on_llm_token` (TTFT marker, fires once per turn on the first LLM token), `on_audio_out` (first TTS audio chunk per turn — TTS warmup, distinct from wire-time). Each event also records an OpenTelemetry span event on the current call span (`patter.event.user_speech_started`, …, `patter.event.llm_first_token` carrying `gen_ai.request.model` + `gen_ai.provider.name` per the OTel GenAI semconv) when `PATTER_OTEL_ENABLED=1` and the `opentelemetry` peer dep is installed; otherwise the OTel branch is a zero-cost no-op. The dispatcher is callback-safe — observer exceptions are caught and logged, never propagated to the live call. State machine tracks per-side `conversation_state` (`user`: `listening`/`speaking`/`thinking`/`away`, `agent`: `initializing`/`idle`/`listening`/`thinking`/`speaking`) and a monotonically-increasing `turn_idx` that increments on every committed EOU. Wired into the realtime stream handler so `user_speech_started/_ended/_eos` and `agent_speech_started/_ended` fire automatically on the OpenAI Realtime + Twilio/Telnyx path; `on_llm_token` and `on_audio_out` are exposed on the dispatcher for adapter / pipeline-mode integrations to call. New files: `libraries/python/getpatter/_speech_events.py`, `libraries/typescript/src/_speech-events.ts`. Public exports: `SpeechEvents`, `SpeechEventCallback`, `ConversationStateSnapshot`, `UserState`, `AgentState`, `EouTrigger`. 16 unit tests Py + 15 unit tests TS covering every event payload, idempotency (LLM/audio fire-once-per-turn), state transitions, OTel attach contract, callback-exception isolation, chained-callback wrapping, and Patter-level proxy mirroring. Motivated by the `patter-agent-runner` acceptance suite which ships 15 turn-taking assertion verbs (barge-in latency, silence-gap, cross-talk, eos-to-first-token, MOS, WER) that previously auto-skipped because the SDK did not surface per-side speech edges. +- **Speech-edge events for turn-taking instrumentation** (Python + TypeScript parity, additive — no breaking changes). Patter now exposes seven optional async callbacks on every `Patter` instance plus a read-only `conversation_state` (Py) / `conversationState` (TS) snapshot, covering the standard voice-agent metric set (user/agent state transitions, turn boundaries, TTFT, audio first-byte) and aligning with OpenAI Realtime (`input_audio_buffer.speech_started/_stopped/_committed`) where applicable. The seven events: `on_user_speech_started` (raw VAD positive edge), `on_user_speech_ended` (raw VAD trailing edge — *not* end-of-utterance), `on_user_speech_eos` (committed EOU — VAD edge + trailing silence + optional semantic turn-detector agreement; the canonical "user finished" signal that anchors `eos_to_first_token_ms`), `on_agent_speech_started` (first wire-time chunk of the agent turn — what the user actually hears, distinct from TTS warmup), `on_agent_speech_ended` (last wire chunk; payload includes `interrupted: bool` for barge-in), `on_llm_token` (TTFT marker, fires once per turn on the first LLM token), `on_audio_out` (first TTS audio chunk per turn — TTS warmup, distinct from wire-time). Each event also records an OpenTelemetry span event on the current call span (`patter.event.user_speech_started`, …, `patter.event.llm_first_token` carrying `gen_ai.request.model` + `gen_ai.provider.name` per the OTel GenAI semconv) when `PATTER_OTEL_ENABLED=1` and the `opentelemetry` peer dep is installed; otherwise the OTel branch is a zero-cost no-op. The dispatcher is callback-safe — observer exceptions are caught and logged, never propagated to the live call. State machine tracks per-side `conversation_state` (`user`: `listening`/`speaking`/`thinking`/`away`, `agent`: `initializing`/`idle`/`listening`/`thinking`/`speaking`) and a monotonically-increasing `turn_idx` that increments on every committed EOU. Wired into the realtime stream handler so `user_speech_started/_ended/_eos` and `agent_speech_started/_ended` fire automatically on the OpenAI Realtime + Twilio/Telnyx path; `on_llm_token` and `on_audio_out` are exposed on the dispatcher for adapter / pipeline-mode integrations to call. New files: `libraries/python/getpatter/_speech_events.py`, `libraries/typescript/src/_speech-events.ts`. Public exports: `SpeechEvents`, `SpeechEventCallback`, `ConversationStateSnapshot`, `UserState`, `AgentState`, `EouTrigger`. 16 unit tests Py + 15 unit tests TS covering every event payload, idempotency (LLM/audio fire-once-per-turn), state transitions, OTel attach contract, callback-exception isolation, chained-callback wrapping, and Patter-level proxy mirroring. Motivated by the `patter-agent-runner` acceptance suite which ships 15 turn-taking assertion verbs (barge-in latency, silence-gap, cross-talk, eos-to-first-token, MOS, WER) that previously auto-skipped because the SDK did not surface per-side speech edges. - **Inworld TTS provider (`inworld-tts-2` + TTS-1.5 family)** (Python + TypeScript parity). New TTS adapter calling Inworld's HTTP NDJSON streaming endpoint `POST https://api.inworld.ai/tts/v1/voice:stream`. Default model is `inworld-tts-2` (sub-200 ms time-to-first-audio, 100+ languages with mid-utterance switching, natural-language voice steering); pass `model: "inworld-tts-1.5-max"` to fall back to the prior generation. Default audio output is `PCM` (PCM_S16LE) at 16 kHz so the result feeds straight into the Patter pipeline without transcoding. Public API: `import { InworldTTS } from "getpatter"` (TS) / `from getpatter import InworldTTS` (Py); pipeline-mode namespace `getpatter/tts/inworld` (TS) / `getpatter.tts.inworld` (Py) with env-var auto-resolve via `INWORLD_API_KEY`. Optional fields: `language` (BCP-47), `temperature` (TTS-1.5 only), `speakingRate` (0.5–1.5), `deliveryMode` (`EXPRESSIVE` / `BALANCED` / `STABLE` — TTS-2 only), `bitrate`. The Inworld dashboard issues a Base64 token already in the form expected by the `Authorization: Basic ` header — paste it as `INWORLD_API_KEY` directly. Pricing entry added to `pricing.ts` / `pricing.py` as `inworld` (placeholder $0.020 / 1k chars — verify against current platform tier). Optional dependency: `getpatter[inworld]` adds `aiohttp>=3.10`. New files: `libraries/typescript/src/providers/inworld-tts.ts`, `libraries/typescript/src/tts/inworld.ts`, `libraries/python/getpatter/providers/inworld_tts.py`, `libraries/python/getpatter/tts/inworld.py`. 7 unit tests per SDK covering payload shape, NDJSON parsing, base64 audio decoding, optional-field omission, env-var fallback, and non-200 error surfacing. @@ -221,7 +973,7 @@ Migration: if your code did `from getpatter.handlers.twilio_handler import ...` ### Cleanup -- All competitor license headers (LiveKit, Pipecat, Apache, etc.) removed from source files. New rule `.claude/rules/no-competitor-references.md` codifies the policy. +- All external license headers removed from source files. New rule `.claude/rules/no-competitor-references.md` codifies the policy. - Root `LICENSE` updated to `Copyright (c) 2026 Patter Contributors`. - `Dockerfile` + `docker-compose.yml` simplified; non-public-repo scripts removed. - `playwright.config.ts` + `@playwright/test` devDep dropped (E2E lives in downstream test repo). @@ -233,10 +985,10 @@ Latency-pass 1: TTFA optimisations grounded in the ElevenLabs latency posts and ### Improved — sentence chunker - **Italian abbreviations** added to the prefix list (Sig, Sgr, Dott, Prof, Avv, Ing, Geom, Rag, Arch, On, Egr, Spett, Gent, Ill) and the suffix list (ecc, cit, cap, sez, art, pag, fig, tab, cfr, vol, ed). Sentences like _"Ho incontrato il Sig. Rossi alla riunione di stamattina."_ are no longer split on the abbreviation period. -- **English abbreviations** expanded with the Pipecat NLTK Punkt set: `Gen.`, `Sen.`, `Rep.`, `Lt.`, `Cpt.`, `Capt.`, `Col.`, `Cmdr.`, `Adm.`, `vs.`, `etc.`, `No.`, `Vol.`, `pp.`, `cf.`, `ca.`, `op.`, plus address forms `Mt.`, `Hwy.`, `Rt.`, `Pl.`, `Ave.`, `Blvd.`, `Sq.`. Phrases like _"Compare A vs. B"_ and _"Met Gen. Smith and Sen. Davis"_ no longer split mid-abbreviation. +- **English abbreviations** expanded with the standard NLTK Punkt abbreviation set: `Gen.`, `Sen.`, `Rep.`, `Lt.`, `Cpt.`, `Capt.`, `Col.`, `Cmdr.`, `Adm.`, `vs.`, `etc.`, `No.`, `Vol.`, `pp.`, `cf.`, `ca.`, `op.`, plus address forms `Mt.`, `Hwy.`, `Rt.`, `Pl.`, `Ave.`, `Blvd.`, `Sq.`. Phrases like _"Compare A vs. B"_ and _"Met Gen. Smith and Sen. Davis"_ no longer split mid-abbreviation. - **Suffix + starter pattern preserves the period** (e.g. _"Patter Inc. He left."_ now keeps `Inc.` in the emitted sentence — previously dropped to `Inc`). -- **All-caps name flush fixed** (Pipecat issue #1692). Previously the gate-5 acronym guard blocked *any* uppercase-preceded period, so _"I was speaking with RAMESH."_ would sit in the buffer forever. Now only purely-uppercase ASCII words ≤3 chars (U, US, USA, NATO patterns) are treated as acronyms. -- **Multilingual terminator support**. The terminator set now includes ASCII semicolon `;`, Unicode ellipsis `…`, full-width semicolon `；`, full-width period `．`, half-width Japanese period `｡`, plus the Pipecat-derived non-Latin set: Hindi/Devanagari `। ॥`, Arabic `؟ ؛ ۔ ؏`, Armenian `։`, Ethiopic `። ፧`, Khmer `។ ៕`, Burmese `။`, Tibetan `༎ ༏`. Hindi text like _"यह हिन्दी का एक वाक्य है।"_ now flushes correctly. +- **All-caps name flush fixed**. Previously the gate-5 acronym guard blocked *any* uppercase-preceded period, so _"I was speaking with RAMESH."_ would sit in the buffer forever. Now only purely-uppercase ASCII words ≤3 chars (U, US, USA, NATO patterns) are treated as acronyms. +- **Multilingual terminator support**. The terminator set now includes ASCII semicolon `;`, Unicode ellipsis `…`, full-width semicolon `；`, full-width period `．`, half-width Japanese period `｡`, plus the non-Latin terminator set: Hindi/Devanagari `। ॥`, Arabic `؟ ؛ ۔ ؏`, Armenian `։`, Ethiopic `። ፧`, Khmer `។ ៕`, Burmese `။`, Tibetan `༎ ༏`. Hindi text like _"यह हिन्दी का एक वाक्य है।"_ now flushes correctly. - **Cross-SDK parity fixture** at `tests/parity/scenarios/sentence_chunker.json` — 61 cases covering EN/IT/CJK/Hindi/Arabic punctuation, decimals, abbreviations, currency, dates, ellipsis, JSON, lists, all-caps names. Standalone runner at `tests/parity/sentence_chunker_parity.py` verifies Python and TypeScript emit identical sentence streams (53 / 61 PASS, 8 documented quirks/regressions). ### Added — opt-in aggressive first-clause flush @@ -349,7 +1101,7 @@ Cost-accuracy, audio-pipeline, and observability hardening across both SDKs, plu ### Deferred to 0.6.0 (tracked) - **Per-model OpenAI Realtime pricing map**: default rates are calibrated for `gpt-4o-mini-realtime-preview`. Users on `gpt-realtime` (~3×) or `gpt-4o-realtime-preview` (~10×) still see under-reported cost. Startup warn (from 0.5.5) is the stopgap. - **Native `ulaw_8000` negotiation per provider when target is Twilio** — ElevenLabs, LMNT, Cartesia, Rime all accept `ulaw_8000` output format natively. Today we fall through a resample-then-mulaw chain that introduces aliasing. Switching to native negotiation per the ElevenLabs Twilio cookbook is the canonical fix. -- **Replace 5-tap binomial FIR with Kaiser-windowed half-band (31-tap)** — industry stopband is 60-80 dB; our binomial is ~20 dB. `soxr` (LiveKit default) or `scipy.signal.resample_poly` if available. +- **Replace 5-tap binomial FIR with Kaiser-windowed half-band (31-tap)** — industry stopband is 60-80 dB; our binomial is ~20 dB. `soxr` or `scipy.signal.resample_poly` if available. - **LLM pipeline token tracking** — `anthropic`, `groq`, `cerebras`, `google`, `openai` LLM adapters report latency but never emit token usage. Pipeline-mode `CostBreakdown.llm` is always $0, regardless of actual spend. New `record_llm_usage()` + per-model pricing entries. - **TS Telnyx outbound wrong codec** — TS `encodePipelineAudio` and `handleAdapterEvent` ship PCM16 16k to Telnyx that negotiated PCMU 8k. Telnyx customers see broken audio. Requires a `TelephonyBridge.encodeAudio` abstraction parity with Python's `TelnyxAudioSender`. - **TS OpenAI Realtime missing `audioFormat` parameter** — Python has it. Blocks TS Telnyx+Realtime. @@ -377,7 +1129,7 @@ Cost-accuracy, audio-pipeline, and observability hardening across both SDKs, plu ### Security - **`agent.model` was interpolated into warn logs without sanitisation** — dev-supplied string with ANSI escapes could inject colour codes into log aggregators. Now passes through `sanitizeLogValue`. -### Added — observability (LiveKit/Pipecat-style) +### Added — observability - `CallMetrics.latency_p50` and `.latency_p99` alongside `latency_p95` and `latency_avg`. Lets dashboards show the full distribution (typical UX / SLA / cold-start outlier). - `CostBreakdown.llm_cached_savings` as described above. - Percentile formula upgraded from `floor(n*p)` (returned max for n<21) to Hyndman-Fan type 7 linear interpolation (same as `numpy.percentile` default). Meaningful on 2-3 sample sets. @@ -411,7 +1163,7 @@ Users running non-default Realtime models (`gpt-realtime`, `gpt-4o-realtime-prev - **p95/p99 returned the sample maximum for any n < 21** — the previous `floor(n * 0.95)` formula was numerically meaningless on short calls. Replaced with linear interpolation between order statistics (Hyndman-Fan type 7, same as `numpy.percentile` default). Both SDKs. - **`firstMessage` latency wasn't measured in Python** (TS measured it for pipeline + realtime). Python now emits a turn-level metric for the first greeting in both modes. -### Added — observability (LiveKit/Pipecat-style) +### Added — observability - `CallMetrics` now exposes `latency_p50` and `latency_p99` alongside `latency_p95` and `latency_avg`. Useful to detect cold-start outliers (p99) and typical UX latency (p50). 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"MIT", + "dependencies": { + "siginfo": "^2.0.0", + "stackback": "0.0.2" + }, + "bin": { + "why-is-node-running": "cli.js" + }, + "engines": { + "node": ">=8" + } + }, "node_modules/yallist": { "version": "3.1.1", "resolved": "https://registry.npmjs.org/yallist/-/yallist-3.1.1.tgz", diff --git a/dashboard-app/package.json b/dashboard-app/package.json index c065f629..20cd78ba 100644 --- a/dashboard-app/package.json +++ b/dashboard-app/package.json @@ -9,6 +9,7 @@ "build": "tsc --noEmit && vite build", "preview": "vite preview", "lint": "tsc --noEmit", + "test": "vitest run", "sync": "node ./scripts/sync.mjs" }, "dependencies": { @@ -21,6 +22,7 @@ "@vitejs/plugin-react": "^4.3.4", "typescript": "^5.6.3", "vite": "^5.4.11", - "vite-plugin-singlefile": "^2.0.3" + "vite-plugin-singlefile": "^2.0.3", + "vitest": "^2.1.4" } } diff --git a/dashboard-app/src/App.tsx b/dashboard-app/src/App.tsx index 98f88632..dd21ea6e 100644 --- a/dashboard-app/src/App.tsx +++ b/dashboard-app/src/App.tsx @@ -3,10 +3,13 @@ import { Topbar } from './components/Topbar'; import { PageHeader } from './components/PageHeader'; import { Metric, type MetricBucket } from './components/Metric'; import { CallTable, type Call } from './components/CallTable'; +import { fmtCostUSD } from './components/format'; import { LiveCallPanel } from './components/LiveCallPanel'; import { MetricsPanel } from './components/MetricsPanel'; import { useDashboardData } from './hooks/useDashboardData'; import { useTranscript } from './hooks/useTranscript'; +import { useUiPrefs } from './hooks/useUiPrefs'; +import { deleteCalls } from './lib/api'; import { bucketStrategyForRange, computeSparkline, @@ -53,7 +56,9 @@ function pickPhoneNumber(calls: readonly Call[]): string { } export function App() { - const { calls, aggregates, isStreaming, error, refresh } = useDashboardData(); + const { calls, aggregates, isStreaming, error, refresh, removeCallsLocal } = + useDashboardData(); + const { revealed, dark, toggleRevealed, toggleDark } = useUiPrefs(); const [selectedId, setSelectedId] = useState(null); const [search, setSearch] = useState(''); const [range, setRange] = useState('24h'); @@ -156,6 +161,22 @@ export function App() { refresh().catch(() => undefined); }; + const handleDeleteCalls = async (ids: readonly string[]): Promise => { + if (ids.length === 0) return; + // Optimistic local removal so the rows vanish immediately. The SSE + // ``calls_deleted`` event triggers a refresh shortly after; if the + // server rejects an id (race with a status callback), it re-appears. + removeCallsLocal(ids); + if (ids.includes(selectedId ?? '')) setSelectedId(null); + try { + await deleteCalls(ids); + } catch { + // Pull the authoritative snapshot back if the call failed so the UI + // is consistent with the server's view. + await refresh().catch(() => undefined); + } + }; + return ( <>
setRange(r as RangeKey)} /> @@ -174,6 +199,7 @@ export function App() { spark={sparkTotalCalls.heights} buckets={toBuckets(sparkTotalCalls)} onSelectCall={setSelectedId} + kind="count" />
@@ -210,6 +239,8 @@ export function App() { newId={null} search={search} setSearch={setSearch} + onDeleteCalls={handleDeleteCalls} + revealed={revealed} />
diff --git a/dashboard-app/src/components/CallTable.tsx b/dashboard-app/src/components/CallTable.tsx index 9617d2b4..c78462fd 100644 --- a/dashboard-app/src/components/CallTable.tsx +++ b/dashboard-app/src/components/CallTable.tsx @@ -1,10 +1,20 @@ -import { useMemo } from 'react'; -import { fmtDuration, fmtPhone } from './format'; -import { IconArrowDown, IconArrowUp, IconSearch } from './icons'; +import { useMemo, useState } from 'react'; +import { fmtDuration, fmtPhone, fmtCostUSD } from './format'; +import { + IconArrowDown, + IconArrowUp, + IconCheck, + IconSearch, + IconTrash, + IconX, +} from './icons'; export interface CallCost { telco?: number; llm?: number; + stt?: number; + tts?: number; + /** @deprecated Sum of stt+tts kept for legacy aggregate-spend callers. */ sttTts?: number; cached?: number; total?: number; @@ -25,12 +35,26 @@ export interface Call { duration?: number; latencyP95?: number; latencyP50?: number; + /** avg(llm_ms) across this call's turns — for the waterfall llm bar. */ + llmAvg?: number; sttAvg?: number; ttsAvg?: number; + /** Number of completed turns. p50/p95 are statistically meaningful only when this is >= 5. */ + turnCount?: number; + /** p50 of agent_response_ms (wait time after user stops speaking) — user-perceived latency. */ + agentResponseP50?: number; + /** p95 of agent_response_ms — user-perceived latency outlier. */ + agentResponseP95?: number; cost: CallCost; agent?: string; model?: string; mode?: CallMode; + sttProvider?: string; + ttsProvider?: string; + /** Model identifier within the provider, e.g. "ink-whisper", "eleven_flash_v2_5", "gpt-oss-120b". */ + sttModel?: string; + ttsModel?: string; + llmModel?: string; transcriptKey?: string; endedAgo?: number; } @@ -40,9 +64,21 @@ interface CallRowProps { isSelected: boolean; onSelect: () => void; isNew: boolean; + isChecked: boolean; + /** ``null`` when the row cannot be checked (live calls). */ + onToggleCheck: ((event: React.MouseEvent) => void) | null; + revealed: boolean; } -function CallRow({ call, isSelected, onSelect, isNew }: CallRowProps) { +function CallRow({ + call, + isSelected, + onSelect, + isNew, + isChecked, + onToggleCheck, + revealed, +}: CallRowProps) { const dur = call.status === 'live' && call.durationStart ? fmtDuration((Date.now() - call.durationStart) / 1000) @@ -59,9 +95,47 @@ function CallRow({ call, isSelected, onSelect, isNew }: CallRowProps) { return ( + { + // The whole cell is the hit area to forgive imprecise clicks. + e.stopPropagation(); + if (onToggleCheck) onToggleCheck(e); + }} + aria-disabled={onToggleCheck === null} + > + + {call.status} @@ -75,8 +149,8 @@ function CallRow({ call, isSelected, onSelect, isNew }: CallRowProps) { > {call.direction === 'inbound' ? : } - - {fmtPhone(call.from)} → {fmtPhone(call.to)} + + {fmtPhone(call.from, revealed)} → {fmtPhone(call.to, revealed)} @@ -98,7 +172,7 @@ function CallRow({ call, isSelected, onSelect, isNew }: CallRowProps) { '—' )} - ${totalCost.toFixed(2)} + {fmtCostUSD(totalCost)} ); } @@ -110,6 +184,15 @@ export interface CallTableProps { newId: string | null; search: string; setSearch: (s: string) => void; + /** + * Confirmed deletion handler. The component owns the per-row checkbox + * state internally; ``onDeleteCalls`` is invoked with the ids the user + * confirmed in the bulk-action bar. Live ids are filtered out before + * this handler is called. + */ + onDeleteCalls?: (ids: readonly string[]) => Promise | void; + /** When ``false`` phone numbers are masked client-side (eye-OFF). */ + revealed: boolean; } export function CallTable({ @@ -119,6 +202,8 @@ export function CallTable({ newId, search, setSearch, + onDeleteCalls, + revealed, }: CallTableProps) { const filtered = useMemo(() => { if (!search.trim()) return calls; @@ -133,6 +218,66 @@ export function CallTable({ ); }, [calls, search]); + // Multi-select state lives in the table — App.tsx doesn't need it for any + // other reason. Persisting across filter changes is fine: a checked id + // that scrolls out of the visible list re-appears checked when it + // returns, matching how Gmail / Linear / Notion handle bulk-select. + const [checked, setChecked] = useState>(new Set()); + const [confirming, setConfirming] = useState(false); + const [busy, setBusy] = useState(false); + + // Ids the user CAN delete in the current visible window (live rows are + // excluded — the server would skip them anyway, but doing it client-side + // keeps the counter honest). + const deletableIds = useMemo( + () => filtered.filter((c) => c.status !== 'live').map((c) => c.id), + [filtered], + ); + const checkedDeletable = useMemo( + () => deletableIds.filter((id) => checked.has(id)), + [deletableIds, checked], + ); + const allDeletableChecked = + deletableIds.length > 0 && checkedDeletable.length === deletableIds.length; + const someChecked = checkedDeletable.length > 0; + + const toggleOne = (id: string) => { + setChecked((prev) => { + const next = new Set(prev); + if (next.has(id)) next.delete(id); + else next.add(id); + return next; + }); + }; + + const toggleAll = () => { + setChecked((prev) => { + const next = new Set(prev); + if (allDeletableChecked) { + for (const id of deletableIds) next.delete(id); + } else { + for (const id of deletableIds) next.add(id); + } + return next; + }); + }; + + const clearSelection = () => { + setChecked(new Set()); + setConfirming(false); + }; + + const handleConfirmDelete = async () => { + if (!onDeleteCalls || checkedDeletable.length === 0 || busy) return; + setBusy(true); + try { + await onDeleteCalls(checkedDeletable); + clearSelection(); + } finally { + setBusy(false); + } + }; + return (
@@ -162,10 +307,104 @@ export function CallTable({ streaming · SSE
-
- + + {someChecked ? ( +
+ + {checkedDeletable.length} + + {checkedDeletable.length === 1 ? 'call selected' : 'calls selected'} + + +
+ {confirming ? ( + <> + + Removes from view + metrics. Logs kept on disk. + + + + + ) : ( + <> + + + + )} +
+ ) : null} + +
+
+ @@ -177,7 +416,7 @@ export function CallTable({ {filtered.length === 0 ? ( - @@ -189,6 +428,11 @@ export function CallTable({ isSelected={c.id === selectedId} onSelect={() => onSelect(c.id)} isNew={c.id === newId} + isChecked={checked.has(c.id)} + onToggleCheck={ + c.status === 'live' ? null : () => toggleOne(c.id) + } + revealed={revealed} /> )) )} diff --git a/dashboard-app/src/components/CostPanel.tsx b/dashboard-app/src/components/CostPanel.tsx index ebeb233a..bf83c5b4 100644 --- a/dashboard-app/src/components/CostPanel.tsx +++ b/dashboard-app/src/components/CostPanel.tsx @@ -1,51 +1,83 @@ import type { Call } from './CallTable'; +import { fmtCostUSD } from './format'; export interface CostPanelProps { call: Call | null; } +function titleCase(s: string): string { + if (s.length === 0) return s; + // Strip provider-key transport suffixes (_ws, _rest) and role suffixes + // (_stt, _tts, _llm). Repeated `+` handles compound suffixes like + // "cartesia_tts_ws" -> "cartesia". The SDK uses provider_key like + // "elevenlabs_ws" / "cartesia_stt" to disambiguate adapter classes; + // the suffix is internal noise in user-facing UI. + const cleaned = s.replace(/(?:_(?:ws|rest|stt|tts|llm))+$/i, ''); + return cleaned.charAt(0).toUpperCase() + cleaned.slice(1); +} + export function CostPanel({ call }: CostPanelProps) { if (!call || !call.cost?.telco) return null; const c = call.cost; const telco = c.telco ?? 0; const llm = c.llm ?? 0; - const sttTts = c.sttTts ?? 0; + const stt = c.stt ?? 0; + const tts = c.tts ?? 0; + const sttTtsLegacy = c.sttTts ?? stt + tts; const cached = c.cached ?? 0; - const subtotal = telco + llm + sttTts; + const subtotal = telco + llm + sttTtsLegacy; const total = subtotal - cached; const seg = (v: number) => (subtotal > 0 ? (v / subtotal) * 100 : 0); + const sttLabel = call.sttProvider + ? `${titleCase(call.sttProvider)} STT${call.sttModel ? ` · ${call.sttModel}` : ''}` + : 'STT'; + const ttsLabel = call.ttsProvider + ? `${titleCase(call.ttsProvider)} TTS${call.ttsModel ? ` · ${call.ttsModel}` : ''}` + : 'TTS'; + const llmLabel = call.llmModel + ? `${call.model ? titleCase(call.model) + ' · ' : ''}${call.llmModel}` + : call.model || 'LLM'; + return (

Cost breakdown

- + +
{call.carrier === 'twilio' ? 'Twilio' : 'Telnyx'} - ${telco.toFixed(3)} + {fmtCostUSD(telco)}
- {call.model || 'LLM'} + {llmLabel} - ${llm.toFixed(3)} - {cached > 0 && −${cached.toFixed(3)} cached} + {fmtCostUSD(llm)} + {cached > 0 && −{fmtCostUSD(cached)} cached}
- STT / TTS + {sttLabel} + + {fmtCostUSD(stt)} +
+
+ + + {ttsLabel} - ${sttTts.toFixed(3)} + {fmtCostUSD(tts)}
@@ -61,7 +93,7 @@ export function CostPanel({ call }: CostPanelProps) { {call.status === 'live' ? '(running)' : ''} - ${total.toFixed(3)} + {fmtCostUSD(total)}
); diff --git a/dashboard-app/src/components/LatencyPanel.tsx b/dashboard-app/src/components/LatencyPanel.tsx index e769b0a1..ce41fdac 100644 --- a/dashboard-app/src/components/LatencyPanel.tsx +++ b/dashboard-app/src/components/LatencyPanel.tsx @@ -4,48 +4,64 @@ export interface LatencyPanelProps { call: Call | null; } +// 2 turn = almeno 1 turn user genuino oltre al firstMessage. Sotto a 2 i +// percentili sono privi di senso (un singolo campione). Sopra a 2 sono +// statisticamente magri ma informativi — meglio mostrarli che lasciare il +// pannello con "—" quando la tabella sopra mostra già una p95 dal fallback +// ad avg. +const MIN_TURNS_FOR_PERCENTILES = 2; + export function LatencyPanel({ call }: LatencyPanelProps) { - if (!call || !call.latencyP95) return null; + if (!call || (!call.latencyP95 && !call.agentResponseP95)) return null; - const stt = call.sttAvg || 0; - const llm = call.latencyP50 || 0; - const tts = call.ttsAvg || 0; + const stt = call.sttAvg ?? 0; + const llm = call.llmAvg ?? 0; + const tts = call.ttsAvg ?? 0; const total = stt + llm + tts; const max = Math.max(total, 800); + const turns = call.turnCount ?? 0; + const showPercentiles = turns >= MIN_TURNS_FOR_PERCENTILES; + const dash = '—'; + return (

Latency · this call

-
p50
+
p50 round-trip
- {call.latencyP50} - ms + {showPercentiles ? call.latencyP50 ?? dash : dash} + {showPercentiles && ms}
-
600 ? ' warn' : '')}> -
p95
+
600 ? ' warn' : '')}> +
p95 round-trip
- {call.latencyP95} - ms + {showPercentiles ? call.latencyP95 ?? dash : dash} + {showPercentiles && ms}
-
stt avg
+
p50 wait
- {call.sttAvg} - ms + {showPercentiles ? call.agentResponseP50 ?? dash : dash} + {showPercentiles && ms}
-
-
tts avg
+
600 ? ' warn' : '')}> +
p95 wait
- {call.ttsAvg} - ms + {showPercentiles ? call.agentResponseP95 ?? dash : dash} + {showPercentiles && ms}
+ {!showPercentiles && ( +
+ {turns} {turns === 1 ? 'turn' : 'turns'} — percentiles need ≥{MIN_TURNS_FOR_PERCENTILES} +
+ )}
@@ -89,7 +105,7 @@ export function LatencyPanel({ call }: LatencyPanelProps) { tts - total {total} ms + avg wait {Math.round(total)} ms
); diff --git a/dashboard-app/src/components/LiveCallPanel.tsx b/dashboard-app/src/components/LiveCallPanel.tsx index dbedb11b..d165ec21 100644 --- a/dashboard-app/src/components/LiveCallPanel.tsx +++ b/dashboard-app/src/components/LiveCallPanel.tsx @@ -1,6 +1,6 @@ import { useEffect, useRef, useState } from 'react'; import type { Call } from './CallTable'; -import { fmtDuration } from './format'; +import { fmtDuration, fmtPhone } from './format'; import { IconForward, IconHangup, IconMic, IconRecord } from './icons'; interface LiveDurationProps { @@ -32,6 +32,8 @@ export interface LiveCallPanelProps { setRecording: (v: boolean) => void; muted: boolean; setMuted: (v: boolean) => void; + /** PII reveal — when false the displayed phone number is masked. */ + revealed: boolean; } export function LiveCallPanel({ @@ -42,6 +44,7 @@ export function LiveCallPanel({ setRecording, muted, setMuted, + revealed, }: LiveCallPanelProps) { const scrollRef = useRef(null); useEffect(() => { @@ -68,7 +71,12 @@ export function LiveCallPanel({ {call.status}
- {call.direction === 'inbound' ? call.from : call.to} + + {fmtPhone( + call.direction === 'inbound' ? call.from : call.to, + revealed, + )} + · {call.agent}
diff --git a/dashboard-app/src/components/Metric.tsx b/dashboard-app/src/components/Metric.tsx index 84fc6349..3de7adc6 100644 --- a/dashboard-app/src/components/Metric.tsx +++ b/dashboard-app/src/components/Metric.tsx @@ -1,5 +1,6 @@ import { useState } from 'react'; import type { Call } from './CallTable'; +import { fmtCostUSD } from './format'; export interface MetricBucket { /** Bar height 0-100. */ @@ -12,6 +13,17 @@ export interface MetricBucket { readonly toMs: number; } +/** + * Tooltip kind drives the headline aggregate shown above the per-call + * sample list: + * - ``count`` → "N CALLS" + * - ``latency`` → "AVG LATENCY MS" + * - ``spend`` → "TOTAL COST $" + * The rendered list of recent calls in the bucket is shared across all + * three so the user can still drill down into a specific call. + */ +export type MetricKind = 'count' | 'latency' | 'spend'; + export interface MetricProps { label: string; value: string | number; @@ -24,6 +36,12 @@ export interface MetricProps { buckets?: readonly MetricBucket[]; /** Called when the user clicks a bar that contains at least one call. */ onSelectCall?: (callId: string) => void; + /** + * Which aggregate the tooltip headline reports for this card. Defaults + * to ``count`` so existing callers (no kind passed) keep their previous + * "N calls" label. + */ + kind?: MetricKind; peach?: boolean; footer?: string; badge?: boolean; @@ -87,11 +105,43 @@ function newestCallId(bucket: MetricBucket): string | undefined { return sorted[0]?.id; } +/** + * Compute the per-bucket headline for the sparkline tooltip. Returns the + * uppercase label (e.g. "AVG LATENCY") and the formatted value (e.g. + * "3048 ms") for the chosen ``kind``. Cleanly separated from the + * presentation so the same card-level numbers shown on the metric tile + * itself can be re-used by tests. + */ +export function bucketHeadline( + bucket: MetricBucket, + kind: MetricKind, +): { label: string; value: string } { + const calls = bucket.calls; + const count = calls.length; + if (kind === 'spend') { + const sum = calls.reduce((acc, c) => acc + callCost(c), 0); + return { label: 'TOTAL COST', value: fmtCostUSD(sum) }; + } + if (kind === 'latency') { + const withLat = calls.filter((c) => typeof c.latencyP95 === 'number'); + const avg = + withLat.length > 0 + ? Math.round( + withLat.reduce((acc, c) => acc + (c.latencyP95 ?? 0), 0) / + withLat.length, + ) + : 0; + return { label: 'AVG LATENCY', value: `${avg} ms` }; + } + return { label: count === 1 ? 'CALL' : 'CALLS', value: `${count}` }; +} + interface SparkTooltipProps { bucket: MetricBucket; + kind: MetricKind; } -function SparkTooltip({ bucket }: SparkTooltipProps) { +function SparkTooltip({ bucket, kind }: SparkTooltipProps) { const range = bucketRange(bucket); const count = bucket.calls.length; @@ -104,12 +154,14 @@ function SparkTooltip({ bucket }: SparkTooltipProps) { ); } + const headline = bucketHeadline(bucket, kind); const sample = bucket.calls.slice(0, 4); return (
{range}
-
- {count} call{count === 1 ? '' : 's'} +
+ {headline.label} + {headline.value}
    {sample.map((c) => { @@ -118,7 +170,7 @@ function SparkTooltip({ bucket }: SparkTooltipProps) {
  • {num} {c.status} - ${callCost(c).toFixed(3)} + {fmtCostUSD(callCost(c))}
    • ); })} @@ -134,10 +186,11 @@ interface SparkBarProps { bucket: MetricBucket | undefined; height: number; interactive: boolean; + kind: MetricKind; onSelect?: (id: string) => void; } -function SparkBar({ bucket, height, interactive, onSelect }: SparkBarProps) { +function SparkBar({ bucket, height, interactive, kind, onSelect }: SparkBarProps) { const [hovered, setHovered] = useState(false); const hasCalls = !!bucket && bucket.calls.length > 0; @@ -165,7 +218,7 @@ function SparkBar({ bucket, height, interactive, onSelect }: SparkBarProps) { onBlur={() => setHovered(false)} aria-label={`${bucket.calls.length} calls in ${bucketRange(bucket)}`} /> - {hovered && } + {hovered && }
); } @@ -179,6 +232,7 @@ export function Metric({ spark, buckets, onSelectCall, + kind = 'count', peach, footer, badge, @@ -204,6 +258,7 @@ export function Metric({ bucket={buckets?.[i]} height={h} interactive={interactive} + kind={kind} onSelect={onSelectCall} /> ))} diff --git a/dashboard-app/src/components/MetricsPanel.tsx b/dashboard-app/src/components/MetricsPanel.tsx index 2b976735..2c5ffd45 100644 --- a/dashboard-app/src/components/MetricsPanel.tsx +++ b/dashboard-app/src/components/MetricsPanel.tsx @@ -1,5 +1,6 @@ import { useState } from 'react'; import type { Call } from './CallTable'; +import { fmtCostUSD } from './format'; export interface MetricsPanelProps { call: Call | null; @@ -54,8 +55,10 @@ export function MetricsPanel({ call }: MetricsPanelProps) {
- {activeTab === 'latency' && showLatency && } - {activeTab === 'cost' && showCost && } +
+ {activeTab === 'latency' && showLatency && } + {activeTab === 'cost' && showCost && } +
); } @@ -72,21 +75,22 @@ function LatencyView({ call }: { call: Call }) { // round-trip, so the SDK only knows the end-to-end latency. Breaking it // into stt/llm/tts is meaningless. Pipeline-mode calls expose all four. if (isRealtime) { + const showPctRt = (call.turnCount ?? 0) >= 2; return ( <>
end-to-end p50
- {p50 || '—'} - ms + {showPctRt ? p50 || '—' : '—'} + {showPctRt && ms}
-
600 ? ' warn' : '')}> +
600 ? ' warn' : '')}>
end-to-end p95
- {p95 || '—'} - ms + {showPctRt ? p95 || '—' : '—'} + {showPctRt && ms}
@@ -115,10 +119,14 @@ function LatencyView({ call }: { call: Call }) { } const stt = call.sttAvg || 0; - const llm = call.latencyP50 || 0; + const llm = call.llmAvg || 0; const tts = call.ttsAvg || 0; const total = stt + llm + tts; const max = Math.max(total, 800); + // Percentile boxes are statistical noise on calls with too few turns + // (with n=4 samples, p95 is interpolation between sample[2] and sample[3] + // and doesn't correspond to any real turn). Show ``—`` until ≥5 turns. + const showPct = (call.turnCount ?? 0) >= 2; return ( <> @@ -126,15 +134,15 @@ function LatencyView({ call }: { call: Call }) {
p50
- {call.latencyP50 ?? '—'} - ms + {showPct ? call.latencyP50 ?? '—' : '—'} + {showPct && ms}
-
600 ? ' warn' : '')}> +
600 ? ' warn' : '')}>
p95
- {p95} - ms + {showPct ? p95 : '—'} + {showPct && ms}
@@ -203,23 +211,51 @@ function LatencyView({ call }: { call: Call }) { // ---------- Cost ---------- +function titleCase(s: string): string { + if (s.length === 0) return s; + // Strip provider-key transport suffixes (_ws, _rest) and role suffixes + // (_stt, _tts, _llm). Repeated `+` handles compound suffixes like + // "cartesia_tts_ws" -> "cartesia". The SDK uses provider_key like + // "elevenlabs_ws" / "cartesia_stt" to disambiguate adapter classes; + // the suffix is internal noise in user-facing UI. + const cleaned = s.replace(/(?:_(?:ws|rest|stt|tts|llm))+$/i, ''); + return cleaned.charAt(0).toUpperCase() + cleaned.slice(1); +} + function CostView({ call }: { call: Call }) { const c = call.cost; const telco = c.telco ?? 0; const llm = c.llm ?? 0; - const sttTts = c.sttTts ?? 0; + // Always prefer the per-component split when present; fall back to the + // legacy combined ``sttTts`` field only when the SDK didn't emit the + // split. Greenfield calls (>=0.6.1) always emit stt + tts separately. + const stt = c.stt ?? 0; + const tts = c.tts ?? 0; + const sttTtsCombined = c.sttTts ?? 0; + const sttTtsLegacy = stt === 0 && tts === 0 ? sttTtsCombined : 0; const cached = c.cached ?? 0; - const subtotal = telco + llm + sttTts; + const subtotal = telco + llm + stt + tts + sttTtsLegacy; const total = c.total ?? subtotal - cached; const seg = (v: number) => (subtotal > 0 ? (v / subtotal) * 100 : 0); + const sttLabel = call.sttProvider + ? `${titleCase(call.sttProvider)} STT${call.sttModel ? ` · ${call.sttModel}` : ''}` + : 'STT'; + const ttsLabel = call.ttsProvider + ? `${titleCase(call.ttsProvider)} TTS${call.ttsModel ? ` · ${call.ttsModel}` : ''}` + : 'TTS'; + const llmLabel = call.llmModel + ? `${call.model ? titleCase(call.model) + ' · ' : ''}${call.llmModel}` + : call.model || 'LLM'; + return ( <> {subtotal > 0 && (
- + +
)} {telco > 0 && ( @@ -228,26 +264,44 @@ function CostView({ call }: { call: Call }) { {call.carrier === 'twilio' ? 'Twilio' : 'Telnyx'} - ${telco.toFixed(3)} + {fmtCostUSD(telco)}
)} {llm > 0 && (
- {call.model || 'LLM'} + {llmLabel} + + {fmtCostUSD(llm)} + {cached > 0 && −{fmtCostUSD(cached)} cached} +
+ )} + {stt > 0 && ( +
+ + + {sttLabel} + + {fmtCostUSD(stt)} +
+ )} + {tts > 0 && ( +
+ + + {ttsLabel} - ${llm.toFixed(3)} - {cached > 0 && −${cached.toFixed(3)} cached} + {fmtCostUSD(tts)}
)} - {sttTts > 0 && ( + {sttTtsLegacy > 0 && (
- STT / TTS + STT / TTS (legacy) - ${sttTts.toFixed(3)} + {fmtCostUSD(sttTtsLegacy)}
)}
@@ -266,7 +320,7 @@ function CostView({ call }: { call: Call }) { )} - ${total.toFixed(3)} + {fmtCostUSD(total)}
); diff --git a/dashboard-app/src/components/PatterLogo.tsx b/dashboard-app/src/components/PatterLogo.tsx index dae0f92c..4e88023a 100644 --- a/dashboard-app/src/components/PatterLogo.tsx +++ b/dashboard-app/src/components/PatterLogo.tsx @@ -69,15 +69,19 @@ function PatterWordmark(props: SVGProps) { // Composite — mark on the left + wordmark on the right, sized // independently so the mark's line weight stays light when the wordmark -// scales up. Both inherit ``currentColor`` so they adapt to dark mode. +// scales up. Both SVGs use ``currentColor`` so they inherit the parent +// ``.brand`` text colour — which is ``var(--ink)`` (#000) in light mode +// and ``#e8e8e8`` in dark mode (cascaded from ``body.dark``). Do NOT pin +// an inline ``color`` here: a hard-coded ``var(--ink)`` would render the +// wordmark black-on-black on the dark page. export function PatterLogo() { return ( diff --git a/dashboard-app/src/components/Topbar.tsx b/dashboard-app/src/components/Topbar.tsx index 207a5df3..c4350782 100644 --- a/dashboard-app/src/components/Topbar.tsx +++ b/dashboard-app/src/components/Topbar.tsx @@ -1,13 +1,32 @@ import { PatterLogo } from './PatterLogo'; +import { fmtPhone } from './format'; +import { IconEye, IconEyeOff, IconMoon, IconSun } from './icons'; export interface TopbarProps { liveCount: number; todayCount: number; + /** Raw phone number as the SDK reported it (may already be masked on disk). */ phoneNumber: string; sdkVersion: string; + /** PII reveal state: when ``true`` show full numbers, when ``false`` mask. */ + revealed: boolean; + /** Dark-theme state for the ``body.dark`` override. */ + dark: boolean; + onToggleRevealed: () => void; + onToggleDark: () => void; } -export function Topbar({ liveCount, todayCount, phoneNumber, sdkVersion }: TopbarProps) { +export function Topbar({ + liveCount, + todayCount, + phoneNumber, + sdkVersion, + revealed, + dark, + onToggleRevealed, + onToggleDark, +}: TopbarProps) { + const displayNumber = fmtPhone(phoneNumber, revealed); return (
@@ -19,7 +38,29 @@ export function Topbar({ liveCount, todayCount, phoneNumber, sdkVersion }: Topba 0 ? ' active' : '')}> {liveCount} live · {todayCount} today - {phoneNumber !== '—' && {phoneNumber}} + {phoneNumber && phoneNumber !== '—' && ( + {displayNumber} + )} + +
); diff --git a/dashboard-app/src/components/format.ts b/dashboard-app/src/components/format.ts index 096e245e..d28753d9 100644 --- a/dashboard-app/src/components/format.ts +++ b/dashboard-app/src/components/format.ts @@ -14,6 +14,65 @@ export function fmtAgo(sec: number): string { return `${Math.floor(sec / 3600)}h ago`; } -export function fmtPhone(p: string): string { - return p; +/** + * Render a phone number for display. + * + * The SDK's CallLogger writes phone numbers to disk in one of three forms + * (controlled by ``PATTER_LOG_REDACT_PHONE``): + * - ``mask`` (default) → ``***`` (three U+002A asterisks) + * - ``full`` → raw E.164 (``+15551234567``) + * - ``hash_only`` → ``sha256:<16-hex>`` + * + * When ``revealed=true`` we hand the value back as-is so the operator + * sees whatever the server provided (full when ``full``, masked when + * ``mask``). When ``revealed=false`` we ENFORCE masking client-side — + * even if the server happens to have full numbers we don't render them. + * The masked form uses U+2022 BULLET ``•`` instead of asterisks because + * bullets sit on the digit baseline; asterisks float toward the cap + * height and look misaligned next to numerals. + * + * Examples (revealed=false): + * "+15556234231" → "•••4231" + * "***4231" → "•••4231" (re-normalise existing server masking) + * "sha256:abcd…" → "••••••••" + * "" / "?" → "" (let callers fall back to "—") + */ +export function fmtPhone(p: string, revealed = true): string { + if (!p) return ''; + if (revealed) { + // Honour whatever the server gave us. Re-render legacy "***" as + // bullets so the alignment is consistent even when the operator opts + // to reveal — the underlying log artefact is unchanged. + if (p.startsWith('***')) return '•••' + p.slice(3); + return p; + } + // Masked mode. Try to keep a last-4 anchor for correlation. + if (p.startsWith('***')) return '•••' + p.slice(3); + if (p.startsWith('sha256:')) return '••••••••'; + const digits = p.replace(/\D/g, ''); + if (digits.length >= 4) return '•••' + digits.slice(-4); + return '••••••••'; +} + +/** + * Render a USD amount with precision adapted to its magnitude so per-call + * costs from cheap providers (Cerebras gpt-oss-120b ≈ $0.0001 / 5-turn call) + * are not flattened to "$0.00" by a fixed `toFixed(2)`. + * + * ≥ $0.01 → 2 decimals "$0.12" + * ≥ $0.001 → 3 decimals "$0.012" + * ≥ $0.0001 → 4 decimals "$0.0001" + * > 0 → 5 decimals "$0.00001" + * 0 / nullish → "$0.00" + */ +export function fmtCostUSD(value: number | undefined | null): string { + if (value === undefined || value === null || !Number.isFinite(value)) { + return '$0.00'; + } + const v = Math.abs(value); + if (v === 0) return '$0.00'; + if (v >= 0.01) return `$${value.toFixed(2)}`; + if (v >= 0.001) return `$${value.toFixed(3)}`; + if (v >= 0.0001) return `$${value.toFixed(4)}`; + return `$${value.toFixed(5)}`; } diff --git a/dashboard-app/src/components/icons.tsx b/dashboard-app/src/components/icons.tsx index 9752028c..78e94724 100644 --- a/dashboard-app/src/components/icons.tsx +++ b/dashboard-app/src/components/icons.tsx @@ -97,6 +97,149 @@ export function IconSettings(props: IconProps) { ); } +export function IconEye(props: IconProps) { + return ( + + + + + ); +} + +export function IconEyeOff(props: IconProps) { + return ( + + + + + + + ); +} + +export function IconSun(props: IconProps) { + return ( + + + + + + + + + + + + ); +} + +export function IconMoon(props: IconProps) { + return ( + + + + ); +} + +export function IconTrash(props: IconProps) { + return ( + + + + + + + + ); +} + +export function IconX(props: IconProps) { + return ( + + + + + ); +} + +export function IconCheck(props: IconProps) { + return ( + + + + ); +} + export function IconPlus(props: IconProps) { return ( = {}): CallRecord { + return { + call_id: callId, + caller: `from-${callId}`, + callee: `to-${callId}`, + direction: 'inbound', + started_at: 1000, + status: 'in-progress', + transcript: [], + turns: [], + metrics: null, + ...overrides, + }; +} + +function makeCall(id: string, overrides: Partial = {}): Call { + return { + id, + status: 'live', + direction: 'inbound', + from: `from-${id}`, + to: `to-${id}`, + carrier: 'twilio', + cost: {}, + ...overrides, + }; +} + +describe('mergeCalls', () => { + it('returns active before recent and dedupes by call_id', () => { + const active = [record('a', { status: 'in-progress' })]; + const recent = [record('b', { status: 'completed', ended_at: 1100 })]; + const result = mergeCalls(active, recent); + expect(result.map((c) => c.id)).toEqual(['a', 'b']); + }); + + it('active wins over recent when call_id appears in both', () => { + const active = [record('a', { status: 'in-progress', caller: 'live' })]; + const recent = [record('a', { status: 'completed', caller: 'stale' })]; + const result = mergeCalls(active, recent); + expect(result).toHaveLength(1); + expect(result[0].status).toBe('live'); + expect(result[0].from).toBe('live'); + }); +}); + +describe('mergeCallPreserving', () => { + it('regression #124: a second call_start refresh keeps the first call visible', () => { + // Step 1: call A is live, no recent. + const stateAfterAStart = mergeCallPreserving( + [], + mergeCalls([record('A', { status: 'in-progress' })], []), + ); + expect(stateAfterAStart.map((c) => c.id)).toEqual(['A']); + + // Step 2: call A ended, snapshot still includes it via /calls. + const stateAfterAEnd = mergeCallPreserving( + stateAfterAStart, + mergeCalls( + [], + [record('A', { status: 'completed', ended_at: 1100 })], + ), + ); + expect(stateAfterAEnd.map((c) => c.id)).toEqual(['A']); + expect(stateAfterAEnd[0].status).toBe('ended'); + + // Step 3: call B starts. The server SSE for call_start fires the refresh + // BEFORE the prior call A propagates to /api/dashboard/calls — simulate + // by having only B in the snapshot. Without the upsert, A would vanish. + const stateAfterBStart = mergeCallPreserving( + stateAfterAEnd, + mergeCalls([record('B', { status: 'in-progress' })], []), + ); + expect(stateAfterBStart.map((c) => c.id).sort()).toEqual(['A', 'B']); + }); + + it('upserts: next replaces prev for same id but unknown prev calls are kept', () => { + const prev: Call[] = [ + makeCall('A', { status: 'live', latencyP95: 250 }), + makeCall('B', { status: 'ended' }), + ]; + const next: Call[] = [makeCall('A', { status: 'ended', latencyP95: 280 })]; + const result = mergeCallPreserving(prev, next); + const a = result.find((c) => c.id === 'A')!; + const b = result.find((c) => c.id === 'B')!; + expect(a.status).toBe('ended'); + expect(a.latencyP95).toBe(280); + expect(b.status).toBe('ended'); + }); + + it('preserves rich fields the fresh payload omits', () => { + const prev: Call[] = [ + makeCall('A', { + latencyP95: 250, + latencyP50: 180, + sttAvg: 90, + ttsAvg: 110, + llmAvg: 320, + turnCount: 7, + agentResponseP50: 420, + agentResponseP95: 980, + cost: { llm: 0.01, stt: 0.002 }, + }), + ]; + const next: Call[] = [ + makeCall('A', { + status: 'ended', + cost: { llm: 0.012 }, + }), + ]; + const merged = mergeCallPreserving(prev, next); + const a = merged[0]; + expect(a.status).toBe('ended'); + expect(a.latencyP95).toBe(250); + expect(a.latencyP50).toBe(180); + expect(a.sttAvg).toBe(90); + expect(a.ttsAvg).toBe(110); + expect(a.llmAvg).toBe(320); + expect(a.turnCount).toBe(7); + expect(a.agentResponseP50).toBe(420); + expect(a.agentResponseP95).toBe(980); + expect(a.cost.llm).toBe(0.012); + expect(a.cost.stt).toBe(0.002); + }); + + it('two consecutive call_start SSE events for different ids end up with both visible', () => { + let state: Call[] = []; + state = mergeCallPreserving(state, mergeCalls([record('one')], [])); + expect(state.map((c) => c.id)).toEqual(['one']); + state = mergeCallPreserving(state, mergeCalls([record('two')], [])); + expect(state.map((c) => c.id).sort()).toEqual(['one', 'two']); + }); + + it('caps the merged UI list at 500 entries (mirrors server ring buffer)', () => { + // 600 prev rows with distinct ids and ascending startedAtMs so the + // sort can stably order them newest-first. The cap drops the + // oldest 100. + const prev: Call[] = Array.from({ length: 600 }, (_, i) => + makeCall(`prev-${i}`, { startedAtMs: 1000 + i }), + ); + // One fresh call from the snapshot. + const next: Call[] = [makeCall('fresh', { startedAtMs: 2000 })]; + + const result = mergeCallPreserving(prev, next); + expect(result.length).toBe(500); + // The newest (``fresh`` at 2000) lands first. + expect(result[0].id).toBe('fresh'); + // 600 prev + 1 fresh = 601 candidates → slice keeps the top 500. + // ``fresh`` (2000) plus prev-599 (1599) down to prev-101 (1101) + // survive; prev-100 (1100) and older are dropped. + const ids = new Set(result.map((c) => c.id)); + expect(ids.has('prev-0')).toBe(false); + expect(ids.has('prev-100')).toBe(false); + // The newest ``prev`` rows survive. + expect(ids.has('prev-599')).toBe(true); + expect(ids.has('prev-101')).toBe(true); + }); + + it('sorts merged calls by startedAtMs descending — newer first', () => { + // ``prev`` holds an older call A; ``next`` adds a newer call B. + // Without the sort, B (a ``next`` entry) would lead and A (a + // ``prev_only`` entry) would land at the bottom regardless of its + // start time. With the sort, ordering is purely by startedAtMs. + const prev: Call[] = [makeCall('A', { startedAtMs: 1000 })]; + const next: Call[] = [makeCall('B', { startedAtMs: 2000 })]; + const result = mergeCallPreserving(prev, next); + expect(result.map((c) => c.id)).toEqual(['B', 'A']); + }); +}); diff --git a/dashboard-app/src/hooks/mergeCalls.ts b/dashboard-app/src/hooks/mergeCalls.ts new file mode 100644 index 00000000..8cf73ece --- /dev/null +++ b/dashboard-app/src/hooks/mergeCalls.ts @@ -0,0 +1,103 @@ +// Pure SSE/refresh merge helpers extracted from useDashboardData so they can +// be unit-tested without a React harness. Both functions are immutable — +// callers must treat the returned arrays as readonly. + +import type { CallRecord } from '../lib/api'; +import { toUiCall, type Call } from '../lib/mappers'; + +/** + * Hard cap on the number of calls retained in the SPA after a merge. + * Mirrors the server-side ``MetricsStore`` ring buffer default (500) so the + * UI cannot accumulate ``prev_only`` rows for calls the server has already + * evicted. Without this cap, ``mergeCallPreserving`` would grow the array + * indefinitely on long-lived sessions: every prior call still pinned by + * ``prev`` would be re-appended on every refresh even after the server has + * dropped it from the ring buffer. + */ +const MAX_UI_CALLS = 500; + +/** + * Project the server's active + recent payloads into a single UI list with + * stable ordering: active calls first (live status surfaces at the top), + * then completed calls newest-first as the server returned them. Duplicate + * ``call_id`` is resolved active-wins-over-recent so a still-running call + * never gets a stale terminal row. + */ +export function mergeCalls(active: CallRecord[], recent: CallRecord[]): Call[] { + const seen = new Set(); + const merged: Call[] = []; + for (const record of active) { + if (seen.has(record.call_id)) continue; + seen.add(record.call_id); + merged.push(toUiCall(record)); + } + for (const record of recent) { + if (seen.has(record.call_id)) continue; + seen.add(record.call_id); + merged.push(toUiCall(record)); + } + return merged; +} + +/** + * Upsert a fresh snapshot of calls into the previous UI state by ``call_id``. + * + * Two reasons this is an upsert rather than a replace: + * + * 1. ``MetricsStore.updateCallStatus`` may write a synthetic terminal + * record with ``metrics: undefined`` ahead of the canonical + * ``recordCallEnd`` write (Twilio statusCallback racing the WS ``stop`` + * frame). The ``next.field ?? prev.field`` per-critical-field merge + * masks the race window so transcripts + latency don't blank out. See + * ``store.ts`` TODO(0.6.2) for the root-cause fix. + * + * 2. When a second call starts back-to-back with the first, the SSE + * ``call_start`` refresh occasionally lands with the freshly-ended call + * not yet visible in ``/api/dashboard/calls`` (the server publishes the + * SSE event for the new call before the prior call's terminal write + * completes, or pagination clips it). Replacing the array verbatim + * with the server response would drop the prior call from the UI even + * though it is still in the ring buffer — exactly the regression + * reported in #124. Treating ``prev`` as the union-anchor keeps the + * prior call visible until the server snapshot stabilises. + * + * The server's ``maxCalls`` ring buffer (default 500) bounds growth on + * long-lived sessions; the UI list is naturally bounded by what + * ``fetchCalls`` paginates plus whatever lives in ``prev`` from the + * current session. + */ +export function mergeCallPreserving(prev: Call[], next: Call[]): Call[] { + const prevById = new Map(prev.map((c) => [c.id, c])); + const nextIds = new Set(next.map((c) => c.id)); + const merged: Call[] = next.map((nc) => { + const pc = prevById.get(nc.id); + if (!pc) return nc; + return { + ...pc, + ...nc, + latencyP95: nc.latencyP95 ?? pc.latencyP95, + latencyP50: nc.latencyP50 ?? pc.latencyP50, + sttAvg: nc.sttAvg ?? pc.sttAvg, + ttsAvg: nc.ttsAvg ?? pc.ttsAvg, + llmAvg: nc.llmAvg ?? pc.llmAvg, + turnCount: nc.turnCount ?? pc.turnCount, + agentResponseP50: nc.agentResponseP50 ?? pc.agentResponseP50, + agentResponseP95: nc.agentResponseP95 ?? pc.agentResponseP95, + cost: { ...pc.cost, ...nc.cost }, + }; + }); + for (const pc of prev) { + if (!nextIds.has(pc.id)) merged.push(pc); + } + // Sort by ``startedAtMs`` descending so the newest call always lands at + // the top, regardless of whether it came from the snapshot or from + // ``prev``. Without this, ``prev_only`` entries appended after the + // snapshot block kept ordering non-deterministic (live row first only + // when the snapshot already contained it). Calls without an + // ``startedAtMs`` (rare — synthetic terminal rows before the canonical + // write) sort to the end so they don't outrank a live call. + merged.sort((a, b) => (b.startedAtMs ?? 0) - (a.startedAtMs ?? 0)); + // Cap to ``MAX_UI_CALLS`` so a long-lived session that has cycled + // through more than 500 calls cannot grow the UI array unbounded. + return merged.slice(0, MAX_UI_CALLS); +} diff --git a/dashboard-app/src/hooks/useDashboardData.ts b/dashboard-app/src/hooks/useDashboardData.ts index 17100704..d5d67557 100644 --- a/dashboard-app/src/hooks/useDashboardData.ts +++ b/dashboard-app/src/hooks/useDashboardData.ts @@ -14,9 +14,9 @@ import { fetchAggregates, fetchCalls, type Aggregates, - type CallRecord, } from '../lib/api'; -import { toUiCall, type Call } from '../lib/mappers'; +import type { Call } from '../lib/mappers'; +import { mergeCalls, mergeCallPreserving } from './mergeCalls'; export interface DashboardData { readonly calls: Call[]; @@ -24,6 +24,12 @@ export interface DashboardData { readonly isStreaming: boolean; readonly error: string | null; readonly refresh: () => Promise; + /** + * Optimistically remove ``ids`` from the local call list before the next + * server refresh lands. Avoids the brief flash of the deleted row + * lingering between the DELETE request and the next snapshot fetch. + */ + readonly removeCallsLocal: (ids: readonly string[]) => void; } const RECONNECT_INITIAL_MS = 1_000; @@ -36,24 +42,9 @@ const RELEVANT_EVENTS = [ 'call_initiated', 'call_status', 'call_end', + 'calls_deleted', ] as const; -function mergeCalls(active: CallRecord[], recent: CallRecord[]): Call[] { - const seen = new Set(); - const merged: Call[] = []; - for (const record of active) { - if (seen.has(record.call_id)) continue; - seen.add(record.call_id); - merged.push(toUiCall(record)); - } - for (const record of recent) { - if (seen.has(record.call_id)) continue; - seen.add(record.call_id); - merged.push(toUiCall(record)); - } - return merged; -} - function describeError(err: unknown): string { if (err instanceof Error) return err.message; return 'Unknown error'; @@ -100,7 +91,7 @@ export function useDashboardData(): DashboardData { fetchAggregates(), ]); if (!mountedRef.current) return; - setCalls(mergeCalls(active, recent)); + setCalls((prev) => mergeCallPreserving(prev, mergeCalls(active, recent))); setAggregates(aggs); setError(null); } catch (err) { @@ -197,5 +188,11 @@ export function useDashboardData(): DashboardData { // eslint-disable-next-line react-hooks/exhaustive-deps }, []); - return { calls, aggregates, isStreaming, error, refresh }; + const removeCallsLocal = useCallback((ids: readonly string[]): void => { + if (ids.length === 0) return; + const drop = new Set(ids); + setCalls((prev) => prev.filter((c) => !drop.has(c.id))); + }, []); + + return { calls, aggregates, isStreaming, error, refresh, removeCallsLocal }; } diff --git a/dashboard-app/src/hooks/useTranscript.ts b/dashboard-app/src/hooks/useTranscript.ts index d5eef572..e6885968 100644 --- a/dashboard-app/src/hooks/useTranscript.ts +++ b/dashboard-app/src/hooks/useTranscript.ts @@ -3,9 +3,24 @@ // Behaviour: // - When `callId` is null, returns [] and does nothing. // - When `callId` changes, fetches the call once and maps the transcript. -// - When `isLive` is true, polls the call detail endpoint every 2 seconds -// so newly streamed turns become visible without an SSE subscription -// here (the parent useDashboardData owns the SSE connection). +// - Subscribes to SSE for the currently-selected call regardless of +// ``isLive`` so the transition from in-progress → ended is observed +// directly: +// * ``turn_complete`` (live calls) — refetch on each completed +// round-trip, primary signal during the call. +// * ``call_end`` — refetch once when the call hangs up so the pane +// picks up the SDK-authoritative ``history.entries`` transcript +// the moment ``recordCallEnd`` lands. Without this the pane could +// go blank in the race window between the carrier statusCallback +// (``completed``) and the WS-driven ``recordCallEnd``. See +// dashboard BUG 2. +// - When ``isLive`` is true, also polls the call detail endpoint every +// 2 s as a backstop in case SSE drops on a flaky network. +// - All SSE handlers refetch the full call rather than appending the +// event payload directly: the call detail endpoint is the single +// source of truth (it merges the active record's ``turns`` / +// ``transcript`` and any persisted ``transcript``), so we never have +// to reason about ordering or de-duplication on the client. import { useEffect, useRef, useState } from 'react'; import { fetchCall } from '../lib/api'; @@ -35,6 +50,7 @@ export function useTranscript( let cancelled = false; let pollTimer: ReturnType | null = null; + let source: EventSource | null = null; const load = async (): Promise => { try { @@ -46,13 +62,55 @@ export function useTranscript( } setTurns(toUiTranscript(record)); } catch { - // Swallow transient errors; the next poll tick will retry. The - // dashboard-level error surface lives in useDashboardData. + // Swallow transient errors; the next poll tick (or the next SSE + // event) will retry. Dashboard-level error surface lives in + // useDashboardData. } }; void load(); + // Filter SSE payloads to the currently-selected call. The SSE stream is + // a shared bus across every dashboard tab and every active call, so + // without this filter we'd refetch on every event of every unrelated + // call. ``MessageEvent.data`` is JSON: parse and compare ``call_id`` + // before triggering a reload. + const isForThisCall = (ev: Event): boolean => { + const messageEvent = ev as MessageEvent; + try { + const payload = JSON.parse(messageEvent.data) as { call_id?: unknown }; + return payload?.call_id === callId; + } catch { + return false; + } + }; + + try { + source = new EventSource('/api/dashboard/events'); + // ``turn_complete`` is the live-call signal; refetching keeps the pane + // in sync as turns accumulate. + source.addEventListener('turn_complete', (ev) => { + if (!isForThisCall(ev)) return; + void load(); + }); + // ``call_end`` fires the moment ``recordCallEnd`` lands the + // SDK-authoritative ``history.entries`` transcript onto the call + // record. Refetching here closes the race window where the pane + // would otherwise display the pre-end snapshot (potentially empty + // when the carrier statusCallback ran first). Subscribing + // unconditionally — even when ``isLive`` is false — covers the case + // where the call ended a moment after the user selected the row but + // before the dashboard list refreshed ``isLive`` to false. + source.addEventListener('call_end', (ev) => { + if (!isForThisCall(ev)) return; + void load(); + }); + } catch { + // EventSource not available (SSR / older browsers): the 2 s polling + // fallback below keeps the pane updated for live calls. + source = null; + } + if (isLive) { pollTimer = setInterval(() => { void load(); @@ -64,6 +122,9 @@ export function useTranscript( if (pollTimer !== null) { clearInterval(pollTimer); } + if (source !== null) { + source.close(); + } }; }, [callId, isLive]); diff --git a/dashboard-app/src/hooks/useUiPrefs.ts b/dashboard-app/src/hooks/useUiPrefs.ts new file mode 100644 index 00000000..d544a377 --- /dev/null +++ b/dashboard-app/src/hooks/useUiPrefs.ts @@ -0,0 +1,85 @@ +// Local-only UI preferences: PII reveal toggle + dark-mode toggle. +// +// Both prefs persist in localStorage so the operator's last choice +// survives a page reload. ``revealed`` defaults to ``false`` (PII hidden) +// so a freshly-opened dashboard is screen-share safe by default. ``dark`` +// defaults to ``false`` (light theme) to match the brand's cream palette. + +import { useCallback, useEffect, useState } from 'react'; + +const REVEAL_KEY = 'patter.dashboard.reveal'; +const THEME_KEY = 'patter.dashboard.theme'; // 'light' | 'dark' + +function readBool(key: string, fallback: boolean): boolean { + try { + const raw = window.localStorage.getItem(key); + if (raw === '1' || raw === 'true') return true; + if (raw === '0' || raw === 'false') return false; + return fallback; + } catch { + return fallback; + } +} + +function readTheme(): 'light' | 'dark' { + try { + const raw = window.localStorage.getItem(THEME_KEY); + if (raw === 'dark') return 'dark'; + if (raw === 'light') return 'light'; + } catch { + // ignore + } + return 'light'; +} + +export interface UiPrefs { + /** When true, render full PII (phone numbers, etc.) — eye-OPEN state. */ + readonly revealed: boolean; + /** When true, force the ``body.dark`` theme. */ + readonly dark: boolean; + readonly toggleRevealed: () => void; + readonly toggleDark: () => void; +} + +export function useUiPrefs(): UiPrefs { + const [revealed, setRevealed] = useState(() => + readBool(REVEAL_KEY, false), + ); + const [theme, setTheme] = useState<'light' | 'dark'>(() => readTheme()); + + // Persist + apply theme to document.body so the existing + // ``body.dark`` CSS overrides flip in lockstep with the toggle. + useEffect(() => { + try { + window.localStorage.setItem(REVEAL_KEY, revealed ? '1' : '0'); + } catch { + // ignore quota / privacy-mode errors — the toggle still works for + // the current session, it just doesn't persist. + } + }, [revealed]); + + useEffect(() => { + try { + window.localStorage.setItem(THEME_KEY, theme); + } catch { + // ignore + } + const cls = document.body.classList; + if (theme === 'dark') cls.add('dark'); + else cls.remove('dark'); + }, [theme]); + + const toggleRevealed = useCallback(() => { + setRevealed((v) => !v); + }, []); + const toggleDark = useCallback(() => { + setTheme((t) => (t === 'dark' ? 'light' : 'dark')); + }, []); + + return { + revealed, + dark: theme === 'dark', + toggleRevealed, + toggleDark, + }; +} diff --git a/dashboard-app/src/lib/api.ts b/dashboard-app/src/lib/api.ts index b4a8f804..b5e79f94 100644 --- a/dashboard-app/src/lib/api.ts +++ b/dashboard-app/src/lib/api.ts @@ -16,6 +16,7 @@ export interface CallCost { readonly llm?: number; readonly telephony?: number; readonly total?: number; + readonly llm_cached_savings?: number; } export interface CallLatency { @@ -23,6 +24,9 @@ export interface CallLatency { readonly llm_ms?: number; readonly tts_ms?: number; readonly total_ms?: number; + readonly agent_response_ms?: number; + readonly endpoint_ms?: number; + readonly user_speech_duration_ms?: number; } export interface CallMetrics { @@ -32,9 +36,14 @@ export interface CallMetrics { readonly stt_provider?: string; readonly tts_provider?: string; readonly llm_provider?: string; + readonly stt_model?: string; + readonly tts_model?: string; + readonly llm_model?: string; readonly cost?: CallCost; readonly latency_avg?: CallLatency; + readonly latency_p50?: CallLatency; readonly latency_p95?: CallLatency; + readonly latency_p99?: CallLatency; readonly turns?: readonly unknown[]; } @@ -89,6 +98,9 @@ function parseLatency(raw: unknown): CallLatency | undefined { llm_ms: asOptionalNumber(raw.llm_ms), tts_ms: asOptionalNumber(raw.tts_ms), total_ms: asOptionalNumber(raw.total_ms), + agent_response_ms: asOptionalNumber(raw.agent_response_ms), + endpoint_ms: asOptionalNumber(raw.endpoint_ms), + user_speech_duration_ms: asOptionalNumber(raw.user_speech_duration_ms), }; } @@ -100,6 +112,7 @@ function parseCost(raw: unknown): CallCost | undefined { llm: asOptionalNumber(raw.llm), telephony: asOptionalNumber(raw.telephony), total: asOptionalNumber(raw.total), + llm_cached_savings: asOptionalNumber(raw.llm_cached_savings), }; } @@ -113,9 +126,14 @@ function parseMetrics(raw: unknown): CallMetrics | null { stt_provider: asOptionalString(raw.stt_provider), tts_provider: asOptionalString(raw.tts_provider), llm_provider: asOptionalString(raw.llm_provider), + stt_model: asOptionalString(raw.stt_model), + tts_model: asOptionalString(raw.tts_model), + llm_model: asOptionalString(raw.llm_model), cost: parseCost(raw.cost), latency_avg: parseLatency(raw.latency_avg), + latency_p50: parseLatency(raw.latency_p50), latency_p95: parseLatency(raw.latency_p95), + latency_p99: parseLatency(raw.latency_p99), turns: Array.isArray(turnsRaw) ? (turnsRaw as readonly unknown[]) : undefined, }; } @@ -237,3 +255,48 @@ export async function fetchCall(callId: string): Promise { const body = (await response.json()) as unknown; return parseCallRecord(body); } + +/** + * Soft-delete a batch of calls. The server keeps the on-disk metadata + + * transcript files as a backup; the dashboard view and aggregate metrics + * exclude the ids going forward. Active calls are silently dropped from + * the request. Idempotent. + * + * Returns the call_ids actually accepted (already-deleted / active ids + * are filtered server-side). + */ +export async function deleteCalls(callIds: readonly string[]): Promise { + if (callIds.length === 0) return []; + if (callIds.length === 1) { + const url = `/api/dashboard/calls/${encodeURIComponent(callIds[0])}`; + const response = await fetch(url, { + method: 'DELETE', + headers: { Accept: 'application/json' }, + }); + if (!response.ok) { + throw new Error(`DELETE ${url} failed with status ${response.status}`); + } + const body = (await response.json()) as { deleted?: unknown }; + return Array.isArray(body.deleted) + ? (body.deleted as unknown[]).filter( + (v): v is string => typeof v === 'string', + ) + : []; + } + const response = await fetch('/api/dashboard/calls/delete', { + method: 'POST', + headers: { 'Content-Type': 'application/json', Accept: 'application/json' }, + body: JSON.stringify({ call_ids: callIds }), + }); + if (!response.ok) { + throw new Error( + `POST /api/dashboard/calls/delete failed with status ${response.status}`, + ); + } + const body = (await response.json()) as { deleted?: unknown }; + return Array.isArray(body.deleted) + ? (body.deleted as unknown[]).filter( + (v): v is string => typeof v === 'string', + ) + : []; +} diff --git a/dashboard-app/src/lib/mappers.ts b/dashboard-app/src/lib/mappers.ts index c137457b..1a7297be 100644 --- a/dashboard-app/src/lib/mappers.ts +++ b/dashboard-app/src/lib/mappers.ts @@ -18,6 +18,13 @@ export type CallMode = 'realtime' | 'pipeline' | 'convai' | 'unknown'; export interface CallCostUi { readonly telco?: number; readonly llm?: number; + readonly stt?: number; + readonly tts?: number; + /** + * @deprecated Sum of stt+tts kept for legacy consumers. New code reads + * ``stt`` and ``tts`` separately so the dashboard can label each with the + * actual provider (e.g. "Cartesia STT" / "ElevenLabs TTS"). + */ readonly sttTts?: number; readonly cached?: number; readonly total?: number; @@ -35,12 +42,26 @@ export interface Call { readonly duration?: number; readonly latencyP95?: number; readonly latencyP50?: number; + /** avg(llm_ms) across this call's turns — for the waterfall llm bar. */ + readonly llmAvg?: number; readonly sttAvg?: number; readonly ttsAvg?: number; + /** Number of completed turns. p50/p95 are statistically meaningful only when this is >= 5. */ + readonly turnCount?: number; + /** p50 of agent_response_ms (wait time after user stops speaking). */ + readonly agentResponseP50?: number; + /** p95 of agent_response_ms — user-perceived latency outlier. */ + readonly agentResponseP95?: number; readonly cost: CallCostUi; readonly agent?: string; readonly model?: string; readonly mode?: CallMode; + readonly sttProvider?: string; + readonly ttsProvider?: string; + /** Model identifier within the provider (e.g. "ink-whisper"). */ + readonly sttModel?: string; + readonly ttsModel?: string; + readonly llmModel?: string; readonly transcriptKey?: string; readonly endedAgo?: number; } @@ -122,6 +143,8 @@ function computeCost(record: CallRecord): CallCostUi { const result: { telco?: number; llm?: number; + stt?: number; + tts?: number; sttTts?: number; cached?: number; total?: number; @@ -129,9 +152,13 @@ function computeCost(record: CallRecord): CallCostUi { if (typeof cost.telephony === 'number') result.telco = cost.telephony; if (typeof cost.llm === 'number') result.llm = cost.llm; - - if (typeof cost.stt === 'number' || typeof cost.tts === 'number') { - result.sttTts = (cost.stt ?? 0) + (cost.tts ?? 0); + if (typeof cost.stt === 'number') result.stt = cost.stt; + if (typeof cost.tts === 'number') result.tts = cost.tts; + if (typeof cost.llm_cached_savings === 'number') { + result.cached = cost.llm_cached_savings; + } + if (result.stt !== undefined || result.tts !== undefined) { + result.sttTts = (result.stt ?? 0) + (result.tts ?? 0); } // Only fall back to total when no granular breakdown is available. @@ -166,7 +193,14 @@ export function toUiCall(record: CallRecord): Call { const status = mapStatus(record.status); const isLive = status === 'live' || (record.status !== undefined && LIVE_STATUSES.has(record.status)); const latencyAvg = record.metrics?.latency_avg; + const latencyP50 = record.metrics?.latency_p50; const latencyP95 = record.metrics?.latency_p95; + // Total turn count from runtime metrics (preferred) — falls back to + // the persisted transcript length for hydrated rows. Percentile boxes + // are hidden in the UI when turnCount < 5 (statistical floor). + const turnCount = + (Array.isArray(record.metrics?.turns) ? record.metrics?.turns?.length : undefined) ?? + (Array.isArray(record.transcript) ? record.transcript.length : undefined); const call: Call = { id: record.call_id, @@ -178,14 +212,29 @@ export function toUiCall(record: CallRecord): Call { startedAtMs: typeof record.started_at === 'number' ? record.started_at * 1000 : undefined, durationStart: isLive ? record.started_at * 1000 : undefined, duration: computeDuration(record, isLive), - latencyP95: latencyP95?.total_ms ?? latencyAvg?.total_ms, - latencyP50: latencyAvg?.total_ms, + // User-perceived "latency" on the dashboard means wait-time AFTER the + // caller stops speaking (a.k.a. agent_response_ms / "response latency" + // — the metric Pipecat, LiveKit, and OpenAI Realtime all surface). + // Falls back to total_ms only for legacy rows that don't carry the + // agent_response_ms breakdown — those rows over-state perceived + // latency by the user-utterance duration but keep the table populated. + latencyP95: latencyP95?.agent_response_ms ?? latencyP95?.total_ms ?? latencyAvg?.total_ms, + latencyP50: latencyP50?.agent_response_ms ?? latencyP50?.total_ms ?? latencyAvg?.total_ms, sttAvg: latencyAvg?.stt_ms, ttsAvg: latencyAvg?.tts_ms, + llmAvg: latencyAvg?.llm_ms, + turnCount, + agentResponseP50: latencyP50?.agent_response_ms, + agentResponseP95: latencyP95?.agent_response_ms, cost: computeCost(record), agent: buildAgentLabel(record), model: record.metrics?.llm_provider, mode: mapMode(record.metrics?.provider_mode), + sttProvider: record.metrics?.stt_provider, + ttsProvider: record.metrics?.tts_provider, + sttModel: record.metrics?.stt_model, + ttsModel: record.metrics?.tts_model, + llmModel: record.metrics?.llm_model, transcriptKey: record.call_id, endedAgo: computeEndedAgo(record), }; @@ -194,26 +243,46 @@ export function toUiCall(record: CallRecord): Call { export function toUiTranscript(record: CallRecord): TranscriptTurn[] { const transcript = record.transcript; - if (!transcript) return []; - const turns: TranscriptTurn[] = []; - for (const entry of transcript) { - const text = entry.text; - switch (entry.role) { - case 'user': - turns.push({ who: 'user', txt: text }); - break; - case 'assistant': - turns.push({ who: 'bot', txt: text }); - break; - case 'tool': - turns.push({ who: 'tool', txt: text }); - break; - default: - turns.push({ who: 'bot', txt: text }); - break; + if (transcript && transcript.length > 0) { + const out: TranscriptTurn[] = []; + for (const entry of transcript) { + const text = entry.text; + switch (entry.role) { + case 'user': + out.push({ who: 'user', txt: text }); + break; + case 'assistant': + out.push({ who: 'bot', txt: text }); + break; + case 'tool': + out.push({ who: 'tool', txt: text }); + break; + default: + out.push({ who: 'bot', txt: text }); + break; + } + } + return out; + } + // Fallback for live calls: completed calls expose ``transcript`` (a flat + // array of {role,text}) but in-flight calls expose ``turns`` (the + // ``TurnMetrics`` shape — one entry per round-trip with both + // ``user_text`` and ``agent_text``). Without this branch the live + // transcript pane is empty until the call ends. See dashboard BUG A. + const turns = record.turns; + if (!turns || turns.length === 0) return []; + const out: TranscriptTurn[] = []; + for (const raw of turns) { + if (typeof raw !== 'object' || raw === null) continue; + const turn = raw as { user_text?: unknown; agent_text?: unknown }; + const userText = typeof turn.user_text === 'string' ? turn.user_text : ''; + const agentText = typeof turn.agent_text === 'string' ? turn.agent_text : ''; + if (userText.length > 0) out.push({ who: 'user', txt: userText }); + if (agentText.length > 0 && agentText !== '[interrupted]') { + out.push({ who: 'bot', txt: agentText }); } } - return turns; + return out; } export type SparklineField = 'totalCalls' | 'latency' | 'spend'; diff --git a/dashboard-app/src/styles/dashboard.css b/dashboard-app/src/styles/dashboard.css index 575e1723..626b14f4 100644 --- a/dashboard-app/src/styles/dashboard.css +++ b/dashboard-app/src/styles/dashboard.css @@ -16,8 +16,14 @@ button{font-family:inherit;color:inherit} .pulse.active{background:#3b6f3b;box-shadow:0 0 0 0 rgba(59,111,59,.6);animation:pulse 1.5s infinite} @keyframes pulse{0%{box-shadow:0 0 0 0 rgba(59,111,59,.6)}70%{box-shadow:0 0 0 8px rgba(59,111,59,0)}100%{box-shadow:0 0 0 0 rgba(59,111,59,0)}} .num-chip{font-family:var(--font-mono);font-size:13px;background:#fff;border:1px solid var(--line);padding:5px 12px;border-radius:8px;color:#000;font-weight:500} -.icon-btn{width:32px;height:32px;border-radius:8px;border:1px solid var(--line);background:#fff;display:inline-flex;align-items:center;justify-content:center;cursor:pointer;color:#1a1a1a;transition:background var(--motion-fast) var(--easing)} -.icon-btn:hover{background:#fafaf8} +.icon-btn{width:32px;height:32px;border-radius:8px;border:1px solid var(--line);background:#fff;display:inline-flex;align-items:center;justify-content:center;cursor:pointer;color:#1a1a1a;transition:background var(--motion-fast) var(--easing),border-color var(--motion-fast) var(--easing),color var(--motion-fast) var(--easing)} +.icon-btn:hover{background:#fafaf8;border-color:#cbcbcb} +.icon-btn:focus-visible{outline:2px solid var(--peach);outline-offset:2px} +.icon-btn.toggle.on{background:#000;border-color:#000;color:#fff} +.icon-btn.toggle.on:hover{background:#1a1a1a;border-color:#1a1a1a} +/* PII cells use tabular numerics so masked + revealed states share a column + width — switching the eye toggle does not jitter the column. */ +.pii{font-variant-numeric:tabular-nums} .avatar{width:32px;height:32px;border-radius:999px;background:var(--peach);border:1.5px solid #000;display:flex;align-items:center;justify-content:center;font-family:var(--font-mono);font-size:11px;font-weight:600} /* page */ @@ -63,7 +69,9 @@ button{font-family:inherit;color:inherit} .spark-tooltip::after{content:"";position:absolute;top:100%;left:50%;transform:translateX(-50%);border:5px solid transparent;border-top-color:var(--ink)} @keyframes spark-tooltip-in{from{opacity:0;transform:translateX(-50%) translateY(-2px)}to{opacity:1;transform:translateX(-50%) translateY(0)}} .spark-tooltip-range{font-family:var(--font-mono);font-size:10px;letter-spacing:.06em;text-transform:uppercase;color:#aaa;margin-bottom:4px} -.spark-tooltip-count{font-weight:600;font-size:13px;margin-bottom:6px} +.spark-tooltip-headline{font-family:var(--font-mono);font-size:12px;display:flex;justify-content:space-between;align-items:baseline;gap:10px;margin-bottom:6px} +.spark-tooltip-headline-l{font-size:10px;letter-spacing:.06em;text-transform:uppercase;color:#aaa;font-weight:500} +.spark-tooltip-headline-v{color:#fff;font-weight:600;font-variant-numeric:tabular-nums} .spark-tooltip-empty{color:#aaa;font-style:italic;font-size:12px} .spark-tooltip-list{list-style:none;margin:0;padding:0;display:flex;flex-direction:column;gap:3px} .spark-tooltip-list li{display:grid;grid-template-columns:1fr auto auto;gap:8px;align-items:baseline;font-family:var(--font-mono);font-size:11px} @@ -99,6 +107,104 @@ tbody tr:last-child td{border-bottom:0} tbody tr.new-row{animation:slideIn 400ms var(--easing)} @keyframes slideIn{from{background:#fff8ef;transform:translateY(-4px);opacity:0}to{background:transparent;transform:translateY(0);opacity:1}} +/* ── Bulk select / delete ───────────────────────────────────────── + Refined-minimal: hand-drawn-feeling square checkboxes (1.5px hard + border), peach destructive accent reserved for the actual confirm + step. Header check and per-row check share .row-check styles so + hit-targets are visually consistent. */ +.call-table th.check-cell, +.call-table td.check-cell{ + width:38px;padding-left:14px;padding-right:0;text-align:left; +} +.call-table td.check-cell{cursor:default} +.row-check{ + appearance:none;display:inline-flex;align-items:center;justify-content:center; + width:16px;height:16px;border:1.5px solid #cbcbcb;border-radius:4px; + background:#fff;color:transparent;cursor:pointer;padding:0; + transition:border-color var(--motion-fast) var(--easing), + background var(--motion-fast) var(--easing), + color var(--motion-fast) var(--easing), + box-shadow var(--motion-fast) var(--easing); +} +.row-check:hover:not(.disabled){border-color:#000} +.row-check:focus-visible{outline:2px solid var(--peach);outline-offset:2px} +.row-check.on{background:#000;border-color:#000;color:#fff} +.row-check.on:hover{background:#1a1a1a} +.row-check.disabled{cursor:not-allowed;opacity:.35;background:#fafaf8} +.row-check.head.indet{background:#fff;border-color:#000;color:transparent} +.row-check.head .indet-mark{ + width:8px;height:1.5px;background:#000;border-radius:1px;display:block; +} +.row-check svg{display:block;width:11px;height:11px} +tbody tr.checked{background:#fbf3eb} +tbody tr.checked:hover{background:#f7ead8} +tbody tr.checked.selected{background:#fff8ef} +tbody tr.checked.selected td:first-child{box-shadow:inset 3px 0 0 #DF9367} + +.bulk-bar{ + display:flex;align-items:center;gap:10px; + padding:10px 16px 10px 12px; + background:#fafaf8;border-bottom:1px solid var(--line); + font-family:var(--font-sans);font-size:13px;color:#1a1a1a; + animation:bulkSlideIn 180ms var(--easing) both; +} +@keyframes bulkSlideIn{from{transform:translateY(-2px);opacity:0}to{transform:translateY(0);opacity:1}} +.bulk-bar.confirming{background:#fff4e8;border-bottom-color:#EFC5AC} +.bulk-count{display:inline-flex;align-items:baseline;gap:6px} +.bulk-count .bulk-num{ + font-family:var(--font-mono);font-size:14px;font-weight:600; + font-variant-numeric:tabular-nums;color:#000; +} +.bulk-count .bulk-lbl{ + font-family:var(--font-mono);font-size:11px;letter-spacing:.04em; + text-transform:uppercase;color:#4a4a4a;font-weight:500; +} +.bulk-spacer{flex:1} +.bulk-warn{ + font-family:var(--font-mono);font-size:11px;letter-spacing:.02em; + color:#7a5b3e;margin-right:4px; +} +.bulk-btn{ + display:inline-flex;align-items:center;gap:6px; + font-family:var(--font-mono);font-size:12px;font-weight:500; + height:30px;padding:0 12px;border-radius:8px;cursor:pointer; + border:1px solid var(--line);background:#fff;color:#1a1a1a; + transition:background var(--motion-fast) var(--easing), + color var(--motion-fast) var(--easing), + border-color var(--motion-fast) var(--easing), + transform var(--motion-fast) var(--easing); +} +.bulk-btn:hover:not(:disabled){background:#fafaf8;border-color:#cbcbcb} +.bulk-btn:disabled{opacity:.5;cursor:not-allowed} +.bulk-btn:focus-visible{outline:2px solid var(--peach);outline-offset:2px} +.bulk-btn.ghost{background:transparent;border-color:transparent;color:#4a4a4a} +.bulk-btn.ghost:hover:not(:disabled){background:#fff;border-color:var(--line);color:#000} +.bulk-btn.destructive{ + background:var(--peach);color:#000;border-color:var(--peach-deep); +} +.bulk-btn.destructive:hover:not(:disabled){background:var(--peach-deep);color:#fff;border-color:var(--peach-deep)} +.bulk-btn.destructive:active:not(:disabled){transform:translateY(1px)} +.bulk-btn svg{display:block} + +/* dark mode */ +body.dark .row-check{background:#0d0d0d;border-color:#3a3a3a} +body.dark .row-check:hover:not(.disabled){border-color:#fff} +body.dark .row-check.on{background:#fff;border-color:#fff;color:#000} +body.dark .row-check.disabled{background:#0a0a0a} +body.dark .row-check.head.indet .indet-mark{background:#fff} +body.dark tbody tr.checked{background:#1a1410} +body.dark tbody tr.checked:hover{background:#221911} +body.dark .bulk-bar{background:#0d0d0d;border-bottom-color:#262626;color:#e5e5e5} +body.dark .bulk-bar.confirming{background:#1f1611;border-bottom-color:#5a3f29} +body.dark .bulk-count .bulk-num{color:#fff} +body.dark .bulk-count .bulk-lbl{color:#888} +body.dark .bulk-warn{color:#d4a578} +body.dark .bulk-btn{background:#171717;border-color:#262626;color:#e5e5e5} +body.dark .bulk-btn:hover:not(:disabled){background:#1f1f1f;border-color:#3a3a3a} +body.dark .bulk-btn.ghost{background:transparent;border-color:transparent;color:#888} +body.dark .bulk-btn.ghost:hover:not(:disabled){color:#fff;border-color:#262626;background:#171717} +body.dark .bulk-btn.destructive{background:var(--peach);color:#000;border-color:var(--peach-deep)} + .num-cell{font-family:var(--font-mono);font-size:13px;font-variant-numeric:tabular-nums} .pill{display:inline-flex;align-items:center;gap:5px;font-family:var(--font-mono);font-size:11px;font-weight:500;padding:3px 9px;border-radius:999px;border:1px solid;white-space:nowrap} .pill.live{background:#fff;border-color:#000;color:#000} @@ -120,9 +226,16 @@ tbody tr.new-row{animation:slideIn 400ms var(--easing)} .dir.out{color:#4a4a4a} /* right rail: live call */ -.rr{display:flex;flex-direction:column;gap:14px} -.rr-card{background:#fff;border:1px solid var(--line);border-radius:14px;padding:18px} +.rr{display:flex;flex-direction:column;gap:14px;min-height:590px} +/* Baseline panel height for the right column (Live call + Metrics): + prevents the card from collapsing when no call is selected and keeps + the layout anchored at the same height as the recent-calls table + (header ~50px + 540px scroll area ≈ 590px). */ +.rr-card{background:#fff;border:1px solid var(--line);border-radius:14px;padding:18px;min-height:280px} .metrics-panel-h{margin-bottom:14px} +/* Lock the body area to the tallest layout (pipeline latency: 4 latboxes + + waterfall + legend ≈ 230px) so switching Latency↔Cost tabs doesn't jump. */ +.metrics-panel-body{min-height:240px} .metrics-panel-h .seg{display:inline-flex;background:#fafaf8;border:1px solid var(--line);border-radius:8px;padding:2px} .metrics-panel-h .seg button{font-family:var(--font-mono);font-size:11px;border:0;background:transparent;padding:5px 14px;border-radius:6px;cursor:pointer;color:#4a4a4a;font-weight:500;transition:all var(--motion-fast) var(--easing);text-transform:uppercase;letter-spacing:.06em} .metrics-panel-h .seg button.on{background:#000;color:#fff} @@ -222,27 +335,165 @@ body.dense tbody td{padding:9px 18px} body.dense .rr-card{padding:14px} body.dense .turn{margin-bottom:9px;font-size:12.5px} -/* dark scheme (subtle, optional) */ -body.dark{background:#0d0d0d;color:#e8e8e8} -body.dark .top,body.dark .panel,body.dark .metric,body.dark .rr-card,body.dark .num-chip,body.dark .icon-btn,body.dark .seg,body.dark .btn,body.dark .latbox,body.dark .ctrl,body.dark .duration-block,body.dark .statusbar{background:#171717;border-color:#262626;color:#e8e8e8} -body.dark thead th{background:#0d0d0d;color:#888;border-color:#262626} -body.dark tbody td,body.dark .rr-card{border-color:#262626} -body.dark tbody tr:hover{background:#1a1a1a} -body.dark tbody tr.selected{background:#1f1611} -body.dark .pill.done{background:#0d0d0d;color:#888;border-color:#262626} -body.dark .pill.live{background:#171717;color:#fff;border-color:#fff} +/* dark scheme (subtle, optional) + Palette lifted on 2026-05-14: page #0d0d0d → #121212, cards + #171717 → #1c1c1c, borders #262626 → #2a2a2a. The previous + pitch-black felt oppressive against the brand's cream/peach + accent. Two-step neutral ramp keeps cards distinct from the + page without contrast loss on borders. */ +body.dark{background:#121212;color:#e8e8e8} +body.dark .top,body.dark .panel,body.dark .metric,body.dark .rr-card,body.dark .num-chip,body.dark .icon-btn,body.dark .seg,body.dark .btn,body.dark .latbox,body.dark .ctrl,body.dark .duration-block,body.dark .statusbar{background:#1c1c1c;border-color:#2a2a2a;color:#e8e8e8} +body.dark thead th{background:#121212;color:#888;border-color:#2a2a2a} +body.dark tbody td,body.dark .rr-card{border-color:#2a2a2a} +body.dark tbody tr:hover{background:#222222} +body.dark tbody tr.selected{background:#2a1d15} +body.dark .pill.done{background:#121212;color:#a0a0a0;border-color:#2a2a2a} +body.dark .pill.live{background:#1c1c1c;color:#fff;border-color:#fff} body.dark .turn .av{border-color:#888} -body.dark .turn.user .av{background:#262626;color:#fff} -body.dark .lat-bar{background:#262626} -body.dark .lat-bar i{background:#fff} -body.dark .stack-row{border-color:#262626} -body.dark .panel-h .search{background:#0d0d0d;border-color:#262626} +body.dark .turn.user .av{background:#2a2a2a;color:#fff} +body.dark .lat-bar{background:#2a2a2a} +body.dark .lat-bar i{background:#e8e8e8} +body.dark .lat-bar.warn i{background:var(--peach)} +body.dark .stack-row{border-color:#2a2a2a} +body.dark .panel-h .search{background:#121212;border-color:#2a2a2a} body.dark .panel-h .search input{color:#e8e8e8} -body.dark .metric .val,body.dark .latbox .v,body.dark .num-cell{color:#fff} -body.dark .seg button.on{background:#fff;color:#000} -body.dark .ctrl.danger{background:#fff;color:#000;border-color:#fff} -body.dark .brand .tag{background:#171717;color:#888;border-color:#262626} -body.dark .metric.peach{background:#1f1611;border-color:#5a3a2a} -body.dark .rr-card.peach{background:#1f1611;border-color:#5a3a2a} -body.dark .duration-block,body.dark .latbox{background:#0d0d0d} +body.dark .metric .val,body.dark .latbox .v,body.dark .num-cell,body.dark .duration-block .v{color:#fff} +body.dark .duration-block .agent{color:#a8a8a8} +body.dark .duration-block .l{color:#888} +/* Active "pill" toggles + theme toggle: peach accent in dark mode (was stark + white — felt like a light-mode leftover floating on the dark page). */ +body.dark .seg button.on{background:var(--peach);color:#000;border-color:var(--peach)} +body.dark .icon-btn{color:#e8e8e8} +body.dark .icon-btn:hover{background:#262626;border-color:#3a3a3a} +body.dark .icon-btn.toggle.on{background:var(--peach);border-color:var(--peach-deep);color:#000} +body.dark .icon-btn.toggle.on:hover{background:var(--peach-deep);border-color:var(--peach-deep);color:#fff} +body.dark .ctrl.danger{background:var(--peach);color:#000;border-color:var(--peach-deep)} +body.dark .brand .tag{background:#1c1c1c;color:#a0a0a0;border-color:#2a2a2a} +body.dark .metric.peach{background:#241814;border-color:#5a3a2a} +body.dark .rr-card.peach{background:#241814;border-color:#5a3a2a} +body.dark .duration-block,body.dark .latbox{background:#121212} body.dark tbody tr.selected td:first-child{box-shadow:inset 3px 0 0 #DF9367} + +/* ── Dark mode polish ──────────────────────────────────────────── + Default light styles use ink (#000) for spark bars, dark grey + (#4a4a4a) for secondary text, and cream (#fafaf8) for chip + surfaces — all unreadable on a dark page. Overrides below fix + contrast across every surface that the screenshot revealed as + muddy: sparklines, kbd chip, "X inbound · X outbound" footer, + carrier label, page subtitle, metric delta, table empty state, + peach metric card, and the latency bar fill. */ +body.dark .ph h1{color:#fff} +body.dark .ph .sub{color:#9a9a9a} +body.dark .kbd{background:#0d0d0d;border-color:#3a3a3a;color:#cfcfcf;border-bottom-color:#262626} +body.dark .metric .delta{color:#9a9a9a} +body.dark .metric .lbl{color:#8a8a8a} +body.dark .metric .footer, +body.dark .metric .meta-foot{color:#a8a8a8} +/* Spark bars: the default ``var(--ink)`` (#000) disappears on the + dark page. Use a light grey (with a subtle white-on-hover) so the + sparkline is legible without screaming. ``:not(.empty)`` is + important — without it the empty (zero-height) bars inherit the + light fill and become a visible grey strip at the bottom of each + card. */ +body.dark .metric .spark .spark-bar-static, +body.dark .metric .spark .spark-bar:not(.empty){background:#a8a8a8;opacity:.85} +body.dark .metric .spark .spark-bar:not(.empty):hover{background:var(--peach);opacity:1} +body.dark .metric.peach .spark .spark-bar-static, +body.dark .metric.peach .spark .spark-bar:not(.empty){background:var(--peach);opacity:.85} +body.dark .metric.peach .spark .spark-bar:not(.empty):hover{background:#fff;opacity:1} +body.dark .metric.peach .lbl, +body.dark .metric.peach .delta{color:#c9a584} +body.dark .car-tw{color:#a8a8a8} +body.dark .empty{color:#666} +body.dark .lat-bar{background:#2a2a2a} +body.dark .lat-bar i{background:#e8e8e8} +body.dark .lat-bar.warn i{background:var(--peach-deep)} +body.dark .btn.primary{background:#fff;color:#000;border-color:#fff} +body.dark .btn.primary:hover{background:#e8e8e8;border-color:#e8e8e8} +body.dark .btn:hover{background:#1f1f1f;border-color:#3a3a3a} +body.dark .seg{background:#0d0d0d;border-color:#262626} +body.dark .seg button{color:#888} +body.dark .seg button:not(.on):hover{color:#fff} +body.dark .panel-h h3{color:#fff} +body.dark .panel-h .sse{color:#888} +body.dark .panel-h .search svg{color:#666} +body.dark .panel-h .search input::placeholder{color:#666} +body.dark .num-chip{color:#e8e8e8} +body.dark .top-r .live-chip{color:#cfcfcf} +body.dark .statusbar{color:#888} +body.dark .statusbar .green{color:#76c777} +body.dark .meta, +body.dark .meta .sep{color:#9a9a9a} +body.dark .pii{color:#e8e8e8} +/* Live-call panel ctrl toggles (Recording, Mute, …): the light-peach + ``.ctrl.active`` state has higher specificity than the generic + ``body.dark .ctrl`` so we need explicit dark variants for active + + danger + hover or the panel sits as a cream blob on the dark page. */ +body.dark .ctrl{background:#1c1c1c;color:#e8e8e8;border-color:#2a2a2a} +body.dark .ctrl:hover{background:#222;border-color:#3a3a3a} +body.dark .ctrl.active{background:#2a1d15;border-color:var(--peach);color:var(--peach-light)} +body.dark .ctrl.active:hover{background:#3a2a1f} +body.dark .ctrl.danger{background:var(--peach);color:#000;border-color:var(--peach-deep)} +body.dark .ctrl.danger:hover{background:var(--peach-deep);color:#fff;border-color:var(--peach-deep)} +/* Queued / no-answer / fail pills had cool-blue / cream backgrounds. */ +body.dark .pill.queued{background:#1c1c1c;border-color:#2a2a2a;color:#9a9a9a} +body.dark .pill.queued::before{background:#666} +body.dark .pill.fail, +body.dark .pill.noanswer{background:#2a1d15;border-color:#5a3a2a;color:var(--peach-light)} +/* New-row flash animation used the light cream end-state — pick a dark + neutral so a freshly inserted row in dark mode doesn't flash white. */ +body.dark tbody tr.new-row{animation:slideInDark 400ms var(--easing)} +@keyframes slideInDark{from{background:#2a1d15;transform:translateY(-4px);opacity:0}to{background:transparent;transform:translateY(0);opacity:1}} +/* Carrier label colour drift between rows + table empty state. */ +body.dark .empty{color:#666;background:transparent} +body.dark .car-tw{color:#a8a8a8} +body.dark .car-dot.tx{background:#5a96f0} +/* Selected + checked rows on the table — without these explicit dark + tones the warm peach overlay clashes with the lifted #1c1c1c card. */ +body.dark tbody tr.checked{background:#2a1d15} +body.dark tbody tr.checked:hover{background:#3a2a1f} +body.dark tbody tr.checked.selected{background:#3a2a1f} +/* Transcript turn body text — default ``#1a1a1a`` is invisible on the + ``#1c1c1c`` card. Latency caption already uses ``var(--fg-tertiary)`` + which resolves to a readable muted grey, but the actual conversation + text needs an explicit dark override. Also bump the avatar palette so + the user/agent rings stand against the lifted card surface. */ +body.dark .turn .body .txt{color:#e8e8e8} +body.dark .turn .body .who{color:#a8a8a8} +body.dark .turn .body .lat{color:#888} +body.dark .turn .av{border-color:#3a3a3a} +body.dark .turn.user .av{background:#2a2a2a;color:#fff;border-color:#3a3a3a} +body.dark .turn.bot .av{color:#000} +body.dark .turn.tool .av{background:#1a2436;color:#93c5fd;border-color:#264062} +body.dark .turn.tool .body .who{color:#93c5fd} +/* Waterfall — the light cream track + near-black STT bar + black value + caption all collapse to invisible/illegible on the dark card. Lift + them: dark track, soft-white STT, peach LLM (unchanged), cool-blue + TTS (unchanged), readable label/value text. */ +body.dark .wf-row .lbl{color:#a8a8a8} +body.dark .wf-row .track{background:#2a2a2a} +body.dark .wf-row .seg-bar{background:#e8e8e8} +body.dark .wf-row .seg-bar.stt{background:#e8e8e8} +body.dark .wf-row .seg-bar.llm{background:var(--peach)} +body.dark .wf-row .seg-bar.tts{background:#5a96f0} +body.dark .wf-row .v{color:#e8e8e8} +body.dark .wf-legend{color:#a8a8a8} +/* Stack rows (latency/cost breakdown) used cream-bordered dashed + separators; replace with the dark border palette + readable label + colour so the "stt / llm / tts" labels don't sink into the bg. */ +body.dark .stack-row{border-color:#2a2a2a} +body.dark .stack-row:last-child{border-top-color:#2a2a2a} +body.dark .stack-row .lbl{color:#a8a8a8} +body.dark .stack-row .v{color:#e8e8e8} +body.dark .stack-row .saved{color:#7fb787} +/* latbox label/value were already mostly OK but the ``.warn`` variant + used a cream background that clashes — lift it to the warm-dark tone + matching the peach metric card. */ +body.dark .latbox.warn{background:#241814;border-color:#5a3a2a} +body.dark .latbox.warn .v{color:var(--peach-light)} +/* Patter logo SVGs inherit ``currentColor`` from the ``.brand`` + ancestor. Force the brand colour to bright white in dark mode so the + black wordmark in ``light.svg`` (stroked via ``currentColor``) + reverses to a legible mark. */ +body.dark .brand, +body.dark .patter-logo{color:#f4f4f4} diff --git a/dashboard-app/vitest.config.ts b/dashboard-app/vitest.config.ts new file mode 100644 index 00000000..171c2ca0 --- /dev/null +++ b/dashboard-app/vitest.config.ts @@ -0,0 +1,11 @@ +import { defineConfig } from 'vitest/config'; + +// Vitest config separate from vite.config.ts so the SPA build (singlefile +// inline) is unaffected by the test runner. Only ``test`` files are picked +// up; the bundle still ships from src/ via vite.config.ts. +export default defineConfig({ + test: { + environment: 'node', + include: ['src/**/*.test.ts', 'src/**/*.test.tsx'], + }, +}); diff --git a/docs/python-sdk/events.mdx b/docs/python-sdk/events.mdx index 1d2703e4..11816ea3 100644 --- a/docs/python-sdk/events.mdx +++ b/docs/python-sdk/events.mdx @@ -305,7 +305,7 @@ asyncio.run(main()) The callbacks above describe the *transcript-level* lifecycle of a call. For **turn-taking instrumentation** — barge-in, end-of-utterance, time-to-first-token, TTS warmup vs. wire-time — Patter exposes seven additional async callbacks plus a read-only `conversation_state` snapshot directly on the `Patter` instance. -These events mirror the public APIs of [LiveKit Agents](https://docs.livekit.io/agents/) (`user_state_changed`, `agent_state_changed`, `user_turn_completed`, `user_interruption_detected`), [Pipecat](https://docs.pipecat.ai/) (`VADUserStartedSpeakingFrame`, `BotStartedSpeakingFrame`, `LLMFullResponseStartFrame`, `OutputAudioRawFrame`, `InterruptionFrame`) and [OpenAI Realtime](https://platform.openai.com/docs/guides/realtime) (`input_audio_buffer.speech_started/_stopped/_committed`) so downstream metrics map onto the canonical voice-agent metric set without translation. +These events expose the canonical voice-agent metric set (user/agent state transitions, turn boundaries, TTFT, audio first-byte) and align with [OpenAI Realtime](https://platform.openai.com/docs/guides/realtime) (`input_audio_buffer.speech_started/_stopped/_committed`) so downstream metrics work without translation. Every callback defaults to `None`. Existing code that does not register any speech-edge callback sees exactly the previous behaviour and zero overhead. The state machine is updated regardless of whether callbacks are registered, so `conversation_state` is always usable. diff --git a/docs/python-sdk/tracing.mdx b/docs/python-sdk/tracing.mdx index a7e790b0..ebd7db2e 100644 --- a/docs/python-sdk/tracing.mdx +++ b/docs/python-sdk/tracing.mdx @@ -48,6 +48,39 @@ All TS spans use the same `getpatter.*` namespace as Python (the legacy `patter. Patter never exports user utterances, tool payloads, or LLM content as span attributes. Only sizes, counts, and identifiers are emitted — traces are safe to ship to a shared Jaeger / Honeycomb / Grafana Cloud instance. +## Cost and latency attributes (`patter.*`) + +Beyond the spans listed above, every billable hot path stamps `patter.cost.*` and `patter.latency.*` attributes on its span starting in 0.6.0. External aggregators (e.g. the [patter-agent-runner acceptance suite](https://github.com/PatterAI/patter-agent-runner)) read these directly to compute per-call USD and latency without touching the SDK's pricing table. + +| Attribute | Type | Where it fires | +|---|---|---| +| `patter.call_id`, `patter.side` | str | Every cost/latency span (routing tags) | +| `patter.cost.telephony_minutes`, `patter.telephony`, `patter.direction` | float / str | Twilio + Telnyx adapters on call end | +| `patter.cost.stt_seconds`, `patter.stt.provider` | float / str | Each final transcript (Deepgram, AssemblyAI, Whisper, OpenAI-Transcribe, Soniox, Speechmatics, Cartesia) | +| `patter.cost.tts_chars`, `patter.tts.provider` | int / str | Each synthesis (ElevenLabs, OpenAI, Cartesia, LMNT, Rime) | +| `patter.cost.llm_input_tokens`, `patter.cost.llm_output_tokens`, `patter.llm.provider` | int / int / str | Each completion (OpenAI, Anthropic, Google Gemini, Groq, Cerebras) | +| `patter.cost.realtime_minutes`, `patter.realtime.provider` | float / str | OpenAI Realtime + ElevenLabs ConvAI on session end | +| `patter.latency.ttfb_ms`, `patter.latency.turn_ms` | float | Per completed agent turn (pipeline mode) | + +The two routing tags (`patter.call_id`, `patter.side`) propagate via asyncio-safe ContextVars set at the top of the per-call WebSocket bridge, so all spans emitted under a call inherit them automatically — no manual wiring per span. + +### Attach a custom exporter (`Patter._attach_span_exporter`) + +Embedding tools that observe Patter from the outside can wire their own exporter without touching `PATTER_OTEL_ENABLED` or `init_tracing`: + +```python +from opentelemetry.sdk.trace.export import SpanExporter +from getpatter import Patter + +exporter: SpanExporter = my_exporter() # any OTel SpanExporter +phone = Patter(...) +phone._attach_span_exporter(exporter, side="driver") # or side="uut" (default) +``` + +`side` is stamped on every cost/latency span this Patter instance emits during its call lifecycle. It exists to disambiguate two-`Patter`-instances-in-one-process layouts (e.g. driver vs unit-under-test in agent-to-agent acceptance tests). Default is `"uut"` if you only have one instance. + +The leading underscore signals this is not part of the customer-facing API surface — it is a stable, public-but-underscore hook for tooling. + ## Shutdown Call `shutdown_tracing()` during graceful shutdown to flush any pending spans: diff --git a/docs/typescript-sdk/events.mdx b/docs/typescript-sdk/events.mdx index 45e0640a..c95b935a 100644 --- a/docs/typescript-sdk/events.mdx +++ b/docs/typescript-sdk/events.mdx @@ -367,7 +367,7 @@ await phone.serve({ The callbacks above describe the *transcript-level* lifecycle of a call. For **turn-taking instrumentation** — barge-in, end-of-utterance, time-to-first-token, TTS warmup vs. wire-time — Patter exposes seven additional async callbacks plus a read-only `conversationState` snapshot directly on the `Patter` instance. -These events mirror the public APIs of [LiveKit Agents](https://docs.livekit.io/agents/) (`user_state_changed`, `agent_state_changed`, `user_turn_completed`, `user_interruption_detected`), [Pipecat](https://docs.pipecat.ai/) (`VADUserStartedSpeakingFrame`, `BotStartedSpeakingFrame`, `LLMFullResponseStartFrame`, `OutputAudioRawFrame`, `InterruptionFrame`) and [OpenAI Realtime](https://platform.openai.com/docs/guides/realtime) (`input_audio_buffer.speech_started/_stopped/_committed`) so downstream metrics map onto the canonical voice-agent metric set without translation. +These events expose the canonical voice-agent metric set (user/agent state transitions, turn boundaries, TTFT, audio first-byte) and align with [OpenAI Realtime](https://platform.openai.com/docs/guides/realtime) (`input_audio_buffer.speech_started/_stopped/_committed`) so downstream metrics work without translation. Every callback defaults to `null`. Existing code that does not register any speech-edge callback sees exactly the previous behaviour and zero overhead. The state machine is updated regardless of whether callbacks are registered, so `conversationState` is always usable. diff --git a/libraries/python/getpatter/__init__.py b/libraries/python/getpatter/__init__.py index 6fc66b4e..058319d9 100644 --- a/libraries/python/getpatter/__init__.py +++ b/libraries/python/getpatter/__init__.py @@ -19,7 +19,7 @@ See ``pyproject.toml`` and the top-level README for the full matrix. """ -__version__ = "0.6.0" +__version__ = "0.6.1" from getpatter._speech_events import ( AgentState, @@ -46,6 +46,10 @@ TTSConfig, TurnMetrics, ) +from getpatter.services.barge_in_strategies import ( + BargeInStrategy, + MinWordsStrategy, +) from getpatter.exceptions import ( ErrorCode, PatterError, @@ -84,9 +88,14 @@ from getpatter.stt.speechmatics import STT as SpeechmaticsSTT from getpatter.stt.assemblyai import STT as AssemblyAISTT -# TTS flat aliases. +# TTS flat aliases. As of 0.6.1, ``ElevenLabsTTS`` (the canonical facade) +# defaults to the WebSocket streaming transport. ``ElevenLabsWebSocketTTS`` +# is kept as a backward-compatible alias of the same class. For callers that +# want to opt **out** of the WS default and use the legacy HTTP REST +# transport explicitly, import ``ElevenLabsRestTTS``. from getpatter.tts.elevenlabs import TTS as ElevenLabsTTS from getpatter.tts.elevenlabs_ws import TTS as ElevenLabsWebSocketTTS +from getpatter.providers.elevenlabs_tts import ElevenLabsTTS as ElevenLabsRestTTS from getpatter.tts.openai import TTS as OpenAITTS from getpatter.tts.cartesia import TTS as CartesiaTTS from getpatter.tts.rime import TTS as RimeTTS @@ -379,6 +388,8 @@ def mix_pcm(agent: bytes, bg: bytes, ratio: float) -> bytes: "STTConfig", "TTSConfig", "TurnMetrics", + "BargeInStrategy", + "MinWordsStrategy", "ErrorCode", "PatterError", "PatterConnectionError", @@ -406,6 +417,7 @@ def mix_pcm(agent: bytes, bg: bytes, ratio: float) -> bytes: "AssemblyAISTT", "ElevenLabsTTS", "ElevenLabsWebSocketTTS", + "ElevenLabsRestTTS", "OpenAITTS", "CartesiaTTS", "RimeTTS", diff --git a/libraries/python/getpatter/_speech_events.py b/libraries/python/getpatter/_speech_events.py index 2bb4bec0..2f17c9b3 100644 --- a/libraries/python/getpatter/_speech_events.py +++ b/libraries/python/getpatter/_speech_events.py @@ -2,9 +2,10 @@ Defines :class:`SpeechEvents`, the per-call dispatcher that fires user-facing async callbacks and (when available) records OpenTelemetry span events on the -current call span. The 7 events mirror the public APIs of LiveKit Agents, -Pipecat and OpenAI Realtime so downstream metrics map onto the canonical -Hamming AI / Coval / Cekura voice-agent metric set without translation. +current call span. The 7 events expose the canonical voice-agent metric set +(user/agent state transitions, turn boundaries, TTFT, first-audio) so +downstream metrics work without translation, and align with OpenAI Realtime +where applicable. This module is private (leading underscore). The public surface is the 7 ``on_*`` attributes plus :meth:`conversation_state` exposed on the @@ -12,19 +13,16 @@ is re-exported at the package root for advanced users (custom adapters, test harnesses). -Industry alignment table:: +Event mapping (with the OpenAI Realtime equivalent where one exists):: - User VAD start : LiveKit ``user_state_changed -> speaking`` / - Pipecat ``VADUserStartedSpeakingFrame`` / - OpenAI Realtime ``input_audio_buffer.speech_started`` - User VAD end : ``..._stopped`` (raw VAD edge — *not* end-of-utterance) - User EOU : LiveKit ``user_turn_completed`` / Pipecat - ``UserStoppedSpeakingFrame`` / OpenAI Realtime - ``input_audio_buffer.committed`` - Agent first wire: Pipecat ``BotStartedSpeakingFrame`` - Agent done : Pipecat ``BotStoppedSpeakingFrame`` - LLM first token : Pipecat ``LLMFullResponseStartFrame`` (per-turn TTFT) - TTS first audio : Pipecat ``OutputAudioRawFrame`` (first per turn) + User VAD start : OpenAI Realtime ``input_audio_buffer.speech_started`` + User VAD end : OpenAI Realtime ``input_audio_buffer.speech_stopped`` + (raw VAD edge — *not* end-of-utterance) + User EOU : OpenAI Realtime ``input_audio_buffer.committed`` + Agent first wire: first audio chunk of the agent turn that crosses the wire + Agent done : last audio chunk of the agent turn (or barge-in) + LLM first token : per-turn TTFT marker + TTS first audio : first TTS audio chunk per turn Both VAD edge and end-of-utterance are surfaced separately because they are two different signals (`silence_gap_ms_max` wants the EOU; `cross_talk_pct` @@ -50,10 +48,10 @@ class UserState(StrEnum): """Per-side user speech state — mirror of the TypeScript ``UserState`` string-literal union. - Values match LiveKit Agents' ``user_state_changed`` vocabulary so - downstream observability dashboards (Hamming AI / Coval / Cekura) can - map Patter events onto the canonical voice-agent metric set without - translation. + Values follow the standard user-state vocabulary (``listening`` / + ``speaking`` / ``thinking`` / ``away``) so downstream observability + dashboards can map Patter events onto the canonical voice-agent metric + set without translation. """ LISTENING = "listening" @@ -66,7 +64,8 @@ class AgentState(StrEnum): """Per-side agent speech state — mirror of the TypeScript ``AgentState`` string-literal union. - Values match LiveKit Agents' ``agent_state_changed`` vocabulary; see + Values follow the standard agent-state vocabulary (``initializing`` / + ``idle`` / ``listening`` / ``thinking`` / ``speaking``); see :class:`UserState` for the rationale. """ @@ -106,7 +105,7 @@ class ConversationStateSnapshot: agent: AgentState -# State-machine values mirror LiveKit's user/agent state vocabulary. +# State-machine values follow the standard user/agent state vocabulary. # Kept as plain tuples for backwards compatibility with callers that # imported the constants directly. New code should prefer :class:`UserState` # / :class:`AgentState`. @@ -176,9 +175,9 @@ def __init__(self) -> None: def conversation_state(self) -> dict[str, str]: """Snapshot of the current per-side state. - Returns ``{"user": , "agent": }``. Mirrors - LiveKit's ``user_state_changed`` / ``agent_state_changed`` payloads - and is safe to call at any time (read-only, no I/O). + Returns ``{"user": , "agent": }`` — the + user_state / agent_state payload shape, safe to call at any time + (read-only, no I/O). """ return {"user": self._user_state, "agent": self._agent_state} diff --git a/libraries/python/getpatter/client.py b/libraries/python/getpatter/client.py index 8b042b3d..32b942f1 100644 --- a/libraries/python/getpatter/client.py +++ b/libraries/python/getpatter/client.py @@ -19,6 +19,7 @@ import asyncio import logging +import time from collections.abc import Awaitable, Callable from typing import TYPE_CHECKING, Any @@ -35,6 +36,27 @@ from getpatter._speech_events import SpeechEventCallback +# Maximum concurrent entries in the prewarm-first-message cache. Bounds +# memory consumption when an outbound flood (or attacker-controlled +# ``Patter.call`` invocations) would otherwise pile up tens of MB of +# orphan TTS bytes that never evict because the carrier never fires +# ``start``. When the cap is reached, new prewarm spawns are refused +# (logged at WARN, call still proceeds with live TTS). See FIX #96 in +# the parity audit. Mirrors ``PREWARM_CACHE_MAX`` in TS client. +_PREWARM_CACHE_MAX = 200 +# Extra grace window beyond ``ring_timeout`` after which a prewarmed +# entry that was never consumed is forcibly evicted. The TTS bill was +# paid; without TTL eviction a carrier that never fires ``start`` (e.g. +# on a never-completed dial that bypassed the status callback) would +# leak the bytes for the lifetime of the Patter instance. +_PREWARM_TTL_GRACE_S = 5.0 + +# Safety TTL after which a parked provider WebSocket whose carrier never +# fired ``start`` is force-closed. 30 s is a comfortable superset of +# typical ring + AMD windows (Twilio ~25 s, Telnyx ~25 s). +_PARKED_CONN_TTL_S = 30.0 + + _CLOUD_NOT_IMPLEMENTED_MSG = ( "Patter Cloud is not yet available in this SDK release. Use local mode " "with a `carrier=` and `phone_number=`. Cloud mode will return in a " @@ -67,6 +89,58 @@ def _resolve_persist_root(persist: bool | str | None) -> str | None: return str(result) if result is not None else None +def _close_parked_slot(slot: dict[str, Any]) -> None: + """Close every parked socket inside a parked-connections slot. + + Each slot may hold provider-specific handles: + + - ``stt`` → ``(aiohttp.ClientSession, aiohttp.ClientWebSocketResponse)`` + (Cartesia STT pattern). + - ``tts`` → :class:`ElevenLabsParkedWS` (or any object exposing ``.ws``). + - ``openai_realtime`` → :class:`websockets.WebSocketClientProtocol`. + + Closes are scheduled fire-and-forget on the running loop because + this helper may be invoked synchronously from waste-record or + disconnect paths that do not own an awaitable scope. + """ + for handle in slot.values(): + try: + asyncio.create_task(_safe_close_handle(handle)) + except RuntimeError: + # No running loop — best-effort sync close where supported. + pass + + +async def _safe_close_handle(handle: Any) -> None: + """Best-effort async close of a parked handle. + + Handles the three flavours used by the SDK: + - tuple ``(session, ws)`` from Cartesia STT. + - :class:`ElevenLabsParkedWS` (or any object with ``.ws``). + - bare WebSocket / WebSocketClientProtocol. + """ + try: + if isinstance(handle, tuple) and len(handle) == 2: + session, ws = handle + try: + await ws.close() + except Exception: + pass + try: + await session.close() + except Exception: + pass + return + ws = getattr(handle, "ws", None) + if ws is not None: + await ws.close() + return + # Bare websocket + await handle.close() + except Exception: + pass + + class Patter: """Main Patter SDK client (local mode only). @@ -164,6 +238,8 @@ def __init__( ) self._server = None self._tunnel_handle = None + # Observability — set by _attach_span_exporter, default safe. + self._patter_side: str = "uut" # tunnel_ready future — resolved once ``serve()`` knows the public # webhook hostname (either statically configured or freshly minted by # the tunnel). Initialised lazily below to avoid pulling asyncio @@ -194,18 +270,65 @@ def __init__( # ``conversation_state`` snapshot. Defaults are no-ops — existing # users who never set a callback see exactly the previous behaviour. # See ``getpatter._speech_events`` for the full event taxonomy and - # the industry-alignment table (LiveKit / Pipecat / OpenAI Realtime). + # the OpenAI Realtime alignment table. # Imported inline to keep client.py's top-level import graph minimal. from getpatter._speech_events import SpeechEvents as _SpeechEvents self.speech_events = _SpeechEvents() + # Pre-rendered first-message TTS audio per outbound call_id. + # Populated by :meth:`call` when ``agent.prewarm_first_message`` is + # True; consumed by the StreamHandler firstMessage emit path so + # the greeting streams instantly on ``start`` instead of paying the + # 200-700 ms TTS first-byte latency. See ``Agent.prewarm_first_message``. + # Stores raw bytes in the TTS provider's native sample rate; the + # carrier-side AudioSender resamples on emit. + self._prewarm_audio: dict[str, bytes] = {} + # Call IDs whose prewarm cache slot has already been consumed — + # either by ``pop_prewarm_audio`` (cache hit OR miss on the + # firstMessage emit path) or by ``_record_prewarm_waste`` (call + # ended before pickup). The prewarm task checks this set BEFORE + # writing bytes so a slow synth that finishes after the consumer + # already polled doesn't orphan bytes in ``_prewarm_audio``. See + # FIX #92 in the parity audit. + self._prewarm_consumed: set[str] = set() + # Background tasks tracked so :meth:`disconnect` can cancel any + # still-running prewarm-first-message synth before tearing down. + self._prewarm_tasks: set[asyncio.Task] = set() + # TTL eviction tasks tracked so :meth:`disconnect` can cancel any + # pending eviction timer before tearing down. Keyed by call_id so + # a follow-up consume / waste-record path can also cancel the + # timer when the slot drains naturally. + self._prewarm_ttl_tasks: dict[str, asyncio.Task] = {} + # Pre-opened, fully-handshaked provider WebSockets keyed by + # carrier-issued call_id. Populated by + # :meth:`_park_provider_connections` during the carrier + # ringing window; consumed by the per-call StreamHandler at + # ``start`` via ``adopt_websocket(...)`` so STT / TTS / + # Realtime audio can flow on the first turn without paying + # the 150-900 ms TLS + WS-upgrade + protocol-handshake + # round-trip again. + # + # Each value is a ``dict`` with optional keys ``stt``, ``tts``, + # ``openai_realtime`` — provider-specific handles that the + # StreamHandler hands to the matching adapter's + # ``adopt_websocket`` method. + # + # Distinct from ``_prewarm_audio`` (pre-rendered TTS bytes for + # the first message); the two features are complementary and + # orthogonal — both can be active for the same call. + self._prewarmed_connections: dict[str, dict[str, Any]] = {} + # TTL eviction tasks for parked connections, keyed by call_id. + self._prewarmed_conn_tasks: dict[str, asyncio.Task] = {} + # ------------------------------------------------------------------ # Speech-edge event callback proxies # ------------------------------------------------------------------ - # The seven ``on_*`` attributes below mirror the public APIs of LiveKit - # Agents, Pipecat and OpenAI Realtime. They proxy to ``self.speech_events`` - # so the dispatcher remains the single source of truth (state + OTel). + # The seven ``on_*`` attributes below follow the canonical voice-agent + # metric set (user/agent state transitions, turn boundaries, TTFT, audio + # first-byte) and align with OpenAI Realtime where applicable. They + # proxy to ``self.speech_events`` so the dispatcher remains the single + # source of truth (state + OTel). @property def on_user_speech_started(self) -> SpeechEventCallback | None: @@ -267,9 +390,8 @@ def on_audio_out(self, cb: SpeechEventCallback | None) -> None: def conversation_state(self) -> dict[str, str]: """Snapshot of the current per-side state of the call. - Returns ``{"user": , "agent": }``. Mirrors LiveKit's - ``user_state_changed`` / ``agent_state_changed`` payloads. Read-only - and safe to call at any time. + Returns ``{"user": , "agent": }`` — the user_state / + agent_state snapshot. Read-only and safe to call at any time. """ return self.speech_events.conversation_state @@ -475,6 +597,15 @@ async def call( wants_amd = bool(machine_detection) or bool(voicemail_message) if self._server is not None: self._server.on_machine_detection = on_machine_detection # type: ignore[attr-defined] + + # Pre-warm provider connections in parallel with the carrier-side + # ``initiate_call`` so DNS / TLS / HTTP/2 handshakes complete during + # the ringing window (3-15 s typically). Best-effort: warmup + # failures are logged at DEBUG and never abort the call. Off when + # the user explicitly sets ``Agent(prewarm=False)``. + if getattr(agent, "prewarm", True): + self._spawn_provider_warmup(agent) + config = self._local_config if config.telephony_provider == "twilio": from getpatter.providers.twilio_adapter import TwilioAdapter # type: ignore[import] @@ -550,6 +681,13 @@ async def call( ) except Exception as exc: logger.debug("record_call_initiated: %s", exc) + self._spawn_prewarm_first_message(agent, call_id, ring_timeout=ring_timeout) + # Park provider WebSockets in parallel so the per-call + # StreamHandler can adopt them at ``start`` instead of + # paying the cold-handshake on first turn. Off when the + # user explicitly sets ``agent.prewarm=False``. + if getattr(agent, "prewarm", True) is not False: + self._park_provider_connections(agent, call_id) elif config.telephony_provider == "telnyx": from getpatter.providers.telnyx_adapter import TelnyxAdapter # type: ignore[import] @@ -581,6 +719,358 @@ async def call( ) except Exception as exc: logger.debug("record_call_initiated: %s", exc) + self._spawn_prewarm_first_message(agent, call_id, ring_timeout=ring_timeout) + # Park provider WebSockets in parallel so the per-call + # StreamHandler can adopt them at ``start`` instead of + # paying the cold-handshake on first turn. Off when the + # user explicitly sets ``agent.prewarm=False``. + if getattr(agent, "prewarm", True) is not False: + self._park_provider_connections(agent, call_id) + + # === Pre-warm helpers === + + def _spawn_provider_warmup(self, agent: Agent) -> None: + """Spawn a fire-and-forget task that warms up STT / TTS / LLM in + parallel with the carrier-side ``initiate_call``. + + Best-effort: each provider's ``warmup()`` is wrapped in + ``asyncio.gather(..., return_exceptions=True)`` so a slow or + failing endpoint cannot block the others. The default + ``warmup()`` on the abstract base classes is a no-op, so providers + that don't override it contribute nothing to call latency. + """ + targets = [] + for provider in ( + getattr(agent, "stt", None), + getattr(agent, "tts", None), + getattr(agent, "llm", None), + ): + if provider is None: + continue + warmup = getattr(provider, "warmup", None) + if warmup is None or not callable(warmup): + continue + targets.append(provider) + + if not targets: + return + + async def _run_all() -> None: + results = await asyncio.gather( + *(p.warmup() for p in targets), + return_exceptions=True, + ) + for provider, result in zip(targets, results): + if isinstance(result, BaseException): + logger.debug( + "Provider warmup failed (%s): %s", + type(provider).__name__, + result, + ) + + task = asyncio.create_task(_run_all()) + # Track but don't await — warmup runs in parallel with the carrier + # call and never blocks the user. + self._prewarm_tasks.add(task) + task.add_done_callback(self._prewarm_tasks.discard) + + def pop_prewarmed_connections(self, call_id: str) -> dict[str, Any] | None: + """Pop and return the parked provider WS handles for ``call_id``, + or ``None`` when no parked connections exist. + + Wired into the per-call ``StreamHandler`` so it can adopt the + parked sockets at the carrier ``start`` event instead of paying + the cold handshake on first turn. + """ + slot = self._prewarmed_connections.pop(call_id, None) + ttl_task = self._prewarmed_conn_tasks.pop(call_id, None) + if ttl_task is not None: + ttl_task.cancel() + return slot + + def close_prewarmed_connections(self, call_id: str) -> None: + """Close any parked provider WSs for ``call_id`` cleanly. + + Wired into call-termination paths (no-answer, busy, failed, + canceled, AMD voicemail) so the sockets drop instead of being + left to the upstream timeout. + """ + slot = self._prewarmed_connections.pop(call_id, None) + ttl_task = self._prewarmed_conn_tasks.pop(call_id, None) + if ttl_task is not None: + ttl_task.cancel() + if slot is not None: + _close_parked_slot(slot) + + def _park_provider_connections(self, agent: Agent, call_id: str) -> None: + """Open and park provider WebSockets in parallel with the + carrier-side ``initiate_call``. Unlike :meth:`_spawn_provider_warmup` + (which closes the WS after a brief idle), the sockets opened here + stay OPEN and are handed off to the per-call ``StreamHandler`` on + ``start``. + + Structural fix for first-turn cold-start: opening + closing a WS + does NOT warm TLS for the next open — every fresh + ``websockets.connect`` re-pays the full TCP + TLS + HTTP-101 + round-trip. Keeping the WS open and adopting it directly skips + the handshake entirely (saves ~150-900 ms depending on provider). + + Best-effort: each provider's parking task is wrapped in + ``asyncio.gather(..., return_exceptions=True)`` so a slow or + failing endpoint cannot block the others. Providers without + ``open_parked_connection`` contribute nothing. + """ + stt = getattr(agent, "stt", None) + tts = getattr(agent, "tts", None) + stt_open = getattr(stt, "open_parked_connection", None) if stt else None + tts_open = getattr(tts, "open_parked_connection", None) if tts else None + if stt_open is None and tts_open is None: + return + + slot: dict[str, Any] = {} + self._prewarmed_connections[call_id] = slot + + started_at = time.monotonic() + + async def _park_stt() -> None: + if stt_open is None: + return + try: + handle = await stt_open() + # Slot may have been drained while we were opening. + if self._prewarmed_connections.get(call_id) is not slot: + await _safe_close_handle(handle) + return + slot["stt"] = handle + logger.info( + "[PREWARM] callId=%s provider=stt ms=%d", + call_id, + int((time.monotonic() - started_at) * 1000), + ) + except Exception as exc: # noqa: BLE001 - best-effort + logger.debug("Park STT failed for %s: %s", call_id, exc) + + async def _park_tts() -> None: + if tts_open is None: + return + try: + handle = await tts_open() + if self._prewarmed_connections.get(call_id) is not slot: + await _safe_close_handle(handle) + return + slot["tts"] = handle + logger.info( + "[PREWARM] callId=%s provider=tts ms=%d", + call_id, + int((time.monotonic() - started_at) * 1000), + ) + except Exception as exc: # noqa: BLE001 - best-effort + logger.debug("Park TTS failed for %s: %s", call_id, exc) + + async def _run_all() -> None: + await asyncio.gather(_park_stt(), _park_tts(), return_exceptions=True) + + task = asyncio.create_task(_run_all()) + self._prewarm_tasks.add(task) + + def _on_park_done(_t: asyncio.Task) -> None: + self._prewarm_tasks.discard(_t) + # Schedule TTL cleanup so a never-adopted slot is force-closed. + if call_id not in self._prewarmed_connections: + return + try: + ttl_task = asyncio.create_task( + self._evict_parked_after(call_id, _PARKED_CONN_TTL_S) + ) + except RuntimeError: + # No running loop — drop synchronously. + orphan = self._prewarmed_connections.pop(call_id, None) + if orphan is not None: + _close_parked_slot(orphan) + return + self._prewarmed_conn_tasks[call_id] = ttl_task + ttl_task.add_done_callback( + lambda _t, cid=call_id: self._prewarmed_conn_tasks.pop(cid, None) + ) + + task.add_done_callback(_on_park_done) + + async def _evict_parked_after(self, call_id: str, ttl_s: float) -> None: + """Sleep ``ttl_s`` then force-close any parked sockets still + present for ``call_id``. No-op if the slot was already + consumed / closed. + """ + try: + await asyncio.sleep(ttl_s) + except asyncio.CancelledError: + return + slot = self._prewarmed_connections.pop(call_id, None) + if slot is not None: + _close_parked_slot(slot) + logger.warning( + "[PREWARM] parked connections evicted by TTL for %s — " + "call never reached start (~%.0fs).", + call_id, + ttl_s, + ) + + def _spawn_prewarm_first_message( + self, agent: Agent, call_id: str, *, ring_timeout: int | None + ) -> None: + """Pre-render ``agent.first_message`` to TTS bytes during the + ringing window and stash them in ``_prewarm_audio[call_id]``. + + Skipped silently when ``agent.prewarm_first_message`` is False or + when ``agent.tts`` / ``agent.first_message`` is missing. The synth + is bounded by ``ring_timeout`` (default 25 s) so a never-answered + call doesn't tie up the TTS connection. On timeout / error the + cache is left empty and the StreamHandler falls back to live TTS. + + **Pipeline mode only.** Realtime / ConvAI provider modes never + consume the prewarm cache (the StreamHandler for those modes runs + its first-message emit through the provider's own audio path). + Spawning the prewarm in those modes pays the TTS bill for nothing + — refused with a WARN. + + **Capped at ``_PREWARM_CACHE_MAX`` concurrent entries.** Refused + with a WARN when the cap is reached (the call still proceeds — + StreamHandler falls back to live TTS). + """ + if not getattr(agent, "prewarm_first_message", False): + return + # FIX #94 — Realtime / ConvAI never consume the cache. Refuse early + # so the user notices the silent TTS waste instead of paying for a + # synth no caller will ever hear. + provider_mode = getattr(agent, "provider", "openai_realtime") + if provider_mode != "pipeline": + logger.warning( + "agent.prewarm_first_message=True is only supported in pipeline " + "mode (provider=%s); skipping pre-synth to avoid wasted TTS spend.", + provider_mode, + ) + return + first_message = getattr(agent, "first_message", "") or "" + tts = getattr(agent, "tts", None) + if not first_message or tts is None: + return + synthesize = getattr(tts, "synthesize", None) + if synthesize is None or not callable(synthesize): + return + + # FIX #96 — refuse to spawn when the cache (live entries + + # in-flight synth tasks) would exceed the cap. Counting both + # active entries AND pending tasks keeps the bound honest under + # outbound-flood conditions where carrier ``start`` events lag. + in_flight = len(self._prewarm_audio) + len(self._prewarm_tasks) + if in_flight >= _PREWARM_CACHE_MAX: + logger.warning( + "Prewarm cache full (%d/%d in-flight) — skipping pre-synth for " + "call %s; falling back to live TTS at pickup.", + in_flight, + _PREWARM_CACHE_MAX, + call_id, + ) + return + + timeout_s = float(ring_timeout) if ring_timeout is not None else 25.0 + + async def _run() -> None: + try: + buf = bytearray() + + async def _accumulate() -> None: + async for chunk in synthesize(first_message): + if isinstance(chunk, (bytes, bytearray)): + buf.extend(chunk) + + await asyncio.wait_for(_accumulate(), timeout=timeout_s) + if buf: + # FIX #92 — race guard. If the consumer already polled + # (cache hit or miss) before the synth finished, the + # StreamHandler has already fallen back to live TTS; + # writing bytes here would orphan them in + # ``_prewarm_audio`` until ``end_call`` ever runs. + if call_id in self._prewarm_consumed: + logger.warning( + "Prewarm orphaned for call %s — synth completed " + "(~%d bytes) AFTER consumer polled; bytes dropped, " + "TTS bill already paid.", + call_id, + len(buf), + ) + return + self._prewarm_audio[call_id] = bytes(buf) + logger.debug( + "Prewarm first-message ready for call %s (%d bytes)", + call_id, + len(buf), + ) + except asyncio.TimeoutError: + logger.debug( + "Prewarm first-message timed out for call %s after %.1fs", + call_id, + timeout_s, + ) + except Exception as exc: # noqa: BLE001 - best-effort + logger.debug( + "Prewarm first-message failed for call %s: %s", call_id, exc + ) + + task = asyncio.create_task(_run()) + self._prewarm_tasks.add(task) + + def _on_synth_done(_t: asyncio.Task) -> None: + self._prewarm_tasks.discard(_t) + # FIX #96 — schedule TTL eviction once the synth task has + # produced (or failed to produce) cache bytes. If the carrier + # never fires ``start`` AND the status / hangup callback + # never runs (e.g. cloud-side telephony quirk), the entry + # would otherwise leak. The eviction task itself is short + # (just an ``asyncio.sleep`` + pop) and is no-op when the + # slot has already been drained by ``pop_prewarm_audio`` / + # ``_record_prewarm_waste``. + if call_id not in self._prewarm_audio: + return + ttl_s = timeout_s + _PREWARM_TTL_GRACE_S + try: + evict_task = asyncio.create_task( + self._evict_prewarm_after(call_id, ttl_s) + ) + except RuntimeError: + # No running loop (process shutting down) — drop the + # entry synchronously to avoid leaking it. + self._prewarm_audio.pop(call_id, None) + return + self._prewarm_ttl_tasks[call_id] = evict_task + evict_task.add_done_callback( + lambda _t, cid=call_id: self._prewarm_ttl_tasks.pop(cid, None) + ) + + task.add_done_callback(_on_synth_done) + + async def _evict_prewarm_after(self, call_id: str, ttl_s: float) -> None: + """Sleep ``ttl_s`` then drop ``call_id`` from the cache if still present. + + The TTS bill was paid by the synth task; this WARN flags the + unconsumed entry so users notice never-answered calls that + slipped past the status / hangup callback. Cancelled by + :meth:`disconnect` and a no-op when the entry was already + consumed via ``pop_prewarm_audio`` / ``_record_prewarm_waste``. + """ + try: + await asyncio.sleep(ttl_s) + except asyncio.CancelledError: + return + bytes_ = self._prewarm_audio.pop(call_id, None) + if bytes_ is not None: + self._prewarm_consumed.add(call_id) + logger.warning( + "Prewarm bytes evicted by TTL — call %s never consumed them " + "(~%d bytes synthesised, %.1fs after ring_timeout).", + call_id, + len(bytes_), + ttl_s, + ) # === Local mode helpers === @@ -944,6 +1434,19 @@ async def serve( f"recording must be a bool, got {type(recording).__name__}." ) + # Pre-import AEC at serve startup so the first call doesn't pay + # the dynamic-import cost on the hot path. ``echo_cancellation`` + # is opt-in and rarely set on PSTN, but when it is the lazy + # ``from getpatter.audio.aec import NlmsEchoCanceller`` inside + # the StreamHandler can serialise with first-message TTS startup + # and eat first-turn latency. Eagerly importing here costs + # nothing for users who never enable AEC. + if getattr(agent, "echo_cancellation", False): + try: + import getpatter.audio.aec # noqa: F401 + except Exception as exc: # noqa: BLE001 - best-effort + logger.debug("AEC pre-import failed at serve(): %s", exc) + # Resolve webhook_url: tunnel or explicit config = self._local_config @@ -1014,6 +1517,23 @@ async def serve( # ``_emit_*_speech_*`` paths short-circuit on ``self.speech_events # is None`` and zero events ever reach the runner's tap. self._server.speech_events = self.speech_events + # Forward the prewarm-audio accessor so the per-call StreamHandler + # can consume the pre-rendered first-message audio (if any) on + # ``start``. The server stores a closure rather than a back-ref to + # avoid a circular reference (Patter → server → Patter). + self._server.pop_prewarm_audio = self.pop_prewarm_audio # type: ignore[attr-defined] + # Forward the parked-connections accessor so the per-call + # StreamHandler can adopt pre-opened STT / TTS / Realtime WSs at + # ``start`` instead of paying the cold handshake on first turn. + self._server.pop_prewarmed_connections = self.pop_prewarmed_connections # type: ignore[attr-defined] + # Forward the waste-recorder so the carrier status / hangup + # webhook handlers can evict the cache when a call terminates + # before the media stream starts (no-answer, busy, failed, + # canceled, or AMD voicemail). Without this, ``_record_prewarm_waste`` + # is only invoked from ``end_call`` — and the server-side teardown + # path leaks the bytes for the lifetime of the Patter instance. + # See FIX #91. + self._server.record_prewarm_waste = self._record_prewarm_waste # type: ignore[attr-defined] # Run uvicorn in a task so we can resolve ``phone.ready`` once it # finishes its startup phase. ``server.start()`` itself awaits @@ -1095,6 +1615,23 @@ async def test( on_call_end=on_call_end, ) + def _attach_span_exporter(self, exporter: Any, *, side: str = "uut") -> None: + """Wire an OTel span exporter into the SDK's tracer provider. + + Public-but-underscore: consumed by ``patter-agent-runner`` via + ``getattr(phone, "_attach_span_exporter")``. The leading underscore + signals it is not part of the customer-facing API surface. + + Args: + exporter: Any OTel ``SpanExporter`` (e.g. ``InMemorySpanExporter``, + ``OTLPSpanExporter``, or the runner's ``PatterSpanExporter``). + side: ``"driver"`` or ``"uut"``. Stamped on every cost/latency + span emitted during this Patter instance's call lifecycle. + """ + from getpatter.observability.attributes import attach_span_exporter + + attach_span_exporter(self, exporter, side=side) + async def disconnect(self) -> None: """Stop the embedded server and any auto-started tunnel. @@ -1102,7 +1639,48 @@ async def disconnect(self) -> None: subsequent ``serve()`` works as if the previous lifecycle never happened (clears tunnel-owned ``webhook_url`` and recreates the ``ready`` / ``tunnel_ready`` Futures). + + Also cancels any in-flight prewarm-first-message synth tasks and + TTL eviction timers, then clears the prewarm cache. Without this + a still-running TTS WS keeps the user billed long after SDK + teardown, and stale entries leak across ``serve`` / + ``disconnect`` cycles. See FIX #93. """ + # Cancel and drain any in-flight prewarm work BEFORE tearing the + # server down so the synth tasks see a clean cancellation point + # and don't end up writing bytes to a cache we're about to drop. + for t in list(self._prewarm_tasks): + t.cancel() + for t in list(self._prewarm_ttl_tasks.values()): + t.cancel() + if self._prewarm_tasks: + await asyncio.gather(*self._prewarm_tasks, return_exceptions=True) + if self._prewarm_ttl_tasks: + await asyncio.gather( + *self._prewarm_ttl_tasks.values(), return_exceptions=True + ) + self._prewarm_tasks.clear() + self._prewarm_ttl_tasks.clear() + self._prewarm_audio.clear() + self._prewarm_consumed.clear() + # Cancel parked-connection TTL tasks and force-close any + # remaining parked sockets so we don't leak across + # ``serve`` / ``disconnect`` cycles. + for t in list(self._prewarmed_conn_tasks.values()): + if not t.done(): + t.cancel() + if self._prewarmed_conn_tasks: + await asyncio.gather( + *self._prewarmed_conn_tasks.values(), return_exceptions=True + ) + self._prewarmed_conn_tasks.clear() + for slot in list(self._prewarmed_connections.values()): + for handle in slot.values(): + try: + await _safe_close_handle(handle) + except Exception: + pass + self._prewarmed_connections.clear() if self._server: await self._server.stop() self._server = None @@ -1128,6 +1706,66 @@ async def disconnect(self) -> None: else: self._tunnel_ready_pre_resolved = None + def pop_prewarm_audio(self, call_id: str) -> bytes | None: + """Pop and return the pre-synthesised first-message audio for ``call_id``. + + Returns ``None`` when ``agent.prewarm_first_message`` was not set + for the originating outbound call, or when the synth was still in + flight at the moment the carrier emitted ``start`` (treated as a + prewarm miss — the StreamHandler falls back to live TTS). + + Called by the per-call StreamHandler at the start of the + firstMessage emit. Returning bytes here lets the handler skip the + live TTS synthesis and stream the cached buffer directly. + + Marks ``call_id`` as consumed regardless of cache hit/miss so a + slow synth task that finishes after this call drops its bytes + instead of orphaning them in ``_prewarm_audio``. See FIX #92. + """ + self._prewarm_consumed.add(call_id) + # Cancel any pending TTL eviction — the slot is being drained + # naturally now. + ttl = self._prewarm_ttl_tasks.pop(call_id, None) + if ttl is not None and not ttl.done(): + ttl.cancel() + return self._prewarm_audio.pop(call_id, None) + + def _record_prewarm_waste(self, call_id: str) -> None: + """Log a WARN if a prewarmed greeting was paid for but never used. + + Called from :meth:`disconnect`, :meth:`end_call`, and from the + carrier status / hangup webhook handlers when a call terminates + before the media stream starts. The TTS bill for + ``agent.first_message`` has already been incurred by the + background synth task, so the user should know — opt-in feature + with a known cost surface. + + Idempotent: the second call for the same ``call_id`` is a no-op, + so the status callback firing first and ``end_call`` running + afterwards (or vice-versa) does not double-WARN. + """ + # Always drain any parked provider WS — they're cheap to discard + # and we don't want to leak open sockets when the call dies. + self.close_prewarmed_connections(call_id) + # Idempotency guard — once consumed (cache hit, cache miss, or a + # prior waste record) the slot is gone and there is nothing to + # warn about a second time. + if call_id in self._prewarm_consumed: + self._prewarm_audio.pop(call_id, None) + return + self._prewarm_consumed.add(call_id) + ttl = self._prewarm_ttl_tasks.pop(call_id, None) + if ttl is not None and not ttl.done(): + ttl.cancel() + bytes_ = self._prewarm_audio.pop(call_id, None) + if bytes_: + logger.warning( + "Prewarm wasted for call %s — first-message TTS already paid " + "(~%d bytes synthesised) but call ended before pickup.", + call_id, + len(bytes_), + ) + async def end_call(self, call_sid: str) -> None: """Terminate an active call on the configured carrier. @@ -1152,6 +1790,9 @@ async def end_call(self, call_sid: str) -> None: """ if not call_sid: raise ValueError("call_sid must be a non-empty string") + # If the call had a prewarmed first-message that was never consumed + # (call ended before pickup), surface the wasted spend to the user. + self._record_prewarm_waste(call_sid) cfg = self._local_config telephony = cfg.telephony_provider if telephony == "twilio": diff --git a/libraries/python/getpatter/dashboard/routes.py b/libraries/python/getpatter/dashboard/routes.py index e5d38f80..6ea74cfa 100644 --- a/libraries/python/getpatter/dashboard/routes.py +++ b/libraries/python/getpatter/dashboard/routes.py @@ -53,7 +53,14 @@ async def dashboard_calls(request: Request): @app.get("/api/dashboard/calls/{call_id}") async def dashboard_call_detail(call_id: str, _=Depends(auth)): + # Fall back to the active record so the live-transcript polling + # path (``useTranscript`` in the dashboard SPA) sees turns as + # they accumulate during the call. Without this fallback the + # route 404s while the call is in flight and the live transcript + # pane stays empty. call = store.get_call(call_id) + if call is None: + call = store.get_active(call_id) if call is None: return JSONResponse(content={"error": "Not found"}, status_code=404) return JSONResponse(content=call) @@ -66,6 +73,35 @@ async def dashboard_active(_=Depends(auth)): async def dashboard_aggregates(_=Depends(auth)): return JSONResponse(content=store.get_aggregates()) + # --- Soft delete --- + # + # ``DELETE /api/dashboard/calls/{call_id}`` removes a single call from + # the dashboard view and aggregate metrics. ``POST + # /api/dashboard/calls/delete`` accepts a batch ``{"call_ids": [...]}``. + # Both are idempotent and never touch the on-disk artefacts written by + # ``CallLogger`` — those serve as the durable backup. Active calls are + # silently skipped so a mid-call delete cannot orphan the live pane. + + @app.delete("/api/dashboard/calls/{call_id}", dependencies=[Depends(auth)]) + async def dashboard_delete_call(call_id: str): + accepted = store.delete_calls([call_id]) + return JSONResponse(content={"deleted": accepted, "count": len(accepted)}) + + @app.post("/api/dashboard/calls/delete", dependencies=[Depends(auth)]) + async def dashboard_delete_calls(request: Request): + try: + body = await request.json() + except Exception: + body = {} + raw = body.get("call_ids") if isinstance(body, dict) else None + if not isinstance(raw, list): + return JSONResponse( + content={"error": "Expected JSON body {'call_ids': [...]}"}, + status_code=400, + ) + accepted = store.delete_calls([cid for cid in raw if isinstance(cid, str)]) + return JSONResponse(content={"deleted": accepted, "count": len(accepted)}) + # --- SSE endpoint --- @app.get("/api/dashboard/events") @@ -78,7 +114,7 @@ async def event_generator(): try: event = await asyncio.wait_for(queue.get(), timeout=30.0) event_type = event.get("type", "message") - event_type = re.sub(r'[\r\n]', '', event_type) + event_type = re.sub(r"[\r\n]", "", event_type) data = json.dumps(event.get("data", {}), default=str) yield f"event: {event_type}\ndata: {data}\n\n" except asyncio.TimeoutError: @@ -89,9 +125,7 @@ async def event_generator(): finally: store.unsubscribe(queue) - return StreamingResponse( - event_generator(), media_type="text/event-stream" - ) + return StreamingResponse(event_generator(), media_type="text/event-stream") # --- Export endpoint --- diff --git a/libraries/python/getpatter/dashboard/store.py b/libraries/python/getpatter/dashboard/store.py index 25ae1b7c..634f377c 100644 --- a/libraries/python/getpatter/dashboard/store.py +++ b/libraries/python/getpatter/dashboard/store.py @@ -62,6 +62,16 @@ def __init__(self, max_calls: int = 500) -> None: self._calls: list[dict[str, Any]] = [] self._active_calls: dict[str, dict[str, Any]] = {} self._subscribers: set[asyncio.Queue] = set() + # User-driven soft delete: call_ids the operator has removed from the + # dashboard. The on-disk artefacts (metadata.json, transcript.jsonl) + # are intentionally NOT touched — they serve as the durable backup. + # All read paths (``get_calls`` / ``get_call`` / ``get_aggregates`` / + # ``get_calls_in_range`` / ``hydrate``) filter against this set so + # the call is invisible to the UI and excluded from rolling metrics. + # Populated from ``/.deleted_call_ids.json`` on hydrate so + # deletions survive a process restart. + self._deleted_call_ids: set[str] = set() + self._deleted_ids_path: str | None = None # --- SSE event bus --- @@ -115,16 +125,19 @@ def record_call_start(self, data: dict[str, Any]) -> None: # If the call was pre-registered with ``record_call_initiated`` # (e.g., outbound dial before media arrives), upgrade its status # to "in-progress" instead of overwriting the from/to metadata. - # Only overwrite ``direction`` when the caller explicitly passed - # one in ``data`` — otherwise we'd clobber the ``outbound`` set - # by ``record_call_initiated`` with the default ``inbound``. + # Only overwrite ``caller`` / ``callee`` / ``direction`` when the + # caller explicitly passed a non-empty value in ``data`` — + # otherwise we'd clobber the values set by + # ``record_call_initiated`` with the empty strings the bridge + # sees on the outbound WS path (``/ws/stream/outbound`` carries + # no caller/callee query parameters). if existing is not None: - update_payload = { - "call_id": event_data["call_id"], - "caller": event_data["caller"], - "callee": event_data["callee"], - } - if "direction" in data: + update_payload: dict[str, Any] = {"call_id": event_data["call_id"]} + if event_data["caller"]: + update_payload["caller"] = event_data["caller"] + if event_data["callee"]: + update_payload["callee"] = event_data["callee"] + if "direction" in data and data["direction"]: update_payload["direction"] = data["direction"] existing.update(update_payload) existing["status"] = "in-progress" @@ -244,27 +257,55 @@ def record_call_end( return with self._lock: active = self._active_calls.pop(call_id, None) + # The Twilio ``statusCallback`` for ``CallStatus=completed`` + # arrives shortly before the WS ``stop`` frame and runs + # ``update_call_status``, which already moved the row from + # ``_active_calls`` into ``_calls``. By the time + # ``record_call_end`` runs the active record is gone and the + # completed entry already exists. Without this lookup we'd + # append a second row with ``started_at=0`` (no active to copy + # from) and empty caller/callee — which is then ranked first + # by ``get_calls`` (newest wins) and the older, well-formed + # row gets shadowed. End result: the call disappears from the + # dashboard's 24 h window. See dashboard BUG C. + existing_idx = -1 + existing: dict[str, Any] | None = None + if active is None: + for idx in range(len(self._calls) - 1, -1, -1): + if self._calls[idx].get("call_id") == call_id: + existing_idx = idx + existing = self._calls[idx] + break + entry: dict[str, Any] = { "call_id": call_id, "ended_at": time.time(), "transcript": data.get("transcript", []), } - if active: - entry["caller"] = active.get("caller", "") - entry["callee"] = active.get("callee", "") - entry["direction"] = active.get("direction", "inbound") - entry["started_at"] = active.get("started_at", 0) + source = active or existing + if source: + entry["caller"] = source.get("caller", "") + entry["callee"] = source.get("callee", "") + entry["direction"] = source.get("direction", "inbound") + entry["started_at"] = source.get("started_at", 0) # Preserve any explicit status (no-answer, busy, ...) set by # a statusCallback during the call. Fall back to "completed". + prior_status = source.get("status") entry["status"] = ( - active.get("status", "completed") - if active.get("status") != "in-progress" + prior_status + if prior_status and prior_status != "in-progress" else "completed" ) else: entry.setdefault("status", "completed") if metrics is not None: entry["metrics"] = asdict(metrics) + elif existing is not None and existing.get("metrics"): + # An earlier ``update_call_status`` may have written a + # placeholder metrics dict — keep it rather than dropping + # it on the floor when ``record_call_end`` is invoked + # without an explicit metrics payload. + entry["metrics"] = existing["metrics"] else: # No metrics payload (e.g. webhook-rejected inbound, or # outbound call that never hit media): synthesise a minimal @@ -288,9 +329,13 @@ def record_call_end( "latency_p99": {"total_ms": 0.0}, "provider_mode": "", } - self._calls.append(entry) - if len(self._calls) > self._max_calls: - self._calls = self._calls[-self._max_calls :] + if existing_idx >= 0: + # Update in place so the buffer doesn't grow a duplicate row. + self._calls[existing_idx] = entry + else: + self._calls.append(entry) + if len(self._calls) > self._max_calls: + self._calls = self._calls[-self._max_calls :] event_metrics = entry.get("metrics") # Publish outside lock to avoid deadlock with subscribe/unsubscribe self._publish( @@ -302,19 +347,117 @@ def record_call_end( ) def get_calls(self, limit: int = 50, offset: int = 0) -> list[dict[str, Any]]: - """Return the most recent completed calls, newest first.""" + """Return the most recent completed calls, newest first. + + Soft-deleted call_ids (see :py:meth:`delete_calls`) are filtered out + so the dashboard never re-shows a row the user removed. The on-disk + artefacts are intentionally preserved as a backup. + """ with self._lock: - ordered = list(reversed(self._calls)) + ordered = [ + c + for c in reversed(self._calls) + if c.get("call_id") not in self._deleted_call_ids + ] return ordered[offset : offset + limit] def get_call(self, call_id: str) -> dict[str, Any] | None: - """Return the completed-call record for ``call_id`` if present.""" + """Return the completed-call record for ``call_id`` if present. + + Soft-deleted call_ids resolve to ``None`` so the SPA's detail pane + cannot render a row the user removed (it falls back to the live + record only when ``get_call`` returns ``None``, but a deleted call + is never live by construction). + """ with self._lock: + if call_id in self._deleted_call_ids: + return None for call in reversed(self._calls): if call["call_id"] == call_id: return call return None + # --- Soft delete --- + + def delete_calls(self, call_ids: list[str] | set[str]) -> list[str]: + """Soft-delete one or more calls from the dashboard view. + + Adds each ``call_id`` to an in-memory set. Subsequent reads via + :py:meth:`get_calls` / :py:meth:`get_call` / + :py:meth:`get_aggregates` / :py:meth:`get_calls_in_range` exclude + the deleted ids, so rolling metrics (avg latency, total spend) are + recomputed without them. The on-disk ``metadata.json`` / + ``transcript.jsonl`` files written by ``CallLogger`` are NOT + touched — they serve as a durable backup the operator can audit + outside the dashboard. + + **Active calls are never deletable.** A call_id that is currently + in ``_active_calls`` is silently skipped so a mid-call delete + from the UI cannot orphan the live transcript pane. + + The deleted set is persisted to ``/.deleted_call_ids.json`` + when :py:meth:`hydrate` has been called with a log root — so the + deletion survives process restart. Persistence is best-effort; an + I/O error is logged at debug level and swallowed. + + Args: + call_ids: Iterable of call_id strings to mark deleted. Empty or + already-deleted ids are de-duplicated. Active call_ids are + filtered out. + + Returns: + The list of call_ids actually accepted as deleted (post-filter). + """ + ids = {cid for cid in (call_ids or []) if isinstance(cid, str) and cid} + if not ids: + return [] + with self._lock: + # Filter out active calls — never delete a live row. + ids -= set(self._active_calls.keys()) + # De-dup against already-deleted. + new_ids = ids - self._deleted_call_ids + if not new_ids: + return [] + self._deleted_call_ids |= new_ids + snapshot = sorted(self._deleted_call_ids) + # Persist outside the lock; SSE publish outside the lock. + self._persist_deleted_ids(snapshot) + accepted = sorted(new_ids) + self._publish("calls_deleted", {"call_ids": accepted}) + return accepted + + def is_deleted(self, call_id: str) -> bool: + """Return ``True`` when ``call_id`` was soft-deleted from the dashboard.""" + with self._lock: + return call_id in self._deleted_call_ids + + def get_deleted_call_ids(self) -> list[str]: + """Return a snapshot of the soft-deleted call_ids (sorted).""" + with self._lock: + return sorted(self._deleted_call_ids) + + def _persist_deleted_ids(self, snapshot: list[str]) -> None: + """Atomically write the deleted-ids set to disk. Best-effort.""" + if self._deleted_ids_path is None: + return + import json + import logging + import os + from pathlib import Path + + path = Path(self._deleted_ids_path) + try: + path.parent.mkdir(parents=True, exist_ok=True) + tmp = path.with_suffix(".json.tmp") + payload = {"version": 1, "deleted_call_ids": snapshot} + with open(tmp, "w", encoding="utf-8") as fh: + json.dump(payload, fh, indent=2) + os.replace(tmp, path) + except OSError as exc: + logging.getLogger("getpatter.dashboard.store").debug( + "MetricsStore._persist_deleted_ids: %s", exc + ) + def get_active_calls(self) -> list[dict[str, Any]]: """Return the currently in-flight calls.""" with self._lock: @@ -331,9 +474,17 @@ def get_active(self, call_id: str) -> dict[str, Any] | None: return self._active_calls.get(call_id) def get_aggregates(self) -> dict[str, Any]: - """Compute aggregate stats (call count, cost, avg duration, latency) across history.""" + """Compute aggregate stats (call count, cost, avg duration, latency) across history. + + Soft-deleted calls are excluded so rolling metrics (avg latency, + total spend) are recomputed without them — matching what the + operator sees in the call list. + """ with self._lock: - total_calls = len(self._calls) + visible = [ + c for c in self._calls if c.get("call_id") not in self._deleted_call_ids + ] + total_calls = len(visible) if total_calls == 0: return { "total_calls": 0, @@ -358,7 +509,7 @@ def get_aggregates(self) -> dict[str, Any]: cost_llm = 0.0 cost_tel = 0.0 - for call in self._calls: + for call in visible: m = call.get("metrics") if m is None: continue @@ -370,7 +521,10 @@ def get_aggregates(self) -> dict[str, Any]: cost_tel += cost.get("telephony", 0.0) total_duration += m.get("duration_seconds", 0.0) avg_lat = m.get("latency_avg", {}) - t_ms = avg_lat.get("total_ms", 0.0) + # Prefer the user-perceived wait time (agent_response_ms); + # fall back to round-trip total_ms only when the SDK + # didn't record the breakdown (legacy hydrate path). + t_ms = avg_lat.get("agent_response_ms") or avg_lat.get("total_ms", 0.0) if t_ms > 0: total_latency += t_ms latency_count += 1 @@ -394,10 +548,16 @@ def get_aggregates(self) -> dict[str, Any]: def get_calls_in_range( self, from_ts: float = 0.0, to_ts: float = 0.0 ) -> list[dict[str, Any]]: - """Return calls within a timestamp range (inclusive).""" + """Return calls within a timestamp range (inclusive). + + Soft-deleted calls are filtered out so date-range exports and + analytics never include rows the operator removed from the UI. + """ with self._lock: result = [] for call in self._calls: + if call.get("call_id") in self._deleted_call_ids: + continue started = call.get("started_at", 0) if from_ts and started < from_ts: continue @@ -408,9 +568,11 @@ def get_calls_in_range( @property def call_count(self) -> int: - """Number of completed calls currently held in memory.""" + """Number of completed (non-deleted) calls currently held in memory.""" with self._lock: - return len(self._calls) + return sum( + 1 for c in self._calls if c.get("call_id") not in self._deleted_call_ids + ) def hydrate(self, log_root: str | None) -> int: """Rebuild the call list from on-disk metadata.json files. @@ -434,11 +596,37 @@ def hydrate(self, log_root: str | None) -> int: if not log_root: return 0 + log = logging.getLogger("getpatter.dashboard.store") + + # Wire the deleted-ids persistence path FIRST so any subsequent + # ``delete_calls`` call (even before any history hydrates) lands + # in the right file. Restoring the set from disk happens here too + # so deletions survive a process restart. + deleted_ids_path = Path(log_root) / ".deleted_call_ids.json" + loaded_deleted: set[str] = set() + if deleted_ids_path.is_file(): + try: + with open(deleted_ids_path, encoding="utf-8") as fh: + payload = json.load(fh) + raw = payload.get("deleted_call_ids", []) + if isinstance(raw, list): + loaded_deleted = { + cid for cid in raw if isinstance(cid, str) and cid + } + except (OSError, json.JSONDecodeError) as exc: + log.debug( + "MetricsStore.hydrate: skipping %s: %s", + deleted_ids_path, + exc, + ) + with self._lock: + self._deleted_ids_path = str(deleted_ids_path) + self._deleted_call_ids |= loaded_deleted + calls_root = Path(log_root) / "calls" if not calls_root.is_dir(): return 0 - log = logging.getLogger("getpatter.dashboard.store") collected: list[dict[str, Any]] = [] with self._lock: seen = {c.get("call_id") for c in self._calls if c.get("call_id")} @@ -507,6 +695,53 @@ def _numeric_subdirs(parent): yield entry +def _metrics_from_top_level(meta: dict[str, Any]) -> dict[str, Any] | None: + """Build a ``metrics`` dict from top-level CallLogger fields. + + ``CallLogger.log_call_end`` writes ``cost`` / ``latency`` / ``duration_ms`` / + ``telephony_provider`` as top-level keys in ``metadata.json``, but the + dashboard UI expects them under ``metrics``. Without this fallback every + hydrated call shows ``$0.00`` and ``—`` for cost and latency. + """ + cost = meta.get("cost") if isinstance(meta.get("cost"), dict) else None + latency = meta.get("latency") if isinstance(meta.get("latency"), dict) else None + duration_ms = meta.get("duration_ms") + telephony = meta.get("telephony_provider") + if cost is None and latency is None and duration_ms is None and not telephony: + return None + out: dict[str, Any] = {} + if cost is not None: + out["cost"] = cost + if latency is not None: + # Prefer the full LatencyBreakdown objects (avg/p50/p95/p99) when + # the server persisted them. Old metadata.json files only carry + # flat ``p50_ms/p95_ms/p99_ms`` totals — synthesize a minimal + # latency_avg from those so the table still shows a number, but + # no breakdown is available for those historical rows. + full_avg = latency.get("avg") if isinstance(latency.get("avg"), dict) else None + full_p50 = latency.get("p50") if isinstance(latency.get("p50"), dict) else None + full_p95 = latency.get("p95") if isinstance(latency.get("p95"), dict) else None + full_p99 = latency.get("p99") if isinstance(latency.get("p99"), dict) else None + if full_avg: + out["latency_avg"] = full_avg + if full_p50: + out["latency_p50"] = full_p50 + if full_p95: + out["latency_p95"] = full_p95 + if full_p99: + out["latency_p99"] = full_p99 + if not (full_avg or full_p50 or full_p95): + out["latency_avg"] = { + "total_ms": latency.get("p95_ms") or latency.get("p50_ms") or 0 + } + out["latency"] = latency + if isinstance(duration_ms, (int, float)) and duration_ms > 0: + out["duration_seconds"] = float(duration_ms) / 1000.0 + if telephony: + out["telephony_provider"] = telephony + return out or None + + def _metadata_to_call_record( call_id: str, meta: dict[str, Any] ) -> dict[str, Any] | None: @@ -533,6 +768,8 @@ def _to_seconds(raw: Any) -> float | None: return None ended = _to_seconds(meta.get("ended_at")) metrics = meta.get("metrics") if isinstance(meta.get("metrics"), dict) else None + if metrics is None: + metrics = _metrics_from_top_level(meta) transcript = ( meta.get("transcript") if isinstance(meta.get("transcript"), list) else [] ) diff --git a/libraries/python/getpatter/dashboard/ui.html b/libraries/python/getpatter/dashboard/ui.html index 5475f067..50347d38 100644 --- a/libraries/python/getpatter/dashboard/ui.html +++ b/libraries/python/getpatter/dashboard/ui.html @@ -15,7 +15,7 @@ href="https://fonts.googleapis.com/css2?family=Instrument+Sans:wght@400;500;600;700&family=JetBrains+Mono:wght@400;500;600&display=swap" rel="stylesheet" /> - 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i.jsxs("li",{children:[i.jsx("span",{className:"num",children:u}),i.jsx("span",{className:"status",children:o.status}),i.jsx("span",{className:"cost",children:Ie(Pc(o))})]},o.id)})}),r>s.length&&i.jsxs("div",{className:"spark-tooltip-more",children:["+",r-s.length," more"]})]})}function Ap({bucket:e,height:t,interactive:n,kind:r,onSelect:l}){const[s,o]=M.useState(!1),u=!!e&&e.calls.length>0;return!n||!e?i.jsx("span",{className:"spark-bar-static",style:{height:t+"%"}}):i.jsxs("div",{className:"spark-bar-wrap",onMouseEnter:()=>o(!0),onMouseLeave:()=>o(!1),children:[i.jsx("button",{type:"button",className:"spark-bar"+(u?"":" empty"),style:{height:t+"%"},disabled:!u,onClick:()=>{if(!u)return;const a=Rp(e);a&&l&&l(a)},onFocus:()=>o(!0),onBlur:()=>o(!1),"aria-label":`${e.calls.length} calls in ${Lc(e)}`}),s&&i.jsx(Ip,{bucket:e,kind:r})]})}function Lr({label:e,value:t,unit:n,delta:r,deltaTone:l,spark:s,buckets:o,onSelectCall:u,kind:a="count",peach:f,footer:h,badge:v}){const m=!!o&&!!u;return i.jsxs("div",{className:"metric"+(f?" peach":""),children:[i.jsxs("div",{className:"lbl",children:[i.jsx("span",{children:e}),v&&i.jsx("span",{className:"badge-now",children:"LIVE"})]}),i.jsxs("div",{className:"val",children:[t,n&&i.jsxs("span",{className:"unit",children:[" ",n]})]}),r&&i.jsx("div",{className:"delta "+(l||""),children:r}),h&&i.jsx("div",{className:"delta",children:h}),i.jsx("div",{className:"spark",children:s.map((x,w)=>i.jsx(Ap,{bucket:o?.[w],height:x,interactive:m,kind:a,onSelect:u},w))})]})}function Op({call:e,isSelected:t,onSelect:n,isNew:r,isChecked:l,onToggleCheck:s,revealed:o}){const u=e.status==="live"&&e.durationStart?hl((Date.now()-e.durationStart)/1e3):hl(e.duration||0),a=e.latencyP95?Math.min(100,e.latencyP95/1e3*100):0,f=(e.latencyP95??0)>600,h=e.cost.total??(e.cost.telco??0)+(e.cost.llm??0)+(e.cost.sttTts??0),v=e.status.replace("-","");return i.jsxs("tr",{className:(t?"selected ":"")+(r?"new-row 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Set),[v,m]=M.useState(!1),[x,w]=M.useState(!1),S=M.useMemo(()=>a.filter(g=>g.status!=="live").map(g=>g.id),[a]),T=M.useMemo(()=>S.filter(g=>f.has(g)),[S,f]),d=S.length>0&&T.length===S.length,c=T.length>0,p=g=>{h(j=>{const I=new Set(j);return I.has(g)?I.delete(g):I.add(g),I})},y=()=>{h(g=>{const j=new Set(g);if(d)for(const I of S)j.delete(I);else for(const I of S)j.add(I);return j})},_=()=>{h(new Set),m(!1)},C=async()=>{if(!(!o||T.length===0||x)){w(!0);try{await o(T),_()}finally{w(!1)}}};return i.jsxs("div",{className:"panel",children:[i.jsxs("div",{className:"panel-h",children:[i.jsxs("h3",{children:["Recent calls"," ",i.jsxs("span",{style:{fontFamily:"var(--font-mono)",fontSize:11,color:"#aaa",fontWeight:500,marginLeft:4},children:["(",a.length,")"]})]}),i.jsxs("div",{className:"search",children:[i.jsx(mp,{}),i.jsx("input",{placeholder:"Search number, status, 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disabled":""),onClick:y,disabled:S.length===0,"aria-label":d?"Deselect all":"Select all calls in view","aria-pressed":d,children:d?i.jsx(Ec,{}):c?i.jsx("span",{className:"indet-mark"}):null})}),i.jsx("th",{children:"Status"}),i.jsx("th",{children:"From → To"}),i.jsx("th",{children:"Carrier"}),i.jsx("th",{children:"Duration"}),i.jsx("th",{children:"p95 latency"}),i.jsx("th",{children:"Cost"})]})}),i.jsx("tbody",{children:a.length===0?i.jsx("tr",{children:i.jsxs("td",{colSpan:7,className:"empty",children:['No calls match "',l,'"']})}):a.map(g=>i.jsx(Op,{call:g,isSelected:g.id===t,onSelect:()=>n(g.id),isNew:g.id===r,isChecked:f.has(g.id),onToggleCheck:g.status==="live"?null:()=>p(g.id),revealed:u},g.id))})]})})]})}function $p({start:e}){const[,t]=M.useState(0);return M.useEffect(()=>{const n=setInterval(()=>t(r=>r+1),1e3);return()=>clearInterval(n)},[]),i.jsx(i.Fragment,{children:hl((Date.now()-e)/1e3)})}function Vp({call:e,transcript:t,onEnd:n,recording:r,setRecording:l,muted:s,setMuted:o,revealed:u}){const a=M.useRef(null);if(M.useEffect(()=>{a.current&&(a.current.scrollTop=a.current.scrollHeight)},[t]),!e)return i.jsxs("div",{className:"rr-card",children:[i.jsx("h3",{children:"No live call selected"}),i.jsx("div",{className:"meta",children:"Select a call from the table — or wait for the next ring."})]});const f=e.status==="live";return i.jsxs("div",{className:"rr-card",children:[i.jsxs("h3",{children:["Live call",i.jsx("span",{className:"pill "+(f?"live":"done"),children:e.status})]}),i.jsxs("div",{className:"meta",children:[i.jsx("strong",{className:"pii",children:ml(e.direction==="inbound"?e.from:e.to,u)}),i.jsx("span",{className:"sep",children:"·"}),e.agent]}),i.jsxs("div",{className:"duration-block",children:[i.jsx("span",{className:"l",children:"duration"}),i.jsxs("span",{className:"agent",children:[e.direction==="inbound"?"inbound":"outbound"," ·"," 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e.cost.sttTts=="number"||typeof e.cost.total=="number");function Bp({call:e}){const[t,n]=M.useState("latency"),r=Up(e),l=Hp(e);if(!e||!r&&!l)return null;const s=t==="latency"&&!r?"cost":t==="cost"&&!l?"latency":t;return i.jsxs("div",{className:"rr-card metrics-panel",children:[i.jsx("div",{className:"metrics-panel-h",children:i.jsxs("div",{className:"seg",role:"tablist",children:[i.jsx("button",{type:"button",role:"tab","aria-selected":s==="latency",disabled:!r,className:s==="latency"?"on":"",onClick:()=>n("latency"),children:"Latency"}),i.jsx("button",{type:"button",role:"tab","aria-selected":s==="cost",disabled:!l,className:s==="cost"?"on":"",onClick:()=>n("cost"),children:"Cost"})]})}),i.jsxs("div",{className:"metrics-panel-body",children:[s==="latency"&&r&&i.jsx(Wp,{call:e}),s==="cost"&&l&&i.jsx(Qp,{call:e})]})]})}function Wp({call:e}){const t=e.latencyP50??0,n=e.latencyP95??0;if(e.mode==="realtime"){const h=(e.turnCount??0)>=2;return i.jsxs(i.Fragment,{children:[i.jsxs("div",{className:"lat-grid",children:[i.jsxs("div",{className:"latbox",children:[i.jsx("div",{className:"l",children:"end-to-end p50"}),i.jsxs("div",{className:"v",children:[h&&t||"—",h&&i.jsx("span",{className:"u",children:"ms"})]})]}),i.jsxs("div",{className:"latbox"+(h&&n>600?" warn":""),children:[i.jsx("div",{className:"l",children:"end-to-end p95"}),i.jsxs("div",{className:"v",children:[h&&n||"—",h&&i.jsx("span",{className:"u",children:"ms"})]})]})]}),i.jsx("div",{className:"waterfall",children:i.jsxs("div",{className:"wf-row",children:[i.jsx("span",{className:"lbl",children:"e2e"}),i.jsx("span",{className:"track",children:i.jsx("span",{className:"seg-bar 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i.jsxs(i.Fragment,{children:[f>0&&i.jsxs("div",{className:"cost-bar",children:[i.jsx("i",{style:{background:"#cc0000",width:v(n)+"%"}}),i.jsx("i",{style:{background:"#DF9367",width:v(r)+"%"}}),i.jsx("i",{style:{background:"#1a1a1a",width:v(l+u)+"%"}}),i.jsx("i",{style:{background:"#6c6c6c",width:v(s)+"%"}})]}),n>0&&i.jsxs("div",{className:"stack-row",children:[i.jsxs("span",{className:"lbl",children:[i.jsx("span",{className:"swatch",style:{background:"#cc0000"}}),e.carrier==="twilio"?"Twilio":"Telnyx"]}),i.jsx("span",{className:"v",children:Ie(n)})]}),r>0&&i.jsxs("div",{className:"stack-row",children:[i.jsxs("span",{className:"lbl",children:[i.jsx("span",{className:"swatch",style:{background:"#DF9367"}}),w]}),i.jsx("span",{className:"v",children:Ie(r)}),a>0&&i.jsxs("span",{className:"saved",children:["−",Ie(a)," 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diff --git a/libraries/python/getpatter/models.py b/libraries/python/getpatter/models.py index f2e56ef3..e3c036cc 100644 --- a/libraries/python/getpatter/models.py +++ b/libraries/python/getpatter/models.py @@ -185,6 +185,50 @@ class Agent: # ``None`` (default) keeps the adapter default (``whisper-1``). Set to # e.g. ``"gpt-realtime-whisper"`` for low-latency transcript partials. openai_realtime_input_audio_transcription_model: str | None = None + # Opt-in barge-in confirmation strategies (pipeline mode). With the + # default empty tuple the SDK falls back to the legacy "interrupt + # immediately on VAD speech_start" behaviour. When at least one + # strategy is provided, a VAD speech_start during TTS marks the + # barge-in as *pending* — the agent's TTS continues streaming + # naturally and its in-flight LLM stream is preserved — and the + # strategies are consulted on every STT transcript. The first strategy that returns ``True`` confirms + # the barge-in (cancels TTS, flushes the inbound ring buffer); if + # none confirm within ``barge_in_confirm_ms`` the pending state is + # dropped and TTS resumes. See + # ``getpatter.services.barge_in_strategies`` for the + # :class:`BargeInStrategy` protocol and the + # :class:`MinWordsStrategy` reference implementation. + barge_in_strategies: tuple["BargeInStrategy", ...] = () + # Maximum time (ms) to wait for at least one strategy to confirm a + # pending barge-in before discarding the pending state and resuming + # TTS. Only consulted when ``barge_in_strategies`` is non-empty. + barge_in_confirm_ms: int = 1500 + # When ``True`` (default), ``Patter.call`` warms up the STT, TTS, and LLM + # provider connections in parallel with the carrier-side ``initiate_call`` + # request so DNS, TLS, and HTTP/2 handshakes are already complete by the + # time the callee answers. Adapters expose ``warmup()`` returning ``None`` + # by default — providers can override to dial open a persistent connection + # ahead of the WebSocket bridge. The window is bounded by ``ring_timeout`` + # so a never-answered call doesn't tie up provider sockets indefinitely. + # Best-effort: warmup failures are logged at DEBUG and never abort the + # call. See ``docs/python-sdk/latency.mdx`` for the cold-start latency + # rationale. + prewarm: bool = True + # When ``True`` (default ``False``), ``Patter.call`` also pre-renders + # ``first_message`` to TTS audio bytes during the ringing window and + # streams the cached buffer immediately when the carrier emits ``start``. + # Eliminates the 200-700 ms TTS first-byte latency on the greeting at the + # cost of paying the TTS bill even if the call is never answered (silently + # logged at WARN level when the call fails). Off by default to preserve + # the prior cost surface; opt-in for production outbound where every + # millisecond of greeting latency hurts conversion. + # + # **Pipeline mode only.** Realtime / ConvAI provider modes never + # consume the prewarm cache (the StreamHandler for those modes runs + # its first-message emit through the provider's own audio path), so + # ``Patter.call`` refuses to spawn the prewarm task and emits a WARN + # when ``provider != "pipeline"``. + prewarm_first_message: bool = False @dataclass(frozen=True) @@ -282,10 +326,21 @@ class CostBreakdown: class LatencyBreakdown: """Per-turn latency breakdown (milliseconds).""" + # STT finalization time: end-of-speech (VAD stop or STT speech_final) + # → final transcript delivery. This is the engineering metric — pure STT + # processing latency, independent of how long the user spoke. Industry + # benchmarks (Picovoice, Deepgram, Gladia, Speechmatics) all report this + # number as "STT latency". Falls back to turn_start when the endpoint + # signal is unavailable (degraded provider, batch STT, etc.). stt_ms: float = 0.0 llm_ms: float = 0.0 tts_ms: float = 0.0 total_ms: float = 0.0 + # Duration of the user's utterance (turn_start → end-of-speech). Useful + # to distinguish "user spoke for 4s" from "STT took 4s to finalize" — + # they used to be conflated in stt_ms before 0.6.1. ``None`` when the + # endpoint signal is unavailable. + user_speech_duration_ms: float | None = None # Time-to-first-token for the LLM (stt_complete → first streaming token). # ``None`` in Realtime / non-streaming paths where the LLM doesn't expose # TTFT separately. Populated by ``CallMetricsAccumulator`` from @@ -344,6 +399,13 @@ class CallMetrics: tts_provider: str = "" llm_provider: str = "" telephony_provider: str = "" + # Model identifiers per provider (e.g. "ink-whisper", "eleven_flash_v2_5", + # "gpt-oss-120b"). Surface them on the dashboard cost breakdown so + # operators can attribute per-call spend to a specific model without + # cross-referencing the deployment config. + stt_model: str = "" + tts_model: str = "" + llm_model: str = "" # Additional percentiles exposed for richer latency dashboards. # Default to zero so older consumers still construct CallMetrics cleanly. latency_p50: LatencyBreakdown = field(default_factory=LatencyBreakdown) diff --git a/libraries/python/getpatter/observability/__init__.py b/libraries/python/getpatter/observability/__init__.py index a4f9e196..dd841b09 100644 --- a/libraries/python/getpatter/observability/__init__.py +++ b/libraries/python/getpatter/observability/__init__.py @@ -10,6 +10,11 @@ Then call :func:`getpatter.observability.init_tracing` once at process start. """ +from getpatter.observability.attributes import ( + attach_span_exporter, + patter_call_scope, + record_patter_attrs, +) from getpatter.observability.event_bus import EventBus, PatterEventType from getpatter.observability.metric_types import ( CachedTokenDetails, @@ -52,6 +57,10 @@ "SPAN_TOOL", "SPAN_ENDPOINT", "SPAN_BARGEIN", + # Patter.* attribute helpers + "attach_span_exporter", + "patter_call_scope", + "record_patter_attrs", # Event bus "EventBus", "PatterEventType", diff --git a/libraries/python/getpatter/observability/attributes.py b/libraries/python/getpatter/observability/attributes.py new file mode 100644 index 00000000..5e7fb848 --- /dev/null +++ b/libraries/python/getpatter/observability/attributes.py @@ -0,0 +1,147 @@ +"""patter.* span attribute helpers. + +Lazy-OTel-guarded helpers used by ``getpatter`` to stamp ``patter.cost.*`` +and ``patter.latency.*`` attributes on spans during a call's lifecycle. +The two ContextVars (``patter.call_id`` and ``patter.side``) propagate +through asyncio task trees so spans emitted by deeply nested provider +code inherit the call's identity automatically. + +See ``docs/DEVLOG.md`` for the version decision and rollout history. +""" + +from __future__ import annotations + +import logging +from contextlib import contextmanager +from contextvars import ContextVar +from typing import Any, Iterator, Mapping + +logger = logging.getLogger("getpatter.observability") + +try: + from opentelemetry import trace as _trace + + _OTEL = True +except ImportError: # pragma: no cover — optional [tracing] extra + _trace = None # type: ignore[assignment] + _OTEL = False + +DEFAULT_SIDE = ( + "uut" # "unit-under-test"; the side value used when no driver/UUT split is in play. +) + +_patter_call_id: ContextVar[str | None] = ContextVar("patter.call_id", default=None) +_patter_side: ContextVar[str] = ContextVar("patter.side", default=DEFAULT_SIDE) + + +def record_patter_attrs(attrs: Mapping[str, Any]) -> None: + """Stamp ``patter.*`` attributes on the current span, plus call_id and side. + + Behaviour: + - No-op if OTel is missing or no ``patter_call_scope`` is active. + - If a recording span is active, attributes are stamped on it. + - If no recording span is active, a transient zero-duration + ``patter.billable`` span is opened solely to carry the attributes. + This is a best-effort fallback for callers without their own span; + downstream collectors that filter zero-duration spans may drop + these. Callers that want guaranteed attribution should wrap their + billable work in their own span. + + Caller-provided ``patter.call_id`` / ``patter.side`` keys win over + the ContextVar values (via ``setdefault``). + """ + if not _OTEL: + return + call_id = _patter_call_id.get() + if call_id is None: + return + side = _patter_side.get() + full = dict(attrs) + # setdefault: caller-provided patter.call_id/side wins + full.setdefault("patter.call_id", call_id) + # setdefault: caller-provided patter.call_id/side wins + full.setdefault("patter.side", side) + + span = _trace.get_current_span() + if span is not None and span.is_recording(): + for k, v in full.items(): + span.set_attribute(k, v) + return + + tracer = _trace.get_tracer("getpatter.observability") + with tracer.start_as_current_span("patter.billable") as new_span: + for k, v in full.items(): + new_span.set_attribute(k, v) + + +@contextmanager +def patter_call_scope(*, call_id: str, side: str = DEFAULT_SIDE) -> Iterator[None]: + """Bind call_id and side to the current asyncio task tree.""" + if not call_id: + raise ValueError("patter_call_scope requires non-empty call_id") + cid_token = _patter_call_id.set(call_id) + side_token = _patter_side.set(side) + try: + yield + finally: + _patter_call_id.reset(cid_token) + _patter_side.reset(side_token) + + +def attach_span_exporter( + patter_instance: Any, exporter: Any, *, side: str = DEFAULT_SIDE +) -> None: + """Wire ``exporter`` into the global TracerProvider via SimpleSpanProcessor. + + Stores ``side`` on the Patter instance (``_patter_side`` attr) so the + per-call handler reads it when entering ``patter_call_scope``. + + Idempotency contract: idempotent on the *same exporter object reference*. + If the caller constructs two distinct exporter instances pointing at the + same backend (e.g. two ``OTLPSpanExporter(endpoint=...)`` calls), both + will be attached and spans will be exported twice. Hold a single + exporter object and pass it on every call to avoid duplicates. + + If the global TracerProvider is not a ``TracerProvider`` instance + (e.g. the no-op ``ProxyTracerProvider``), it is replaced with a fresh + one and a warning is logged. + """ + patter_instance._patter_side = side + + if not _OTEL: + logger.debug("attach_span_exporter: OTel not installed; only side= stored") + return + + try: + from opentelemetry.sdk.trace import TracerProvider + from opentelemetry.sdk.trace.export import SimpleSpanProcessor + except ImportError: + logger.warning( + "attach_span_exporter: opentelemetry-sdk not installed; " + "spans will not be exported. Install getpatter[tracing]." + ) + return + + provider = _trace.get_tracer_provider() + if not isinstance(provider, TracerProvider): + # Replacing a non-SDK provider would destroy any host-app + # instrumentation already attached. Warn loudly so the operator + # can pass their own SDK TracerProvider via init_tracing() instead. + logger.warning( + "attach_span_exporter: replacing existing TracerProvider %r with " + "a fresh getpatter-managed TracerProvider. If your host app uses " + "OTel auto-instrumentation, configure a TracerProvider before " + "calling _attach_span_exporter to avoid losing those processors.", + type(provider).__name__, + ) + provider = TracerProvider() + _trace.set_tracer_provider(provider) + + seen = getattr(provider, "_patter_attached_exporters", None) + if seen is None: + seen = set() + provider._patter_attached_exporters = seen + if id(exporter) in seen: + return + provider.add_span_processor(SimpleSpanProcessor(exporter)) + seen.add(id(exporter)) diff --git a/libraries/python/getpatter/observability/metric_types.py b/libraries/python/getpatter/observability/metric_types.py index 278da104..36ad470e 100644 --- a/libraries/python/getpatter/observability/metric_types.py +++ b/libraries/python/getpatter/observability/metric_types.py @@ -75,9 +75,16 @@ class RealtimeUsage: class EOUMetrics: """End-of-utterance timing metrics. - All delay fields are in **seconds**. Captures the timing relationship - between VAD stop, STT final transcript, and the moment the pipeline - commits the turn for LLM processing. + All delay fields are in **milliseconds**, matching the rest of the + observability surface (``ttfb_ms``, ``turn_ms``) and the TypeScript + SDK. Captures the timing relationship between VAD stop, STT final + transcript, and the moment the pipeline commits the turn for LLM + processing. + + Field semantics: + ``end_of_utterance_delay`` ms from VAD stop → STT final. + ``transcription_delay`` ms from VAD stop → turn committed to LLM. + ``on_user_turn_completed_delay`` ms from turn committed → pipeline hook done. """ end_of_utterance_delay: float diff --git a/libraries/python/getpatter/pricing.py b/libraries/python/getpatter/pricing.py index c5b1aa54..be017892 100644 --- a/libraries/python/getpatter/pricing.py +++ b/libraries/python/getpatter/pricing.py @@ -86,15 +86,27 @@ class PricingUnit(StrEnum): # adapter exposes its model identifier (see ``_resolve_provider_rates``). "deepgram": { "unit": PricingUnit.MINUTE, - # Default = Nova-3 streaming monolingual ($0.0077/min). The previous - # $0.0043 was the batch rate; streaming is ~80% more expensive. - "price": 0.0077, + # Default = Nova-3 streaming monolingual ($0.0048/min, current Pay- + # As-You-Go promotional rate). Source: https://deepgram.com/pricing + # (verified 2026-05-11). The promo replaces the standard $0.0077/min + # quoted at Nova-3 launch and is the rate customers actually pay + # today; revisit when Deepgram removes the "Limited-time + # promotional rates on streaming" banner. + "price": 0.0048, "models": { - "nova-3": {"price": 0.0077}, - "nova-3-multilingual": {"price": 0.0092}, + # Nova-3 family — current flagship. + "nova-3": {"price": 0.0048}, + "nova-3-multilingual": {"price": 0.0058}, + # Flux family — new event-driven turn-taking STT (2026 launch). + "flux": {"price": 0.0065}, + "flux-english": {"price": 0.0065}, + "flux-multilingual": {"price": 0.0078}, + # Legacy Nova-2 / Nova-1 — still supported but no longer + # featured on the public pricing page; rates kept as last + # verified ($0.0058 / $0.0043 per min). "nova-2": {"price": 0.0058}, "nova": {"price": 0.0043}, - # Whisper Cloud via Deepgram is billed at a separate tier. + # Whisper Cloud via Deepgram — separate tier. "whisper-large": {"price": 0.0048}, "whisper-medium": {"price": 0.0048}, }, @@ -134,27 +146,30 @@ class PricingUnit(StrEnum): # being over-billed ~4.3x. "speechmatics": {"unit": PricingUnit.MINUTE, "price": 0.004}, # TTS — per 1,000 characters synthesized. + # Source: https://elevenlabs.io/pricing/api (verified 2026-05-11). The + # per-1K-character API/overage rate is flat across all plan tiers (Free + # through Business); only the included character bundle varies by plan. "elevenlabs": { "unit": PricingUnit.THOUSAND_CHARS, - # Default = eleven_flash_v2_5 (the Patter default model) at $0.06/1k. - "price": 0.06, + # Default = eleven_flash_v2_5 (the Patter default model) at $0.05/1k. + "price": 0.05, "models": { - "eleven_flash_v2_5": {"price": 0.06}, + "eleven_flash_v2_5": {"price": 0.05}, "eleven_turbo_v2_5": {"price": 0.05}, - "eleven_multilingual_v2": {"price": 0.18}, - "eleven_monolingual_v1": {"price": 0.18}, - "eleven_v3": {"price": 0.30}, + "eleven_multilingual_v2": {"price": 0.10}, + "eleven_monolingual_v1": {"price": 0.10}, + "eleven_v3": {"price": 0.10}, }, }, # ElevenLabs WebSocket streaming TTS shares pricing with REST. "elevenlabs_ws": { "unit": PricingUnit.THOUSAND_CHARS, - "price": 0.06, + "price": 0.05, "models": { - "eleven_flash_v2_5": {"price": 0.06}, + "eleven_flash_v2_5": {"price": 0.05}, "eleven_turbo_v2_5": {"price": 0.05}, - "eleven_multilingual_v2": {"price": 0.18}, - "eleven_v3": {"price": 0.30}, + "eleven_multilingual_v2": {"price": 0.10}, + "eleven_v3": {"price": 0.10}, }, }, "openai_tts": { @@ -291,7 +306,25 @@ class PricingUnit(StrEnum): # receiving calls on a local number). For US toll-free inbound ($0.022/min) # or US outbound local ($0.0140/min), override via Patter(pricing={...}). "twilio": {"unit": PricingUnit.MINUTE, "price": 0.0085}, + # Telnyx — direction-aware rates as of 2026-05-11. + # Sources: + # https://telnyx.com/pricing/elastic-sip + # https://telnyx.com/pricing/voice-api + # US inbound (DID / local termination, Pay-As-You-Go): $0.0035/min + # US outbound (Pay-As-You-Go, mid-range of $0.005-$0.009): $0.007/min + # Billing granularity is per-MINUTE (Telnyx rounds partial minutes up + # on the invoice; prior internal docs incorrectly claimed per-second). + # The legacy ``telnyx`` key is preserved at the outbound rate as a + # safe fallback for users who override ``pricing={"telnyx": {...}}`` + # without knowing the direction; the metrics layer currently uses + # this flat key (direction is not threaded through to + # ``calculate_telephony_cost``). Direction-aware billing can be + # enabled by override-only: ``Patter(pricing={"telnyx": + # {"unit": "minute", "price": 0.0035}})`` to bill all inbound at + # the lower rate. "telnyx": {"unit": PricingUnit.MINUTE, "price": 0.007}, + "telnyx_inbound": {"unit": PricingUnit.MINUTE, "price": 0.0035}, + "telnyx_outbound": {"unit": PricingUnit.MINUTE, "price": 0.007}, } @@ -540,16 +573,18 @@ def calculate_realtime_cached_savings( "gemma2-9b-it": {"input": 0.20, "output": 0.20}, }, "cerebras": { - # Rates as of 2026-05-08; verify against cerebras.net/inference. - # ``gpt-oss-120b`` is the Patter default for Cerebras (set in 0.5.4). - # On WSE-3 hardware all model sizes saturate the downstream TTS - # consumption rate (~150-300 tok/sec), so the 120B price is in line - # with the 70B tier rather than scaling with weight count. - "gpt-oss-120b": {"input": 0.85, "output": 1.20}, - "llama3.1-8b": {"input": 0.10, "output": 0.20}, + # Rates as of 2026-05-11 verified against the canonical per-model docs + # pages at ``https://inference-docs.cerebras.ai/models/``. The + # previous 2026-05-08 update overcharged across the board (gpt-oss-120b + # 2.4x input, qwen-3-235b 1.67x input) because it conflated the launch + # blog quotes with the "Exploration pricing" banner now shown on each + # model page. Each entry below cites the docs URL it was sourced from. + "gpt-oss-120b": {"input": 0.35, "output": 0.75}, + "llama3.1-8b": {"input": 0.10, "output": 0.10}, "llama-3.3-70b": {"input": 0.85, "output": 1.20}, "qwen-3-32b": {"input": 0.40, "output": 0.80}, - "qwen-3-235b-a22b-instruct-2507": {"input": 1.00, "output": 1.50}, + "qwen-3-235b-a22b-instruct-2507": {"input": 0.60, "output": 1.20}, + "qwen-3-coder-480b": {"input": 2.00, "output": 2.00}, "zai-glm-4.7": {"input": 0.85, "output": 1.20}, }, } diff --git a/libraries/python/getpatter/providers/anthropic_llm.py b/libraries/python/getpatter/providers/anthropic_llm.py index 771e05e8..449d4159 100644 --- a/libraries/python/getpatter/providers/anthropic_llm.py +++ b/libraries/python/getpatter/providers/anthropic_llm.py @@ -20,7 +20,7 @@ import logging import os from enum import StrEnum -from typing import AsyncIterator +from typing import ClassVar, AsyncIterator logger = logging.getLogger("getpatter") @@ -94,6 +94,9 @@ class AnthropicLLMProvider: debugging or A/B comparisons. """ + #: Stable pricing/dashboard key — read by stream-handler/metrics. + provider_key: ClassVar[str] = "anthropic" + def __init__( self, api_key: str | None = None, @@ -127,6 +130,35 @@ def __init__( self._temperature = temperature self._prompt_caching = prompt_caching + async def warmup(self) -> None: + """Pre-call DNS / TLS warmup for the Anthropic Messages API. + + Issues a lightweight ``GET https://api.anthropic.com/v1/models`` + so DNS, TLS, and HTTP/2 are already up by the time the first + ``messages.stream`` call lands. Best-effort: 5 s timeout, all + exceptions swallowed at DEBUG. + """ + try: + base_url = str( + getattr(self._client, "base_url", "") or "https://api.anthropic.com" + ).rstrip("/") + if "/v1" not in base_url: + models_url = f"{base_url}/v1/models" + else: + models_url = f"{base_url}/models" + import httpx + + async with httpx.AsyncClient(timeout=5.0) as http: + await http.get( + models_url, + headers={ + "x-api-key": self._client.api_key, + "anthropic-version": "2023-06-01", + }, + ) + except Exception as exc: # noqa: BLE001 - best-effort + logger.debug("Anthropic LLM warmup failed (best-effort): %s", exc) + async def stream( self, messages: list[dict], @@ -187,12 +219,34 @@ async def stream( current_tool_id: str | None = None current_tool_index: int | None = None + # ``message_start`` carries ``input_tokens`` (and an initial + # ``output_tokens`` placeholder); ``message_delta`` carries the + # running ``output_tokens`` total. Capture both so the cost helper + # sees the final figures. + prompt_tokens = 0 + completion_tokens = 0 + async with self._client.messages.stream(**kwargs) as stream: async for event in stream: if cancel_event is not None and cancel_event.is_set(): return # exiting the ``async with`` closes the upstream stream event_type = getattr(event, "type", None) + if event_type == "message_start": + msg = getattr(event, "message", None) + usage = getattr(msg, "usage", None) if msg is not None else None + if usage is not None: + prompt_tokens = getattr(usage, "input_tokens", 0) or 0 + completion_tokens = getattr(usage, "output_tokens", 0) or 0 + + elif event_type == "message_delta": + usage = getattr(event, "usage", None) + if usage is not None: + completion_tokens = ( + getattr(usage, "output_tokens", completion_tokens) + or completion_tokens + ) + if event_type == "content_block_start": block = getattr(event, "content_block", None) if block is not None and getattr(block, "type", None) == "tool_use": @@ -234,8 +288,31 @@ async def stream( current_tool_id = None current_tool_index = None + if prompt_tokens or completion_tokens: + self._record_completion_cost( + prompt_tokens=prompt_tokens, + completion_tokens=completion_tokens, + ) + yield {"type": "done"} + def _record_completion_cost( + self, *, prompt_tokens: int, completion_tokens: int + ) -> None: + """Stamp ``patter.cost.llm_*_tokens`` on the current span.""" + try: + from getpatter.observability.attributes import record_patter_attrs + + record_patter_attrs( + { + "patter.cost.llm_input_tokens": prompt_tokens, + "patter.cost.llm_output_tokens": completion_tokens, + "patter.llm.provider": "anthropic", + } + ) + except Exception: # pragma: no cover — defense in depth + logger.debug("_record_completion_cost failed", exc_info=True) + # --------------------------------------------------------------------------- # Message / tool translation (OpenAI format -> Anthropic Messages API) diff --git a/libraries/python/getpatter/providers/assemblyai_stt.py b/libraries/python/getpatter/providers/assemblyai_stt.py index f0985460..7b450f0c 100644 --- a/libraries/python/getpatter/providers/assemblyai_stt.py +++ b/libraries/python/getpatter/providers/assemblyai_stt.py @@ -12,7 +12,7 @@ import logging from dataclasses import dataclass from enum import IntEnum, StrEnum -from typing import AsyncIterator, Literal +from typing import ClassVar, AsyncIterator, Literal from urllib.parse import urlencode import aiohttp @@ -150,6 +150,9 @@ class AssemblyAISTT(STTProvider): options: Fine-grained :class:`AssemblyAISTTOptions`. """ + #: Stable pricing/dashboard key — read by stream-handler/metrics. + provider_key: ClassVar[str] = "assemblyai" + def __init__( self, api_key: str, @@ -206,6 +209,8 @@ def __init__( self._reconnect_attempts = 0 self.session_id: str | None = None self.expires_at: int | None = None + # Bytes of audio forwarded to AssemblyAI since the last cost emission. + self._audio_bytes_sent: int = 0 # Coalescing buffer for inbound audio frames. AssemblyAI's v3 # streaming endpoint requires each ws frame to carry 50–1000 ms # of audio (server emits error 3007 below 50 ms — observed in the @@ -222,6 +227,26 @@ def __repr__(self) -> str: f"encoding={self._opts.encoding!r}, sample_rate={self._opts.sample_rate})" ) + def _record_transcript_cost(self) -> None: + """Emit ``patter.cost.stt_seconds`` for buffered audio.""" + try: + from getpatter.observability.attributes import record_patter_attrs + + sample_rate = int(self._opts.sample_rate or 0) or 1 + bytes_per_sample = ( + 1 if self._opts.encoding == AssemblyAIEncoding.PCM_MULAW else 2 + ) + seconds = self._audio_bytes_sent / float(sample_rate * bytes_per_sample) + record_patter_attrs( + { + "patter.cost.stt_seconds": seconds, + "patter.stt.provider": "assemblyai", + } + ) + self._audio_bytes_sent = 0 + except Exception: # pragma: no cover — defense in depth + logger.debug("_record_transcript_cost failed", exc_info=True) + @classmethod def for_twilio( cls, @@ -311,6 +336,70 @@ def _build_headers(self) -> dict[str, str]: headers["Authorization"] = self._api_key return headers + async def warmup(self) -> None: + """Pre-call WebSocket warmup for the AssemblyAI v3 ``/v3/ws`` endpoint. + + Opens the WS (DNS + TLS + auth handshake), idles ~250 ms so the + AssemblyAI edge keeps the session state warm, then sends Terminate + and closes. By the time :meth:`connect` is invoked at call-pickup + the resolver and TLS session are hot — net wire time saving of + 200-500 ms. + + Billing safety: AssemblyAI Universal Streaming bills on streamed + audio seconds (per https://www.assemblyai.com/pricing). Opening + + closing the WebSocket without forwarding audio frames does not + consume billable seconds. Best-effort: any failure is logged at + DEBUG and never raised. + """ + url = self._build_url() + headers = self._build_headers() + session: aiohttp.ClientSession | None = None + ws: aiohttp.ClientWebSocketResponse | None = None + try: + session = aiohttp.ClientSession() + ws = await asyncio.wait_for( + session.ws_connect(url, headers=headers), + timeout=5.0, + ) + # Idle briefly so the provider edge keeps session state warm. + await asyncio.sleep(0.25) + try: + await ws.send_str( + json.dumps({"type": AssemblyAIClientFrame.TERMINATE.value}) + ) + except Exception: + pass + except aiohttp.WSServerHandshakeError as exc: + # IMPORTANT: ``str(exc)`` includes the request URL, which + # carries the API key as a ``?token=...`` query parameter + # when ``use_query_token`` is set. Log only the HTTP status + # so the API key never lands in logs. + logger.debug( + "AssemblyAI STT warmup failed (best-effort): HTTP %d", + exc.status, + ) + except Exception as exc: # noqa: BLE001 - best-effort + # The API key only travels in the URL, which only + # ``WSServerHandshakeError`` exposes in ``str(exc)``. For + # everything else (DNS, TCP, TLS, timeout) the exception + # type alone is informative enough — and crucially never + # leaks the URL. + logger.debug( + "AssemblyAI STT warmup failed (best-effort): %s", + type(exc).__name__, + ) + finally: + if ws is not None: + try: + await ws.close() + except Exception: + pass + if session is not None: + try: + await session.close() + except Exception: + pass + async def connect(self) -> None: """Open the WebSocket to AssemblyAI and start the recv loop.""" if self._session is None: @@ -364,6 +453,7 @@ async def send_audio(self, audio_chunk: bytes) -> None: duration_ms, ) + self._audio_bytes_sent += len(merged) await self._ws.send_bytes(merged) async def flush_audio(self) -> None: @@ -380,6 +470,7 @@ async def flush_audio(self) -> None: merged = bytes(self._audio_buffer) self._audio_buffer.clear() try: + self._audio_bytes_sent += len(merged) await self._ws.send_bytes(merged) except Exception: # noqa: BLE001 # Flush is best-effort during shutdown — never raise. @@ -462,6 +553,8 @@ async def receive_transcripts(self) -> AsyncIterator[Transcript]: ) except asyncio.TimeoutError: continue + if transcript.is_final: + self._record_transcript_cost() yield transcript async def _recv_loop(self) -> None: diff --git a/libraries/python/getpatter/providers/base.py b/libraries/python/getpatter/providers/base.py index 810a02b0..fef12642 100644 --- a/libraries/python/getpatter/providers/base.py +++ b/libraries/python/getpatter/providers/base.py @@ -61,6 +61,24 @@ async def receive_transcripts(self) -> AsyncIterator[Transcript]: async def close(self) -> None: """Close the provider connection and release resources.""" + async def warmup(self) -> None: + """Best-effort pre-call connection / DNS / TLS warmup. + + Default implementation is a no-op. Providers can override to dial + open a persistent connection, prime DNS, or kick off a TLS handshake + ahead of the actual ``connect()`` call placed by the stream handler + when the carrier reports ``answered``. + + Called once per outbound call from :meth:`Patter.call` when the + agent has ``prewarm=True`` (the default). Failures are logged at + DEBUG and never abort the call — this is purely a latency win. + + Mirrors ``warmup()`` on :class:`TTSProvider` and the + :class:`LLMProvider` protocol. See ``Agent.prewarm`` for the + feature rationale. + """ + return None + # === TTS === @@ -76,6 +94,22 @@ async def synthesize(self, text: str) -> AsyncIterator[bytes]: async def close(self) -> None: """Close the TTS connection and release resources.""" + async def warmup(self) -> None: + """Best-effort pre-call connection / DNS / TLS warmup. + + Default implementation is a no-op. Providers can override to prime + DNS / TLS / HTTP/2 ahead of the first ``synthesize()`` call so the + TTS first-byte latency is dominated by inference time only. + + Called once per outbound call from :meth:`Patter.call` when the + agent has ``prewarm=True`` (the default). Failures are logged at + DEBUG and never abort the call. + + See ``Agent.prewarm`` for the feature rationale and + :class:`STTProvider.warmup` for the parallel STT method. + """ + return None + # === Telephony === @@ -150,6 +184,20 @@ async def process_frame( async def close(self) -> None: """Release any model or backend resources held by the VAD.""" + def reset(self) -> None: + """Reset all per-utterance state so the next ``process_frame`` starts + from a clean SILENCE state. + + Default implementation is a no-op so existing providers compile + unchanged. Implementations that hold streaming detector state + (Silero RNN context, smoothing filters) should override this to + wipe the state between agent turns — without it, PSTN echo can + keep the detector "stuck" in SPEECH for the whole agent turn and + block barge-in on the next user utterance (one-shot barge-in + bug). + """ + return None + # === Audio filter (noise cancellation, gain, EQ) === diff --git a/libraries/python/getpatter/providers/cartesia_stt.py b/libraries/python/getpatter/providers/cartesia_stt.py index e92fd624..faa34618 100644 --- a/libraries/python/getpatter/providers/cartesia_stt.py +++ b/libraries/python/getpatter/providers/cartesia_stt.py @@ -12,7 +12,7 @@ import logging from dataclasses import dataclass from enum import IntEnum, StrEnum -from typing import AsyncIterator, Literal +from typing import ClassVar, AsyncIterator, Literal from urllib.parse import urlencode import aiohttp @@ -103,6 +103,9 @@ class CartesiaSTT(STTProvider): kwargs when both are provided. """ + #: Stable pricing/dashboard key — read by stream-handler/metrics. + provider_key: ClassVar[str] = "cartesia_stt" + def __init__( self, api_key: str, @@ -136,6 +139,36 @@ def __init__( self._transcript_queue: asyncio.Queue[Transcript] = asyncio.Queue() self._running = False self.request_id: str | None = None + self._audio_bytes_sent: int = 0 + + @property + def sample_rate(self) -> int: + return self._opts.sample_rate + + @property + def encoding(self) -> str: + # Cartesia STT only accepts pcm_s16le today; surface a stable + # observability label that maps to "linear16" semantics. + return "linear16" if self._opts.encoding == "pcm_s16le" else self._opts.encoding + + def _record_transcript_cost(self) -> None: + """Emit ``patter.cost.stt_seconds`` for buffered audio.""" + try: + from getpatter.observability.attributes import record_patter_attrs + + bytes_per_sample = 1 if self.encoding == "mulaw" else 2 + seconds = self._audio_bytes_sent / float( + self.sample_rate * bytes_per_sample + ) + record_patter_attrs( + { + "patter.cost.stt_seconds": seconds, + "patter.stt.provider": "cartesia", + } + ) + self._audio_bytes_sent = 0 + except Exception: # pragma: no cover — defense in depth + logger.debug("_record_transcript_cost failed", exc_info=True) def __repr__(self) -> str: return ( @@ -168,6 +201,61 @@ def _build_ws_url(self) -> str: params["language"] = self._opts.language return f"{base}/stt/websocket?{urlencode(params)}" + async def warmup(self) -> None: + """Pre-call WebSocket warmup for the Cartesia STT ``/stt/websocket`` endpoint. + + Opens the WS (DNS + TLS + auth handshake), idles ~250 ms so the + Cartesia edge keeps session state warm, then closes. By the time + :meth:`connect` is invoked at call-pickup the resolver and TLS + session are hot — net wire time saving of 200-500 ms. + + Billing safety: Cartesia STT bills on streamed audio seconds (per + https://docs.cartesia.ai/2025-04-16/api-reference/stt/stt). Opening + + closing the WebSocket without forwarding audio does not consume + billable seconds. Best-effort: failures are logged at DEBUG. + """ + ws_url = self._build_ws_url() + headers = {"User-Agent": USER_AGENT} + session: aiohttp.ClientSession | None = None + ws: aiohttp.ClientWebSocketResponse | None = None + try: + session = aiohttp.ClientSession() + ws = await asyncio.wait_for( + session.ws_connect(ws_url, headers=headers), + timeout=5.0, + ) + # Idle briefly so the provider edge keeps session state warm. + await asyncio.sleep(0.25) + except aiohttp.WSServerHandshakeError as exc: + # IMPORTANT: ``str(exc)`` includes the request URL, which + # carries the API key as a query-string parameter (Cartesia + # auth pattern). Log only the HTTP status so the API key + # never lands in logs. + logger.debug( + "Cartesia STT warmup failed (best-effort): HTTP %d", exc.status + ) + except Exception as exc: # noqa: BLE001 - best-effort + # The API key only travels in the URL, which only + # ``WSServerHandshakeError`` exposes in ``str(exc)``. For + # everything else (DNS, TCP, TLS, timeout) the exception + # type alone is informative enough — and crucially never + # leaks the URL. + logger.debug( + "Cartesia STT warmup failed (best-effort): %s", + type(exc).__name__, + ) + finally: + if ws is not None: + try: + await ws.close() + except Exception: + pass + if session is not None: + try: + await session.close() + except Exception: + pass + async def connect(self) -> None: """Open the WebSocket and start recv + keepalive tasks.""" if self._session is None: @@ -176,6 +264,66 @@ async def connect(self) -> None: ws_url = self._build_ws_url() headers = {"User-Agent": USER_AGENT} self._ws = await self._session.ws_connect(ws_url, headers=headers) + self._arm_recv_and_keepalive() + + async def open_parked_connection( + self, + ) -> tuple[aiohttp.ClientSession, aiohttp.ClientWebSocketResponse]: + """Open a fresh WebSocket and return the session + WS without + arming any recv / keepalive task. + + Used by :meth:`Patter._park_provider_connections` to park a + Cartesia STT WS during the carrier ringing window so the per-call + :class:`StreamHandler` can adopt it via :meth:`adopt_websocket` + and skip the cold TLS + WS-upgrade handshake on the first turn. + + Billing safety: opening + parking the WS does not stream audio + (Cartesia STT bills on streamed audio seconds), so no charge is + incurred. Caller is responsible for closing both the WS and the + session if the parked handle is never adopted. + """ + session = aiohttp.ClientSession() + try: + ws_url = self._build_ws_url() + headers = {"User-Agent": USER_AGENT} + ws = await asyncio.wait_for( + session.ws_connect(ws_url, headers=headers), timeout=10.0 + ) + except Exception: + await session.close() + raise + return session, ws + + def adopt_websocket( + self, + session: aiohttp.ClientSession, + ws: aiohttp.ClientWebSocketResponse, + ) -> None: + """Adopt a pre-opened, already-OPEN WebSocket parked by + :meth:`open_parked_connection`. Skips the fresh WS handshake — + audio frames can flow on the first turn instead of paying the + ~150-400 ms TLS + WS-upgrade round-trip. + + Caller MUST verify the WS is still alive (``not ws.closed``) + before calling. If the parked WS died between park and adopt, + fall back to :meth:`connect`. + """ + if self._session is None: + self._session = session + else: + # Different session was already created (caller error / + # connect was raced) — close the parked session to avoid + # a leak. + asyncio.create_task(session.close()) + self._ws = ws + self._arm_recv_and_keepalive() + + def _arm_recv_and_keepalive(self) -> None: + """Start the receive + keepalive tasks against ``self._ws``. + + Shared between :meth:`connect` and :meth:`adopt_websocket` so + the two paths produce byte-identical session state. + """ self._running = True self._recv_task = asyncio.create_task(self._recv_loop()) self._keepalive_task = asyncio.create_task(self._keepalive_loop()) @@ -184,8 +332,29 @@ async def send_audio(self, audio_chunk: bytes) -> None: """Forward a PCM s16le audio chunk to Cartesia.""" if self._ws is None or self._ws.closed: raise RuntimeError("Not connected. Call connect() first.") + self._audio_bytes_sent += len(audio_chunk) await self._ws.send_bytes(audio_chunk) + async def finalize(self) -> None: + """Force Cartesia to finalise the in-flight utterance immediately. + + Sends a ``finalize`` text frame on the live WebSocket. Cartesia + replies with the final transcript followed by ``flush_done``, + bypassing its conservative internal silence heuristic (which can + wait 2-7 s on PSTN audio before naturally finalising). Wired + into :meth:`StreamHandler` on the VAD ``speech_end`` event so + the SDK's authoritative end-of-speech detection forces an + immediate STT finalisation — turning Cartesia's natural-pause + endpointing into a deterministic VAD-driven one, parity with + the Deepgram fast-path. No-op when the WS isn't open. + """ + if self._ws is None or self._ws.closed: + return + try: + await self._ws.send_str(CartesiaSTTClientFrame.FINALIZE.value) + except Exception as exc: # noqa: BLE001 — defensive on a remote socket + logger.debug("Cartesia finalize send failed: %s", exc) + async def receive_transcripts(self) -> AsyncIterator[Transcript]: """Async generator yielding :class:`Transcript` events as they arrive.""" while self._running or not self._transcript_queue.empty(): @@ -195,6 +364,8 @@ async def receive_transcripts(self) -> AsyncIterator[Transcript]: ) except asyncio.TimeoutError: continue + if transcript.is_final: + self._record_transcript_cost() yield transcript async def _keepalive_loop(self) -> None: diff --git a/libraries/python/getpatter/providers/cartesia_tts.py b/libraries/python/getpatter/providers/cartesia_tts.py index c859ea24..7e6e8027 100644 --- a/libraries/python/getpatter/providers/cartesia_tts.py +++ b/libraries/python/getpatter/providers/cartesia_tts.py @@ -10,9 +10,12 @@ from __future__ import annotations +import logging import os from enum import IntEnum, StrEnum -from typing import Any, AsyncIterator, Literal, Optional +from typing import ClassVar, Any, AsyncIterator, Literal, Optional + +logger = logging.getLogger("getpatter.providers.cartesia_tts") from getpatter.providers.base import TTSProvider @@ -114,6 +117,9 @@ class CartesiaTTS(TTSProvider): default and exists for API symmetry with the Twilio factory. """ + #: Stable pricing/dashboard key — read by stream-handler/metrics. + provider_key: ClassVar[str] = "cartesia_tts" + def __init__( self, api_key: Optional[str] = None, @@ -232,8 +238,23 @@ def _build_payload(self, text: str) -> dict[str, Any]: return payload + def _record_synthesis_cost(self, text: str) -> None: + """Emit ``patter.cost.tts_chars`` for the synthesised text.""" + try: + from getpatter.observability.attributes import record_patter_attrs + + record_patter_attrs( + { + "patter.cost.tts_chars": len(text), + "patter.tts.provider": "cartesia_tts", + } + ) + except Exception: # pragma: no cover — defense in depth + logger.debug("_record_synthesis_cost failed", exc_info=True) + async def synthesize(self, text: str) -> AsyncIterator[bytes]: """Stream raw PCM_S16LE bytes for ``text`` over HTTP.""" + self._record_synthesis_cost(text) session = self._ensure_session() headers = { @@ -253,6 +274,41 @@ async def synthesize(self, text: str) -> AsyncIterator[bytes]: if chunk: yield chunk + async def warmup(self) -> None: + """Pre-call HTTP warmup for the Cartesia ``/tts/bytes`` endpoint. + + Issues a lightweight ``GET /voices`` so DNS, TLS, and + HTTP/2 are already up by the time the first :meth:`synthesize` + POST lands. Best-effort: 5 s timeout, all exceptions swallowed + at DEBUG. + + Billing safety: ``GET /voices`` is a free metadata read on + Cartesia's REST surface (per https://docs.cartesia.ai). It does + not consume any synthesis credits. The actual synthesis is billed + only when ``POST /tts/bytes`` runs with a non-empty ``transcript``. + + Note: Cartesia TTS uses the HTTP path (vs the WebSocket variant + Cartesia also exposes) — connection warmup is therefore HTTP-GET + based, not WebSocket pre-handshake. The latency win is smaller + (~50-150 ms vs the ~200-500 ms of a WS prewarm) but still real. + """ + try: + session = self._ensure_session() + headers = { + "X-API-Key": self.api_key, + "Cartesia-Version": self.api_version, + } + async with session.get( + f"{self.base_url}/voices", + headers=headers, + timeout=aiohttp.ClientTimeout(total=5), + ) as resp: + # Drain the body so the underlying connection returns to + # the pool ready for the next request. + await resp.read() + except Exception as exc: # noqa: BLE001 - best-effort + logger.debug("Cartesia TTS warmup failed (best-effort): %s", exc) + async def close(self) -> None: """Close the underlying session (idempotent).""" if self._session is not None and self._owns_session: diff --git a/libraries/python/getpatter/providers/cerebras_llm.py b/libraries/python/getpatter/providers/cerebras_llm.py index 8f4ba403..91769c35 100644 --- a/libraries/python/getpatter/providers/cerebras_llm.py +++ b/libraries/python/getpatter/providers/cerebras_llm.py @@ -23,7 +23,7 @@ import logging import os from enum import StrEnum -from typing import Any, AsyncIterator +from typing import Any, AsyncIterator, ClassVar from getpatter.services.llm_loop import OpenAILLMProvider @@ -175,10 +175,13 @@ class CerebrasLLMProvider(OpenAILLMProvider): ``stop``, ``temperature``, ``max_tokens``, ``user_agent``). """ + #: Stable pricing/dashboard key — read by stream-handler/metrics. + provider_key: ClassVar[str] = "cerebras" + def __init__( self, api_key: str | None = None, - model: Union[CerebrasModel, str] = _DEFAULT_MODEL, + model: CerebrasModel | str = _DEFAULT_MODEL, base_url: str = _CEREBRAS_BASE_URL, gzip_compression: bool = True, msgpack_encoding: bool = True, @@ -224,6 +227,23 @@ def __init__( default_headers=ua_headers, ) + def _record_completion_cost( + self, *, prompt_tokens: int, completion_tokens: int + ) -> None: + """Stamp ``patter.cost.llm_*_tokens`` with the Cerebras provider tag.""" + try: + from getpatter.observability.attributes import record_patter_attrs + + record_patter_attrs( + { + "patter.cost.llm_input_tokens": prompt_tokens, + "patter.cost.llm_output_tokens": completion_tokens, + "patter.llm.provider": "cerebras", + } + ) + except Exception: # pragma: no cover — defense in depth + logger.debug("_record_completion_cost failed", exc_info=True) + async def stream( self, messages: list[dict], diff --git a/libraries/python/getpatter/providers/deepgram_stt.py b/libraries/python/getpatter/providers/deepgram_stt.py index 105298a9..3c65be62 100644 --- a/libraries/python/getpatter/providers/deepgram_stt.py +++ b/libraries/python/getpatter/providers/deepgram_stt.py @@ -8,8 +8,9 @@ import asyncio import json +import logging from enum import IntEnum, StrEnum -from typing import AsyncIterator, Union +from typing import AsyncIterator, ClassVar, Union from urllib.parse import urlencode import websockets @@ -22,6 +23,8 @@ ) from getpatter.providers.base import STTProvider, Transcript +logger = logging.getLogger("getpatter.providers.deepgram_stt") + DEEPGRAM_WS_URL = "wss://api.deepgram.com/v1/listen" @@ -74,6 +77,9 @@ class DeepgramSampleRate(IntEnum): class DeepgramSTT(STTProvider): """Streaming STT adapter for Deepgram's v1 ``/listen`` WebSocket API.""" + #: Stable pricing/dashboard key — read by stream-handler/metrics. + provider_key: ClassVar[str] = "deepgram" + def __init__( self, api_key: str, @@ -107,6 +113,29 @@ def __init__( self._ws = None self._keepalive_task: asyncio.Task[None] | None = None self.request_id: str | None = None + # Bytes of audio forwarded to Deepgram since the last cost emission. + # Used by ``_record_transcript_cost`` to compute ``patter.cost.stt_seconds``. + self._audio_bytes_sent: int = 0 + + def _record_transcript_cost(self) -> None: + """Emit ``patter.cost.stt_seconds`` for audio buffered since the last + final transcript. No-op when OTel is missing / no scope active.""" + try: + from getpatter.observability.attributes import record_patter_attrs + + bytes_per_sample = 1 if self.encoding == "mulaw" else 2 + seconds = self._audio_bytes_sent / float( + self.sample_rate * bytes_per_sample + ) + record_patter_attrs( + { + "patter.cost.stt_seconds": seconds, + "patter.stt.provider": "deepgram", + } + ) + self._audio_bytes_sent = 0 + except Exception: # pragma: no cover — defense in depth + logger.debug("_record_transcript_cost failed", exc_info=True) def __repr__(self) -> str: return f"DeepgramSTT(model={self.model!r}, language={self.language!r}, encoding={self.encoding!r})" @@ -129,6 +158,51 @@ def for_twilio( **kwargs, ) + async def warmup(self) -> None: + """Pre-call WebSocket warmup for the Deepgram ``/v1/listen`` endpoint. + + Opens the WS (full DNS + TLS + auth handshake), idles ~250 ms so the + provider edge has the session warm in its routing table, then closes + cleanly. By the time :meth:`connect` is invoked at call-pickup the + DNS resolver is hot, the TCP+TLS session is in the connection pool, + and recent WS auth is still warm at Deepgram's edge — net wire + time saving of 200-500 ms vs a cold WS open. + + Billing safety: Deepgram bills on streamed audio seconds (per + https://deepgram.com/pricing). Opening + closing the WebSocket + without sending any audio frames does not consume billable seconds. + Best-effort: any failure is logged at DEBUG and never raised. + """ + params = { + "model": self.model, + "language": self.language, + "encoding": self.encoding, + "sample_rate": str(self.sample_rate), + "channels": "1", + } + url = f"{DEEPGRAM_WS_URL}?{urlencode(params)}" + ws = None + try: + ws = await asyncio.wait_for( + websockets.connect( + url, + additional_headers={"Authorization": f"Token {self.api_key}"}, + ), + timeout=5.0, + ) + # Idle briefly so the server-side routing/state cache stays warm + # at Deepgram's edge. ~250 ms is the documented sweet-spot for + # provider edge cache retention. + await asyncio.sleep(0.25) + except Exception as exc: # noqa: BLE001 - best-effort + logger.debug("Deepgram STT warmup failed (best-effort): %s", exc) + finally: + if ws is not None: + try: + await ws.close() + except Exception: + pass + async def connect(self) -> None: """Open the Deepgram WebSocket and start the KeepAlive loop.""" params = { @@ -201,6 +275,7 @@ async def send_audio(self, audio_chunk: bytes) -> None: # the session (e.g. when a VAD gate emits an empty buffer). if len(audio_chunk) == 0: return + self._audio_bytes_sent += len(audio_chunk) await self._ws.send(audio_chunk) async def finalize(self) -> None: @@ -291,6 +366,8 @@ async def receive_transcripts(self) -> AsyncIterator[Transcript]: continue # Skip binary frames transcript = self._parse_message(raw_message) if transcript is not None: + if transcript.is_final: + self._record_transcript_cost() yield transcript async def close(self) -> None: diff --git a/libraries/python/getpatter/providers/elevenlabs_convai.py b/libraries/python/getpatter/providers/elevenlabs_convai.py index e09a417e..3c3e43d5 100644 --- a/libraries/python/getpatter/providers/elevenlabs_convai.py +++ b/libraries/python/getpatter/providers/elevenlabs_convai.py @@ -89,6 +89,27 @@ def __init__( # Silence-tracking for synthetic `response_done` emission. self._silence_task: asyncio.Task | None = None self._agent_speaking = False + # Session start time for ``patter.cost.realtime_minutes`` emission on close. + import time as _time + + self._session_start_monotonic: float = _time.monotonic() + + def record_session_end(self) -> None: + """Emit ``patter.cost.realtime_minutes`` for the elapsed session duration.""" + try: + import time as _time + + from getpatter.observability.attributes import record_patter_attrs + + elapsed = _time.monotonic() - self._session_start_monotonic + record_patter_attrs( + { + "patter.cost.realtime_minutes": elapsed / 60.0, + "patter.realtime.provider": "elevenlabs_convai", + } + ) + except Exception: # pragma: no cover — defense in depth + logger.debug("record_session_end failed", exc_info=True) def __repr__(self) -> str: return f"ElevenLabsConvAIAdapter(agent_id={self.agent_id!r}, model_id={self.model_id!r})" @@ -400,6 +421,7 @@ async def receive_events(self): async def close(self) -> None: """Close the connection and cancel the background reader.""" + self.record_session_end() self._running = False self._reset_silence_timer() if self._reader_task and not self._reader_task.done(): diff --git a/libraries/python/getpatter/providers/elevenlabs_tts.py b/libraries/python/getpatter/providers/elevenlabs_tts.py index c9a5c022..0e226680 100644 --- a/libraries/python/getpatter/providers/elevenlabs_tts.py +++ b/libraries/python/getpatter/providers/elevenlabs_tts.py @@ -5,12 +5,15 @@ sensitive use cases prefer :mod:`elevenlabs_ws_tts` (WebSocket). """ +import logging from enum import StrEnum -from typing import AsyncIterator, Optional, Union +from typing import AsyncIterator, ClassVar, Optional, Union import re import httpx from getpatter.providers.base import TTSProvider +logger = logging.getLogger("getpatter.providers.elevenlabs_tts") + # Known stable ElevenLabs voice models (from # https://elevenlabs.io/docs/api-reference/text-to-speech). ``StrEnum`` keeps @@ -152,6 +155,14 @@ class ElevenLabsTTS(TTSProvider): explicitly in that case. """ + # Stable pricing/dashboard key — read by stream-handler/metrics via + # ``getattr(type(agent.tts), "provider_key", None)``. Without this + # the cost calculator falls back to the class name ``ElevenLabsTTS`` + # which does NOT match the pricing table key ``elevenlabs``, + # silently zeroing TTS cost for callers that construct the raw REST + # class directly (exposed at top level as ``ElevenLabsRestTTS``). + provider_key: ClassVar[str] = "elevenlabs" + def __init__( self, api_key: str, @@ -257,8 +268,23 @@ def for_telnyx( chunk_size=chunk_size, ) + def _record_synthesis_cost(self, text: str) -> None: + """Emit ``patter.cost.tts_chars`` for the synthesised text.""" + try: + from getpatter.observability.attributes import record_patter_attrs + + record_patter_attrs( + { + "patter.cost.tts_chars": len(text), + "patter.tts.provider": "elevenlabs", + } + ) + except Exception: # pragma: no cover — defense in depth + logger.debug("_record_synthesis_cost failed", exc_info=True) + async def synthesize(self, text: str) -> AsyncIterator[bytes]: """Stream TTS audio for *text* one chunk at a time.""" + self._record_synthesis_cost(text) body: dict = {"text": text, "model_id": self.model_id} if self.voice_settings: body["voice_settings"] = self.voice_settings diff --git a/libraries/python/getpatter/providers/elevenlabs_ws_tts.py b/libraries/python/getpatter/providers/elevenlabs_ws_tts.py index 6adfe20b..426d523b 100644 --- a/libraries/python/getpatter/providers/elevenlabs_ws_tts.py +++ b/libraries/python/getpatter/providers/elevenlabs_ws_tts.py @@ -40,8 +40,9 @@ import base64 import json import logging +from dataclasses import dataclass from enum import StrEnum -from typing import AsyncGenerator, Optional, Union +from typing import ClassVar, AsyncGenerator, Optional, Union from urllib.parse import quote, urlencode try: @@ -124,8 +125,8 @@ class ElevenLabsPlanError(ElevenLabsTTSError): _PLAN_REQUIRED_MSG = ( "ElevenLabs WS streaming requires a Pro plan or higher (the WS endpoint " "returned `payment_required`). Either upgrade at " - "https://elevenlabs.io/pricing, or use the HTTP `ElevenLabsTTS` class " - "which works on all plans (drop-in API)." + "https://elevenlabs.io/pricing, or use `ElevenLabsRestTTS` for HTTP REST " + "instead which works on all plans (drop-in API)." ) @@ -139,6 +140,21 @@ def _sanitise_log_str(value: object, *, limit: int = 200) -> str: return text.replace("\r", " ").replace("\n", " ").replace("\x00", " ")[:limit] +@dataclass +class ElevenLabsParkedWS: + """Parked WS handle returned by :meth:`ElevenLabsWebSocketTTS.open_parked_connection`. + + ``bos_sent`` records whether the BOS frame (``{"text": " ", ...}``) + has already been written to the wire. The prewarm pipeline sends + the BOS so the upstream worker is selected on the parked + connection; :meth:`synthesize` adopts the WS and SKIPS its own BOS + send to avoid a protocol error. + """ + + ws: "websockets.WebSocketClientProtocol" # noqa: F821 - websockets type + bos_sent: bool = False + + class ElevenLabsWebSocketTTS(TTSProvider): """ElevenLabs streaming TTS via WebSocket (``/stream-input`` endpoint). @@ -147,6 +163,13 @@ class ElevenLabsWebSocketTTS(TTSProvider): via the ``synthesize`` async iterator, identically to the HTTP variant. """ + # Stable provider key for pricing / metrics lookup. Read by + # ``stream_handler`` via ``getattr(type(agent.tts), "provider_key", None)``. + # Without this the cost calculator falls through to the class name + # ("ElevenLabsWebSocketTTS") which doesn't match any pricing.py entry, + # making TTS cost = $0 silently. + provider_key: ClassVar[str] = "elevenlabs_ws" + def __init__( self, api_key: str, @@ -168,7 +191,7 @@ def __init__( if str(model_id).startswith("eleven_v3"): raise ValueError( f"{model_id!r} is not supported by the WebSocket stream-input " - "endpoint — use the HTTP ElevenLabsTTS class instead." + "endpoint — use `ElevenLabsRestTTS` for HTTP REST instead." ) # Stored privately so it is not surfaced via ``vars(tts)`` or accidental # log serialisation. Public read access goes through ``api_key`` below. @@ -197,6 +220,12 @@ def __init__( self.chunk_length_schedule = chunk_length_schedule self.open_timeout = open_timeout self.frame_timeout = frame_timeout + # Single-slot adoption queue. The prewarm pipeline parks one WS + # per outbound call here; the next :meth:`synthesize` call + # consumes it (skipping ``websockets.connect`` and the BOS + # send) instead of opening a fresh socket. The slot is + # consumed exactly once. + self._adopted_connection: Optional[ElevenLabsParkedWS] = None @property def api_key(self) -> str: @@ -325,6 +354,26 @@ def _build_url(self) -> str: params["language_code"] = self.language_code return f"{_WS_BASE}/{quote(self.voice_id)}/stream-input?{urlencode(params)}" + def _build_bos_frame(self) -> dict: + """Build the protocol-required BOS frame sent on every fresh WS. + + The single-space ``{"text": " "}`` keep-alive establishes the + session without committing any synthesis (no ``flush: true``, + no real text). Production ``synthesize()`` and ``warmup()`` + share this exact construction so the upstream worker chooses + the same per-session config in both cases — otherwise the + warm session is on a different worker than the live request, + which defeats the warmup goal. + """ + init: dict = {"text": " "} + if self.voice_settings: + init["voice_settings"] = self.voice_settings + if self.chunk_length_schedule and not self.auto_mode: + init["generation_config"] = { + "chunk_length_schedule": self.chunk_length_schedule, + } + return init + async def synthesize(self, text: str) -> AsyncGenerator[bytes, None]: """Open a WebSocket, stream ``text``, yield raw audio bytes, then close. @@ -348,29 +397,43 @@ async def synthesize(self, text: str) -> AsyncGenerator[bytes, None]: url = self._build_url() headers = {"xi-api-key": self._api_key} - ws = await asyncio.wait_for( - websockets.connect( - url, - additional_headers=headers, - open_timeout=self.open_timeout, - ping_interval=20, - ping_timeout=10, - close_timeout=2, - ), - timeout=self.open_timeout, - ) + # Adopt a parked WS if one is queued AND it is still open. A WS + # that died between park and adopt is discarded silently and a + # fresh socket is opened — preserving the cold backward-compat + # path. + parked = self._adopted_connection + self._adopted_connection = None + bos_already_sent = False + ws = None + if parked is not None and not parked.ws.closed: + ws = parked.ws + bos_already_sent = parked.bos_sent + else: + if parked is not None: + # Parked WS was closed — drop it cleanly. + try: + await parked.ws.close() + except Exception: + pass + ws = await asyncio.wait_for( + websockets.connect( + url, + additional_headers=headers, + open_timeout=self.open_timeout, + ping_interval=20, + ping_timeout=10, + close_timeout=2, + ), + timeout=self.open_timeout, + ) try: # Initial keep-alive packet establishes the session. Per the # ElevenLabs docs the first message must contain a single space # ``" "`` — sending ``""`` would close the socket immediately. - init: dict = {"text": " "} - if self.voice_settings: - init["voice_settings"] = self.voice_settings - if self.chunk_length_schedule and not self.auto_mode: - init["generation_config"] = { - "chunk_length_schedule": self.chunk_length_schedule, - } - await ws.send(json.dumps(init)) + # Skipped on the adopt path when the prewarm pipeline already + # sent it (sending it twice is a protocol error). + if not bos_already_sent: + await ws.send(json.dumps(self._build_bos_frame())) # Send the actual text + flush so ElevenLabs commits the # synthesis without waiting for further chunks. EOS @@ -457,7 +520,135 @@ async def synthesize(self, text: str) -> AsyncGenerator[bytes, None]: except Exception: pass + async def warmup(self) -> None: + """Pre-call WebSocket warmup for the ElevenLabs ``/stream-input`` endpoint. + + Opens the WS (DNS + TLS + auth handshake), sends the EXACT same + BOS frame the production :meth:`synthesize` path sends — including + ``voice_settings`` and (when configured) ``generation_config`` — + so ElevenLabs instantiates the same per-session worker for both + warmup and the live request. If the BOS frames differ, the server + may route warmup and the real call to two different workers, and + the warmed worker is wasted. Idles ~250 ms, then closes. By the + time the first :meth:`synthesize` call lands during the call, the + connection pool has the upstream warm — net wire time saving of + 200-500 ms. + + Billing safety: ElevenLabs bills on synthesised characters delivered + via the ``audio`` frames in the response (per + https://elevenlabs.io/pricing). The keepalive (single-space + ``text``, no ``flush: true``, no real transcript) is documented + as the session-establishment frame and does NOT generate + synthesis. Closing without sending the actual transcript + therefore does not consume billable characters. Best-effort: + failures are logged at DEBUG. + """ + url = self._build_url() + headers = {"xi-api-key": self._api_key} + ws = None + try: + ws = await asyncio.wait_for( + websockets.connect( + url, + additional_headers=headers, + open_timeout=self.open_timeout, + ping_interval=20, + ping_timeout=10, + close_timeout=2, + ), + timeout=self.open_timeout, + ) + # Send the EXACT BOS frame the live synthesize() path sends so + # the server-side worker selection is identical between warmup + # and the live call. + try: + await ws.send(json.dumps(self._build_bos_frame())) + except Exception: + pass + # Brief idle so the provider edge keeps session state warm. + await asyncio.sleep(0.25) + except Exception as exc: # noqa: BLE001 - best-effort + logger.debug("ElevenLabs WS TTS warmup failed (best-effort): %s", exc) + finally: + if ws is not None: + try: + await ws.close() + except Exception: + pass + + async def open_parked_connection(self) -> ElevenLabsParkedWS: + """Open a fresh WS, send the EXACT BOS frame the live + :meth:`synthesize` sends, and return the OPEN socket without + closing it. Used by the prewarm pipeline to park a TTS + connection during the carrier ringing window so the next + :meth:`synthesize` call adopts it via :meth:`adopt_websocket` + and skips ~400-900 ms of TLS + BOS round-trip. + + Billing safety: BOS is the documented session-establishment + frame (single-space ``text``, no ``flush: true``) and does not + generate synthesis. ElevenLabs bills on ``audio`` frames + received from the server, not on BOS bytes sent by the client. + """ + url = self._build_url() + headers = {"xi-api-key": self._api_key} + ws = await asyncio.wait_for( + websockets.connect( + url, + additional_headers=headers, + open_timeout=self.open_timeout, + ping_interval=20, + ping_timeout=10, + close_timeout=2, + ), + timeout=self.open_timeout, + ) + # Send the BOS frame so the upstream worker selection is + # committed BEFORE the live ``synthesize`` adopts this socket. + # Do NOT ``flush: true`` — that would commit synthesis and + # bill characters. + bos_sent = False + try: + await ws.send(json.dumps(self._build_bos_frame())) + bos_sent = True + except Exception: + # BOS send failed — let the consumer re-send. + pass + return ElevenLabsParkedWS(ws=ws, bos_sent=bos_sent) + + def adopt_websocket(self, parked: ElevenLabsParkedWS) -> None: + """Stash a parked WS handle so the next :meth:`synthesize` call + adopts it instead of opening a fresh socket. Caller is + responsible for holding the handle alive until either the live + request consumes it or the call ends (in which case + :meth:`discard_adopted_connection` cleans it up). + """ + prev = self._adopted_connection + self._adopted_connection = parked + if prev is not None and prev is not parked: + try: + asyncio.create_task(prev.ws.close()) + except RuntimeError: + pass + + async def discard_adopted_connection(self) -> None: + """Drop and close any pending parked WS without consuming it. + + Used on call-failure paths so a never-started call does not + leak a TTS WS that ElevenLabs will close after its inactivity + timeout anyway. + """ + parked = self._adopted_connection + self._adopted_connection = None + if parked is not None: + try: + await parked.ws.close() + except Exception: + pass + async def close(self) -> None: - """No-op: connections are per-utterance and closed inline.""" - # No persistent state to clean up — connections are per-utterance. + """No-op: connections are per-utterance and closed inline. + + Drops any orphaned parked WS so we never leak past close. + """ + await self.discard_adopted_connection() return None diff --git a/libraries/python/getpatter/providers/google_llm.py b/libraries/python/getpatter/providers/google_llm.py index e0144611..403a2a0e 100644 --- a/libraries/python/getpatter/providers/google_llm.py +++ b/libraries/python/getpatter/providers/google_llm.py @@ -17,7 +17,7 @@ import logging import os from enum import StrEnum -from typing import Any, AsyncIterator +from typing import ClassVar, Any, AsyncIterator logger = logging.getLogger("getpatter") @@ -72,6 +72,9 @@ class GoogleLLMProvider: max_output_tokens: Optional output token cap. """ + #: Stable pricing/dashboard key — read by stream-handler/metrics. + provider_key: ClassVar[str] = "google" + def __init__( self, api_key: str | None = None, @@ -131,6 +134,33 @@ def __init__( self._temperature = temperature self._max_output_tokens = max_output_tokens + async def warmup(self) -> None: + """Pre-call DNS / TLS warmup for the Gemini API. + + Issues a lightweight ``GET https://generativelanguage.googleapis.com/v1beta/models`` + so DNS, TLS, and HTTP/2 are already up by the time the first + ``generate_content_stream`` call lands. Best-effort: 5 s timeout, + all exceptions swallowed at DEBUG. Skipped on Vertex AI which + requires Application Default Credentials we don't want to mint + for a probe. + """ + try: + api_key = getattr(self._client, "_api_client", None) + # google-genai's Client doesn't expose the API key once it has + # constructed its inner http client; fall back to env var. + key = os.environ.get("GOOGLE_API_KEY") or "" + if not key: + return + import httpx + + async with httpx.AsyncClient(timeout=5.0) as http: + await http.get( + "https://generativelanguage.googleapis.com/v1beta/models", + params={"key": key}, + ) + except Exception as exc: # noqa: BLE001 - best-effort + logger.debug("Google LLM warmup failed (best-effort): %s", exc) + async def stream( self, messages: list[dict], @@ -214,10 +244,16 @@ async def stream( yield {"type": "text", "content": text} if last_usage is not None: + prompt_tokens = getattr(last_usage, "prompt_token_count", 0) or 0 + completion_tokens = getattr(last_usage, "candidates_token_count", 0) or 0 + self._record_completion_cost( + prompt_tokens=prompt_tokens, + completion_tokens=completion_tokens, + ) yield { "type": "usage", - "input_tokens": getattr(last_usage, "prompt_token_count", 0) or 0, - "output_tokens": getattr(last_usage, "candidates_token_count", 0) or 0, + "input_tokens": prompt_tokens, + "output_tokens": completion_tokens, "cache_read_tokens": getattr( last_usage, "cached_content_token_count", 0 ) @@ -226,6 +262,23 @@ async def stream( yield {"type": "done"} + def _record_completion_cost( + self, *, prompt_tokens: int, completion_tokens: int + ) -> None: + """Stamp ``patter.cost.llm_*_tokens`` on the current span.""" + try: + from getpatter.observability.attributes import record_patter_attrs + + record_patter_attrs( + { + "patter.cost.llm_input_tokens": prompt_tokens, + "patter.cost.llm_output_tokens": completion_tokens, + "patter.llm.provider": "google", + } + ) + except Exception: # pragma: no cover — defense in depth + logger.debug("_record_completion_cost failed", exc_info=True) + # --------------------------------------------------------------------------- # Message / tool translation (OpenAI format -> google.genai types) diff --git a/libraries/python/getpatter/providers/groq_llm.py b/libraries/python/getpatter/providers/groq_llm.py index cc62bcea..55504c97 100644 --- a/libraries/python/getpatter/providers/groq_llm.py +++ b/libraries/python/getpatter/providers/groq_llm.py @@ -11,13 +11,17 @@ from __future__ import annotations +import logging import os from enum import StrEnum +from typing import ClassVar from getpatter.services.llm_loop import OpenAILLMProvider __all__ = ["GroqLLMProvider", "GroqModel"] +logger = logging.getLogger("getpatter.providers.groq_llm") + class GroqModel(StrEnum): """Known Groq Chat Completions models. Availability depends on account tier.""" @@ -54,10 +58,13 @@ class GroqLLMProvider(OpenAILLMProvider): :class:`OpenAILLMProvider`. """ + #: Stable pricing/dashboard key — read by stream-handler/metrics. + provider_key: ClassVar[str] = "groq" + def __init__( self, api_key: str | None = None, - model: Union[GroqModel, str] = _DEFAULT_MODEL, + model: GroqModel | str = _DEFAULT_MODEL, base_url: str = _GROQ_BASE_URL, **kwargs, ) -> None: @@ -85,3 +92,20 @@ def __init__( base_url=base_url, default_headers={"User-Agent": self._user_agent}, ) + + def _record_completion_cost( + self, *, prompt_tokens: int, completion_tokens: int + ) -> None: + """Stamp ``patter.cost.llm_*_tokens`` with the Groq provider tag.""" + try: + from getpatter.observability.attributes import record_patter_attrs + + record_patter_attrs( + { + "patter.cost.llm_input_tokens": prompt_tokens, + "patter.cost.llm_output_tokens": completion_tokens, + "patter.llm.provider": "groq", + } + ) + except Exception: # pragma: no cover — defense in depth + logger.debug("_record_completion_cost failed", exc_info=True) diff --git a/libraries/python/getpatter/providers/inworld_tts.py b/libraries/python/getpatter/providers/inworld_tts.py index d0706080..80511c5d 100644 --- a/libraries/python/getpatter/providers/inworld_tts.py +++ b/libraries/python/getpatter/providers/inworld_tts.py @@ -13,18 +13,27 @@ import base64 import json +import logging import os from enum import StrEnum -from typing import Any, AsyncIterator, Optional, Union +from typing import ClassVar, Any, AsyncIterator, Optional, Union from getpatter.providers.base import TTSProvider +logger = logging.getLogger("getpatter.providers.inworld_tts") + try: # pragma: no cover - trivial import guard import aiohttp except ImportError: # pragma: no cover aiohttp = None # type: ignore INWORLD_BASE_URL = "https://api.inworld.ai/tts/v1/voice:stream" +# Voice metadata endpoint used as a billing-safe warmup target. The +# streaming endpoint above is POST-only so HEAD against it returns 405. +# ``GET /tts/v1/voices`` is documented as a free metadata read that +# returns the configured voice catalogue without invoking the synthesis +# pipeline (per https://docs.inworld.ai/). +INWORLD_VOICES_URL = "https://api.inworld.ai/tts/v1/voices" class InworldModel(StrEnum): @@ -63,6 +72,9 @@ class InworldTTS(TTSProvider): yourself before calling the constructor. """ + #: Stable pricing/dashboard key — read by stream-handler/metrics. + provider_key: ClassVar[str] = "inworld" + def __init__( self, auth_token: Optional[str] = None, @@ -172,6 +184,48 @@ async def synthesize(self, text: str) -> AsyncIterator[bytes]: if audio: yield audio + async def warmup(self) -> None: + """Pre-call HTTP warmup for the Inworld TTS API. + + Issues a lightweight ``GET /tts/v1/voices`` against the API host + so DNS + TLS + HTTP/2 connection are already up by the time the + first :meth:`synthesize` POST lands. Best-effort: 5 s timeout, + all exceptions swallowed at DEBUG. + + Earlier revisions issued ``HEAD`` against the streaming endpoint + (``/tts/v1/voice:stream``). That endpoint is POST-only so HEAD + returns ``405 Method Not Allowed`` — the warmup still completed + the TLS handshake but spammed 405 errors into Inworld's audit + logs and into our own logs. Switching to a documented + ``GET /tts/v1/voices`` metadata read is a 2xx-clean equivalent. + + Billing safety: ``GET /tts/v1/voices`` is a free metadata + endpoint (per https://docs.inworld.ai/). It returns the voice + catalogue without invoking the synthesis pipeline. The actual + synthesis is billed only when ``POST /tts/v1/voice:stream`` runs + with a non-empty ``text``. + + Note: Inworld TTS uses the HTTP NDJSON streaming path rather than + a persistent WebSocket — connection warmup is therefore HTTP-based, + not WebSocket pre-handshake. The latency win is smaller (~50-150 ms) + than the WS-based prewarms but still real on cold-start calls. + """ + try: + session = self._ensure_session() + headers = {"Authorization": f"Basic {self.auth_token}"} + # ``GET /tts/v1/voices`` is a billing-safe metadata read that + # returns 2xx (unlike HEAD against the POST-only streaming + # endpoint, which returns 405). + async with session.get( + INWORLD_VOICES_URL, + headers=headers, + timeout=aiohttp.ClientTimeout(total=5), + ) as resp: + # Drain so the underlying connection returns cleanly to the pool. + await resp.read() + except Exception as exc: # noqa: BLE001 - best-effort + logger.debug("Inworld TTS warmup failed (best-effort): %s", exc) + async def close(self) -> None: """Close the underlying session (idempotent).""" if self._session is not None and self._owns_session: diff --git a/libraries/python/getpatter/providers/lmnt_tts.py b/libraries/python/getpatter/providers/lmnt_tts.py index d9decc44..d1058ed1 100644 --- a/libraries/python/getpatter/providers/lmnt_tts.py +++ b/libraries/python/getpatter/providers/lmnt_tts.py @@ -7,12 +7,15 @@ from __future__ import annotations +import logging import os from enum import IntEnum, StrEnum -from typing import Any, AsyncIterator, Optional +from typing import ClassVar, Any, AsyncIterator, Optional from getpatter.providers.base import TTSProvider +logger = logging.getLogger("getpatter.providers.lmnt_tts") + try: # pragma: no cover - trivial import guard import aiohttp except ImportError: # pragma: no cover @@ -58,6 +61,9 @@ class LMNTTTS(TTSProvider): Patter pipeline's standard telephony sample rate. """ + #: Stable pricing/dashboard key — read by stream-handler/metrics. + provider_key: ClassVar[str] = "lmnt" + def __init__( self, api_key: Optional[str] = None, @@ -126,12 +132,27 @@ def _build_payload(self, text: str) -> dict[str, Any]: "top_p": self.top_p, } + def _record_synthesis_cost(self, text: str) -> None: + """Emit ``patter.cost.tts_chars`` for the synthesised text.""" + try: + from getpatter.observability.attributes import record_patter_attrs + + record_patter_attrs( + { + "patter.cost.tts_chars": len(text), + "patter.tts.provider": "lmnt", + } + ) + except Exception: # pragma: no cover — defense in depth + logger.debug("_record_synthesis_cost failed", exc_info=True) + async def synthesize(self, text: str) -> AsyncIterator[bytes]: """Stream audio bytes for ``text``. With the default ``format='raw'`` these are PCM_S16LE chunks at the configured ``sample_rate``. """ + self._record_synthesis_cost(text) session = self._ensure_session() headers = { diff --git a/libraries/python/getpatter/providers/openai_realtime.py b/libraries/python/getpatter/providers/openai_realtime.py index 689e9377..d9462b38 100644 --- a/libraries/python/getpatter/providers/openai_realtime.py +++ b/libraries/python/getpatter/providers/openai_realtime.py @@ -160,6 +160,27 @@ def __init__( # here and drained by ``receive_events`` before reading the socket. self._pending_events: deque[str] = deque() self._receive_task: asyncio.Task | None = None + # Session start time for ``patter.cost.realtime_minutes`` emission on close. + import time as _time + + self._session_start_monotonic: float = _time.monotonic() + + def record_session_end(self) -> None: + """Emit ``patter.cost.realtime_minutes`` for the elapsed session duration.""" + try: + import time as _time + + from getpatter.observability.attributes import record_patter_attrs + + elapsed = _time.monotonic() - self._session_start_monotonic + record_patter_attrs( + { + "patter.cost.realtime_minutes": elapsed / 60.0, + "patter.realtime.provider": "openai_realtime", + } + ) + except Exception: # pragma: no cover — defense in depth + logger.debug("record_session_end failed", exc_info=True) def __repr__(self) -> str: return f"OpenAIRealtimeAdapter(model={self.model!r}, voice={self.voice!r}, audio_format={self.audio_format!r})" @@ -183,6 +204,140 @@ def _build_tool_wire_format(tool: dict) -> dict: wire["strict"] = True return wire + async def warmup(self) -> None: + """Pre-call WebSocket warmup for the OpenAI Realtime endpoint. + + The canonical session-only warm step on the Realtime API: open + the WS, wait for ``session.created``, send a single + ``session.update`` containing the same fields that the production + ``connect()`` path applies (``input_audio_format``, + ``output_audio_format``, ``voice``, ``instructions``, + ``turn_detection``, ``input_audio_transcription``, plus any + opt-in fields populated on the adapter), wait for the matching + ``session.updated`` ack, then close cleanly. This primes the + per-session state on the OpenAI side — DNS + TLS + auth handshake + + initial config exchange — without ever invoking the model. + + Earlier revisions sent ``response.create`` with + ``{"response": {"generate": false}}`` to prime the inference path. + That field is NOT in the OpenAI Realtime API schema; the server + either ignores it (and bills tokens for a real model response) or + rejects the request with ``invalid_request_error``. Both + behaviours are billing-unsafe or a no-op beyond TLS warm. The + ``session.update`` flow is documented and side-effect-free. + + Billing safety: ``session.update`` only mutates session + configuration. It does NOT invoke the model, does NOT consume + any audio buffer, and does NOT trigger token generation, so no + per-token cost is accrued. Best-effort: failures are logged at + DEBUG and never raised. + """ + url = f"{self.OPENAI_REALTIME_URL}?model={self.model}" + ws = None + try: + ws = await asyncio.wait_for( + websockets.connect( + url, + additional_headers={ + "Authorization": f"Bearer {self.api_key}", + "OpenAI-Beta": "realtime=v1", + }, + ping_interval=20, + ping_timeout=20, + ), + timeout=5.0, + ) + # Wait for session.created. + try: + raw = await asyncio.wait_for(ws.recv(), timeout=2.0) + data = json.loads(raw) + if data.get("type") != "session.created": + # Anything else is unexpected but not fatal — we close. + return + except Exception: + return + # Send session.update with the same fields the production + # ``connect()`` path applies, so the upstream session state is + # primed identically to a real call. + try: + await ws.send( + json.dumps( + { + "type": "session.update", + "session": self._build_session_config(), + } + ) + ) + except Exception: + return + # Best-effort: drain frames until we see ``session.updated`` + # (or time out). We don't strictly need the ack to keep the + # warmup correct — the TLS + session prime is already done by + # the time the server processes our update — but waiting for + # it lets us close after a clean handshake instead of mid-frame. + deadline = asyncio.get_event_loop().time() + 1.5 + while True: + remaining = deadline - asyncio.get_event_loop().time() + if remaining <= 0: + break + try: + raw = await asyncio.wait_for(ws.recv(), timeout=remaining) + except Exception: + break + try: + data = json.loads(raw) + except Exception: + continue + if isinstance(data, dict) and data.get("type") == "session.updated": + break + except Exception as exc: # noqa: BLE001 - best-effort + logger.debug("OpenAI Realtime warmup failed (best-effort): %s", exc) + finally: + if ws is not None: + try: + await ws.close() + except Exception: + pass + + def _build_session_config(self) -> dict[str, Any]: + """Build the session.update body shared by ``connect`` / + ``open_parked_connection`` / ``warmup`` so all three paths + prime the upstream session identically. + """ + session_config: dict[str, Any] = { + "input_audio_format": self.audio_format, + "output_audio_format": self.audio_format, + "voice": self.voice, + "instructions": self.instructions + or f"You are a helpful voice assistant. Respond in {self.language}. Be concise and natural.", + "turn_detection": { + "type": self.vad_type, + "threshold": 0.5, + "prefix_padding_ms": 300, + "silence_duration_ms": self.silence_duration_ms, + }, + "input_audio_transcription": { + "model": self.input_audio_transcription_model, + }, + } + if self.temperature is not None: + session_config["temperature"] = self.temperature + if self.max_response_output_tokens is not None: + session_config["max_response_output_tokens"] = ( + self.max_response_output_tokens + ) + if self.modalities is not None: + session_config["modalities"] = self.modalities + if self.tool_choice is not None: + session_config["tool_choice"] = self.tool_choice + if self.tools: + session_config["tools"] = [ + self._build_tool_wire_format(t) for t in self.tools + ] + if self.reasoning_effort is not None: + session_config["reasoning"] = {"effort": self.reasoning_effort} + return session_config + async def connect(self) -> None: """Connect to OpenAI Realtime API and wait for ``session.updated`` ack.""" url = f"{self.OPENAI_REALTIME_URL}?model={self.model}" @@ -207,44 +362,11 @@ async def connect(self) -> None: if data.get("type") != "session.created": raise RuntimeError(f"Expected session.created, got {data.get('type')}") - # Configure session audio format (g711_ulaw for Twilio, pcm16 for Telnyx) - session_config: dict[str, Any] = { - "input_audio_format": self.audio_format, - "output_audio_format": self.audio_format, - "voice": self.voice, - "instructions": self.instructions - or f"You are a helpful voice assistant. Respond in {self.language}. Be concise and natural.", - "turn_detection": { - "type": self.vad_type, - "threshold": 0.5, - "prefix_padding_ms": 300, - "silence_duration_ms": self.silence_duration_ms, - }, - "input_audio_transcription": { - "model": self.input_audio_transcription_model, - }, - } - if self.temperature is not None: - session_config["temperature"] = self.temperature - if self.max_response_output_tokens is not None: - session_config["max_response_output_tokens"] = ( - self.max_response_output_tokens - ) - if self.modalities is not None: - session_config["modalities"] = self.modalities - if self.tool_choice is not None: - session_config["tool_choice"] = self.tool_choice - if self.tools: - session_config["tools"] = [ - self._build_tool_wire_format(t) for t in self.tools - ] - if self.reasoning_effort is not None: - session_config["reasoning"] = {"effort": self.reasoning_effort} await self._ws.send( json.dumps( { "type": "session.update", - "session": session_config, + "session": self._build_session_config(), } ) ) @@ -259,6 +381,87 @@ async def connect(self) -> None: self._running = False raise + async def open_parked_connection(self): # type: ignore[no-untyped-def] + """Open a fresh Realtime WS, exchange ``session.created`` / + ``session.update`` / ``session.updated`` so the upstream session + is fully primed, and return the OPEN socket WITHOUT taking it + on ``self._ws``. + + Used by the prewarm pipeline to park a Realtime connection + during ringing; the live consumer adopts it via + :meth:`adopt_websocket`. + + Bounded by 8 s. Raises on timeout / handshake failure — the + prewarm pipeline treats any error as a cache miss and the call + falls through to the cold ``connect()`` path. + + Billing safety: ``session.update`` does not invoke the model. + No tokens are billed. + """ + url = f"{self.OPENAI_REALTIME_URL}?model={self.model}" + ws = await asyncio.wait_for( + websockets.connect( + url, + additional_headers={ + "Authorization": f"Bearer {self.api_key}", + "OpenAI-Beta": "realtime=v1", + }, + ping_interval=20, + ping_timeout=20, + ), + timeout=8.0, + ) + try: + response = await asyncio.wait_for(ws.recv(), timeout=2.0) + data = json.loads(response) + if data.get("type") != "session.created": + raise RuntimeError(f"Expected session.created, got {data.get('type')}") + await ws.send( + json.dumps( + { + "type": "session.update", + "session": self._build_session_config(), + } + ) + ) + # Drain frames until session.updated (or 1.5 s timeout). + deadline = asyncio.get_event_loop().time() + 1.5 + while True: + remaining = deadline - asyncio.get_event_loop().time() + if remaining <= 0: + break + try: + raw = await asyncio.wait_for(ws.recv(), timeout=remaining) + except Exception: + break + try: + data = json.loads(raw) + except Exception: + continue + if isinstance(data, dict) and data.get("type") == "session.updated": + break + except Exception: + try: + await ws.close() + except Exception: + pass + raise + return ws + + def adopt_websocket(self, ws) -> None: # type: ignore[no-untyped-def] + """Adopt a pre-opened, already-``session.updated`` Realtime WS + produced by the prewarm pipeline. Skips the fresh + ``websockets.connect`` + ``session.created`` / + ``session.update`` round-trip — saves ~250-450 ms on first turn. + + Caller MUST verify the WS is still alive (``not ws.closed``) + before calling and MUST have already received + ``session.updated`` on the parked socket. If the parked WS died + between park and adopt, fall back to :meth:`connect`. + """ + self._ws = ws + self._running = True + async def _await_session_updated(self) -> None: """Read a single post-``session.update`` message and return. @@ -539,6 +742,7 @@ async def send_function_result(self, call_id: str, result: str) -> None: async def close(self) -> None: """Close the connection and cancel any in-flight receive task.""" + self.record_session_end() self._running = False task = self._receive_task if task is not None and not task.done(): diff --git a/libraries/python/getpatter/providers/openai_transcribe_stt.py b/libraries/python/getpatter/providers/openai_transcribe_stt.py index e0e95d77..fa0b9954 100644 --- a/libraries/python/getpatter/providers/openai_transcribe_stt.py +++ b/libraries/python/getpatter/providers/openai_transcribe_stt.py @@ -13,7 +13,7 @@ from __future__ import annotations -from typing import Literal +from typing import ClassVar, Literal from getpatter.providers.whisper_stt import WhisperSTT @@ -38,6 +38,9 @@ class OpenAITranscribeSTT(WhisperSTT): response_format: ``"json"`` (default) or ``"verbose_json"``. """ + #: Stable pricing/dashboard key — read by stream-handler/metrics. + provider_key: ClassVar[str] = "openai_transcribe" + def __init__( self, api_key: str, @@ -57,3 +60,8 @@ def __init__( model=model, response_format=response_format, ) + # Observability: ``_record_transcript_cost`` is inherited from + # ``WhisperSTT`` and emits ``patter.stt.provider="whisper"``. This + # is intentional — gpt-4o-transcribe shares the Whisper cost table, + # so a single tag keeps the dashboard's per-provider rollup + # consistent across the OpenAI transcription family. diff --git a/libraries/python/getpatter/providers/openai_tts.py b/libraries/python/getpatter/providers/openai_tts.py index 76e85353..319b52f6 100644 --- a/libraries/python/getpatter/providers/openai_tts.py +++ b/libraries/python/getpatter/providers/openai_tts.py @@ -5,11 +5,14 @@ forward bytes without an additional resample stage. """ +import logging from enum import StrEnum -from typing import AsyncIterator, Union +from typing import ClassVar, AsyncIterator, Union import httpx +logger = logging.getLogger("getpatter.providers.openai_tts") + try: # Python ≤ 3.12 ships ``audioop``; on 3.13+ the ``audioop-lts`` PyPI # package exposes the same C API (pinned in our pyproject). @@ -65,6 +68,9 @@ class OpenAITTSResponseFormat(StrEnum): class OpenAITTS(TTSProvider): """OpenAI HTTP TTS provider with built-in 24k→target-rate resampling.""" + #: Stable pricing/dashboard key — read by stream-handler/metrics. + provider_key: ClassVar[str] = "openai_tts" + def __init__( self, api_key: str, @@ -95,8 +101,23 @@ def __init__( def __repr__(self) -> str: return f"OpenAITTS(model={self.model!r}, voice={self.voice!r})" + def _record_synthesis_cost(self, text: str) -> None: + """Emit ``patter.cost.tts_chars`` for the synthesised text.""" + try: + from getpatter.observability.attributes import record_patter_attrs + + record_patter_attrs( + { + "patter.cost.tts_chars": len(text), + "patter.tts.provider": "openai_tts", + } + ) + except Exception: # pragma: no cover — defense in depth + logger.debug("_record_synthesis_cost failed", exc_info=True) + async def synthesize(self, text: str) -> AsyncIterator[bytes]: """Stream PCM audio for *text* resampled to ``target_sample_rate``.""" + self._record_synthesis_cost(text) if audioop is None: # Without ``audioop`` / ``audioop-lts`` we would emit 24 kHz # audio that the telephony pipeline transcodes as 16 kHz — diff --git a/libraries/python/getpatter/providers/rime_tts.py b/libraries/python/getpatter/providers/rime_tts.py index 78f1f272..5187af5d 100644 --- a/libraries/python/getpatter/providers/rime_tts.py +++ b/libraries/python/getpatter/providers/rime_tts.py @@ -7,12 +7,15 @@ from __future__ import annotations +import logging import os from enum import StrEnum -from typing import Any, AsyncIterator, Optional +from typing import ClassVar, Any, AsyncIterator, Optional from getpatter.providers.base import TTSProvider +logger = logging.getLogger("getpatter.providers.rime_tts") + try: # pragma: no cover - trivial import guard import aiohttp except ImportError: # pragma: no cover @@ -65,6 +68,9 @@ class RimeTTS(TTSProvider): PCM_S16LE at the configured ``sample_rate`` (default 16000 Hz). """ + #: Stable pricing/dashboard key — read by stream-handler/metrics. + provider_key: ClassVar[str] = "rime" + def __init__( self, api_key: Optional[str] = None, @@ -169,8 +175,23 @@ def _build_payload(self, text: str) -> dict[str, Any]: return payload + def _record_synthesis_cost(self, text: str) -> None: + """Emit ``patter.cost.tts_chars`` for the synthesised text.""" + try: + from getpatter.observability.attributes import record_patter_attrs + + record_patter_attrs( + { + "patter.cost.tts_chars": len(text), + "patter.tts.provider": "rime", + } + ) + except Exception: # pragma: no cover — defense in depth + logger.debug("_record_synthesis_cost failed", exc_info=True) + async def synthesize(self, text: str) -> AsyncIterator[bytes]: """Stream raw PCM_S16LE bytes for ``text`` over HTTP.""" + self._record_synthesis_cost(text) session = self._ensure_session() headers = { diff --git a/libraries/python/getpatter/providers/silero_onnx.py b/libraries/python/getpatter/providers/silero_onnx.py index 49fe01a5..57d92ac9 100644 --- a/libraries/python/getpatter/providers/silero_onnx.py +++ b/libraries/python/getpatter/providers/silero_onnx.py @@ -157,3 +157,8 @@ def __call__(self, x: np.ndarray) -> float: out, self._rnn_state = self._sess.run(None, ort_inputs) self._context = self._input_buffer[:, -self._context_size :] # type: ignore return out.item() # type: ignore + + def reset(self) -> None: + """Reset the RNN hidden state + rolling context to a fresh inference.""" + self._context = np.zeros((1, self._context_size), dtype=np.float32) + self._rnn_state = np.zeros((2, 1, 128), dtype=np.float32) diff --git a/libraries/python/getpatter/providers/silero_vad.py b/libraries/python/getpatter/providers/silero_vad.py index d10d8d6d..a0b813bf 100644 --- a/libraries/python/getpatter/providers/silero_vad.py +++ b/libraries/python/getpatter/providers/silero_vad.py @@ -122,7 +122,12 @@ def load( cls, *, min_speech_duration: float = 0.25, - min_silence_duration: float = 0.1, + # Bumped 0.1 → 0.4s after round 10f confirmed VAD speech_end fired on + # natural inter-sentence pauses < 250ms, causing double-talk dispatch. + # 0.4s is the industry-standard default for telephony agents — enough + # to bridge natural inter-sentence pauses without delaying single- + # sentence turns excessively. + min_silence_duration: float = 0.4, prefix_padding_duration: float = 0.03, activation_threshold: float = 0.5, sample_rate: Union[ @@ -361,6 +366,29 @@ async def close(self) -> None: self._onnx_session = None # type: ignore[assignment] self._model = None # type: ignore[assignment] + def reset(self) -> None: + """Reset all per-utterance state so the next ``process_frame`` starts + from a clean SILENCE state. + + Called by the stream handler between agent turns to prevent a "stuck + SPEECH" condition where PSTN echo / loopback kept the detector's + probability above ``deactivation_threshold`` for the entire agent + turn. Without this reset the next user utterance would never + trigger a SILENCE→SPEECH transition and barge-in would feel + "one-shot" (works once, then never again until the call ends). + + Safe to call any time including on a closed instance (no-op). + """ + if self._closed: + return + self._pending = np.zeros(0, dtype=np.float32) + self._pub_speaking = False + self._speech_threshold_duration = 0.0 + self._silence_threshold_duration = 0.0 + self._exp_filter.reset() + if self._model is not None: + self._model.reset() + def _run_inference(model: OnnxModel, window: np.ndarray) -> float: """Blocking inference entrypoint executed in an executor thread.""" diff --git a/libraries/python/getpatter/providers/soniox_stt.py b/libraries/python/getpatter/providers/soniox_stt.py index 23067ef3..f616365b 100644 --- a/libraries/python/getpatter/providers/soniox_stt.py +++ b/libraries/python/getpatter/providers/soniox_stt.py @@ -18,7 +18,7 @@ import json import logging from enum import IntEnum, StrEnum -from typing import Any, AsyncIterator +from typing import ClassVar, Any, AsyncIterator import aiohttp @@ -153,6 +153,9 @@ class SonioxSTT(STTProvider): base_url: Override the Soniox WebSocket URL (used by tests). """ + #: Stable pricing/dashboard key — read by stream-handler/metrics. + provider_key: ClassVar[str] = "soniox" + def __init__( self, api_key: str, @@ -189,6 +192,29 @@ def __init__( self._owns_session: bool = False self._ws: aiohttp.ClientWebSocketResponse | None = None self._keepalive_task: asyncio.Task[None] | None = None + # Soniox always sends pcm_s16le (see ``_build_config``), so encoding + # is fixed; expose for observability cost computation. + self.encoding: str = "linear16" + self._audio_bytes_sent: int = 0 + + def _record_transcript_cost(self) -> None: + """Emit ``patter.cost.stt_seconds`` for buffered audio.""" + try: + from getpatter.observability.attributes import record_patter_attrs + + bytes_per_sample = 1 if self.encoding == "mulaw" else 2 + seconds = self._audio_bytes_sent / float( + self.sample_rate * bytes_per_sample + ) + record_patter_attrs( + { + "patter.cost.stt_seconds": seconds, + "patter.stt.provider": "soniox", + } + ) + self._audio_bytes_sent = 0 + except Exception: # pragma: no cover — defense in depth + logger.debug("_record_transcript_cost failed", exc_info=True) def __repr__(self) -> str: return ( @@ -285,6 +311,7 @@ async def send_audio(self, audio_chunk: bytes) -> None: raise RuntimeError("SonioxSTT is not connected. Call connect() first.") if not audio_chunk: return + self._audio_bytes_sent += len(audio_chunk) await self._ws.send_bytes(audio_chunk) # ------------------------------------------------------------------ @@ -341,6 +368,7 @@ async def receive_transcripts(self) -> AsyncIterator[Transcript]: if _is_end_token(token): # Endpoint detected — flush the accumulated final text. if final_acc.text: + self._record_transcript_cost() yield Transcript( text=final_acc.text.strip(), is_final=True, @@ -372,6 +400,7 @@ async def receive_transcripts(self) -> AsyncIterator[Transcript]: if content.get("finished"): # Final flush on server-side finish. if final_acc.text: + self._record_transcript_cost() yield Transcript( text=final_acc.text.strip(), is_final=True, diff --git a/libraries/python/getpatter/providers/speechmatics_stt.py b/libraries/python/getpatter/providers/speechmatics_stt.py index 7c99377f..125f7e1c 100644 --- a/libraries/python/getpatter/providers/speechmatics_stt.py +++ b/libraries/python/getpatter/providers/speechmatics_stt.py @@ -21,7 +21,7 @@ import asyncio import logging from enum import Enum, IntEnum, StrEnum -from typing import Any, AsyncIterator, Union +from typing import ClassVar, Any, AsyncIterator, Union from getpatter.providers.base import STTProvider, Transcript @@ -109,6 +109,9 @@ class SpeechmaticsSTT(STTProvider): output_locale: Optional output locale (e.g. ``"en-GB"``). """ + #: Stable pricing/dashboard key — read by stream-handler/metrics. + provider_key: ClassVar[str] = "speechmatics" + # Sentinel used to shut down the receive loop. _STOP = object() @@ -172,6 +175,29 @@ def __init__( self._client: Any | None = None self._queue: asyncio.Queue[Any] = asyncio.Queue() + # Speechmatics always streams pcm_s16le (see ``_build_config``); + # expose encoding for observability cost computation. + self.encoding: str = "linear16" + self._audio_bytes_sent: int = 0 + + def _record_transcript_cost(self) -> None: + """Emit ``patter.cost.stt_seconds`` for buffered audio.""" + try: + from getpatter.observability.attributes import record_patter_attrs + + bytes_per_sample = 1 if self.encoding == "mulaw" else 2 + seconds = self._audio_bytes_sent / float( + self.sample_rate * bytes_per_sample + ) + record_patter_attrs( + { + "patter.cost.stt_seconds": seconds, + "patter.stt.provider": "speechmatics", + } + ) + self._audio_bytes_sent = 0 + except Exception: # pragma: no cover — defense in depth + logger.debug("_record_transcript_cost failed", exc_info=True) def __repr__(self) -> str: return ( @@ -254,6 +280,7 @@ async def send_audio(self, audio_chunk: bytes) -> None: ) if not audio_chunk: return + self._audio_bytes_sent += len(audio_chunk) await self._client.send_audio(audio_chunk) # ------------------------------------------------------------------ @@ -317,6 +344,8 @@ async def receive_transcripts(self) -> AsyncIterator[Transcript]: logger.exception("SpeechmaticsSTT handler error: %s", exc) continue if transcript is not None: + if transcript.is_final: + self._record_transcript_cost() yield transcript # ------------------------------------------------------------------ diff --git a/libraries/python/getpatter/providers/telnyx_adapter.py b/libraries/python/getpatter/providers/telnyx_adapter.py index e9f9d941..fb844002 100644 --- a/libraries/python/getpatter/providers/telnyx_adapter.py +++ b/libraries/python/getpatter/providers/telnyx_adapter.py @@ -5,9 +5,14 @@ attached separately from ``call.answered`` (see :mod:`telephony.telnyx`). """ +import logging + import httpx + from getpatter.providers.base import TelephonyProvider +logger = logging.getLogger("getpatter.providers.telnyx_adapter") + TELNYX_API_BASE = "https://api.telnyx.com/v2" @@ -157,6 +162,25 @@ async def end_call(self, call_id: str, *, command_id: str | None = None) -> None f"/calls/{_quote(call_id, safe='')}/actions/hangup", json=body ) + def record_call_end_cost(self, *, duration_seconds: float, direction: str) -> None: + """Emit ``patter.cost.telephony_minutes`` on the active span. + + Called by the embedded server's bridge cleanup once the call's + wall-clock duration is known. + """ + try: + from getpatter.observability.attributes import record_patter_attrs + + record_patter_attrs( + { + "patter.cost.telephony_minutes": duration_seconds / 60.0, + "patter.telephony": "telnyx", + "patter.direction": direction, + } + ) + except Exception: # pragma: no cover — defense in depth + logger.debug("record_call_end_cost failed", exc_info=True) + async def close(self) -> None: """Close the underlying HTTP client.""" await self._client.aclose() diff --git a/libraries/python/getpatter/providers/telnyx_stt.py b/libraries/python/getpatter/providers/telnyx_stt.py index 572ba651..217a2af5 100644 --- a/libraries/python/getpatter/providers/telnyx_stt.py +++ b/libraries/python/getpatter/providers/telnyx_stt.py @@ -12,7 +12,7 @@ import json import struct from enum import IntEnum, StrEnum -from typing import AsyncIterator, Literal +from typing import ClassVar, AsyncIterator, Literal import aiohttp @@ -92,6 +92,9 @@ class TelnyxSTT(STTProvider): session is created and closed with :meth:`close`. """ + #: Stable pricing/dashboard key — read by stream-handler/metrics. + provider_key: ClassVar[str] = "telnyx_stt" + def __init__( self, api_key: str, diff --git a/libraries/python/getpatter/providers/telnyx_tts.py b/libraries/python/getpatter/providers/telnyx_tts.py index 2d3d33f8..483f438d 100644 --- a/libraries/python/getpatter/providers/telnyx_tts.py +++ b/libraries/python/getpatter/providers/telnyx_tts.py @@ -12,7 +12,7 @@ import base64 import json from enum import IntEnum, StrEnum -from typing import AsyncIterator +from typing import ClassVar, AsyncIterator import aiohttp @@ -55,6 +55,9 @@ class TelnyxTTS(TTSProvider): session is created and closed with :meth:`close`. """ + #: Stable pricing/dashboard key — read by stream-handler/metrics. + provider_key: ClassVar[str] = "telnyx_tts" + def __init__( self, api_key: str, diff --git a/libraries/python/getpatter/providers/twilio_adapter.py b/libraries/python/getpatter/providers/twilio_adapter.py index d7b24d26..0ed71026 100644 --- a/libraries/python/getpatter/providers/twilio_adapter.py +++ b/libraries/python/getpatter/providers/twilio_adapter.py @@ -5,11 +5,14 @@ """ import asyncio +import logging from functools import partial from twilio.rest import Client as TwilioClient from twilio.twiml.voice_response import VoiceResponse, Connect from getpatter.providers.base import TelephonyProvider +logger = logging.getLogger("getpatter.providers.twilio_adapter") + class TwilioAdapter(TelephonyProvider): """:class:`TelephonyProvider` implementation backed by the Twilio REST API.""" @@ -75,6 +78,25 @@ async def end_call(self, call_id: str) -> None: self._twilio_client.calls(call_id).update, status="completed" ) + def record_call_end_cost(self, *, duration_seconds: float, direction: str) -> None: + """Emit ``patter.cost.telephony_minutes`` on the active span. + + Called by the embedded server's bridge cleanup once the call's + wall-clock duration is known. + """ + try: + from getpatter.observability.attributes import record_patter_attrs + + record_patter_attrs( + { + "patter.cost.telephony_minutes": duration_seconds / 60.0, + "patter.telephony": "twilio", + "patter.direction": direction, + } + ) + except Exception: # pragma: no cover — defense in depth + logger.debug("record_call_end_cost failed", exc_info=True) + @staticmethod def generate_stream_twiml(stream_url: str) -> str: """Return TwiML that connects the inbound call to the given media stream URL.""" diff --git a/libraries/python/getpatter/providers/whisper_stt.py b/libraries/python/getpatter/providers/whisper_stt.py index 397f5b1d..520b6553 100644 --- a/libraries/python/getpatter/providers/whisper_stt.py +++ b/libraries/python/getpatter/providers/whisper_stt.py @@ -7,7 +7,7 @@ import logging import wave from enum import StrEnum -from typing import AsyncIterator, Literal +from typing import ClassVar, AsyncIterator, Literal import httpx @@ -74,6 +74,9 @@ class WhisperSTT(STTProvider): surface per-segment timestamps / confidence. """ + #: Stable pricing/dashboard key — read by stream-handler/metrics. + provider_key: ClassVar[str] = "whisper" + def __init__( self, api_key: str, @@ -92,7 +95,12 @@ def __init__( self.language = language self.model = model self.response_format = response_format + # Whisper buffers PCM and uploads as 16 kHz / 16-bit / mono WAV; + # expose these so observability cost computation has explicit values. + self.sample_rate = 16000 + self.encoding = "linear16" self._buffer = bytearray() + self._audio_bytes_sent: int = 0 self._transcript_queue: asyncio.Queue[Transcript] = asyncio.Queue() self._running = False self._pending: set[asyncio.Task] = set() @@ -101,6 +109,25 @@ def __init__( timeout=10.0, ) + def _record_transcript_cost(self) -> None: + """Emit ``patter.cost.stt_seconds`` for buffered audio.""" + try: + from getpatter.observability.attributes import record_patter_attrs + + bytes_per_sample = 1 if self.encoding == "mulaw" else 2 + seconds = self._audio_bytes_sent / float( + self.sample_rate * bytes_per_sample + ) + record_patter_attrs( + { + "patter.cost.stt_seconds": seconds, + "patter.stt.provider": "whisper", + } + ) + self._audio_bytes_sent = 0 + except Exception: # pragma: no cover — defense in depth + logger.debug("_record_transcript_cost failed", exc_info=True) + @classmethod def for_twilio( cls, @@ -123,6 +150,7 @@ async def connect(self) -> None: async def send_audio(self, audio_chunk: bytes) -> None: """Buffer incoming PCM audio and transcribe when the buffer is full.""" + self._audio_bytes_sent += len(audio_chunk) self._buffer.extend(audio_chunk) if len(self._buffer) >= BUFFER_SIZE_BYTES: buf = bytes(self._buffer) @@ -171,9 +199,14 @@ async def receive_transcripts(self) -> AsyncIterator[Transcript]: """Async generator that yields transcripts as they arrive.""" while self._running: try: - yield await asyncio.wait_for(self._transcript_queue.get(), timeout=0.1) + transcript = await asyncio.wait_for( + self._transcript_queue.get(), timeout=0.1 + ) except asyncio.TimeoutError: continue + if transcript.is_final: + self._record_transcript_cost() + yield transcript async def close(self) -> None: """Flush remaining buffer and close the HTTP client. diff --git a/libraries/python/getpatter/server.py b/libraries/python/getpatter/server.py index b5c7de45..e02ac74c 100644 --- a/libraries/python/getpatter/server.py +++ b/libraries/python/getpatter/server.py @@ -237,6 +237,28 @@ def __init__( # recent outbound call. Cleared after firing once per call so a result # for a previous call cannot leak into a new caller's callback. self.on_machine_detection = None + # Pre-warm first-message audio accessor wired by ``Patter.serve()``. + # The per-call StreamHandler invokes this with its ``call_id`` at the + # start of the firstMessage emit; a non-None return is sent verbatim + # in place of running TTS again. ``None`` means "no prewarm cache for + # this call — fall back to live synthesis". Default is a no-op so + # callers that instantiate ``EmbeddedServer`` directly (tests) work + # without further setup. + self.pop_prewarm_audio = lambda _cid: None + # Pre-warmed provider WebSocket accessor wired by + # ``Patter.serve()``. The per-call StreamHandler invokes this + # with its ``call_id`` at pipeline init; a defined return hands + # off pre-opened STT / TTS / Realtime sockets so the live first + # turn skips the cold-handshake. ``None`` means "no parked + # sockets — fall back to fresh ``connect()``". + self.pop_prewarmed_connections = lambda _cid: None + # Prewarm waste recorder wired by ``Patter.serve()``. Invoked from + # the Twilio status callback (no-answer / busy / failed / canceled) + # and the Telnyx call.hangup / AMD-machine handlers so the cache + # entry is evicted when the call terminates before the media stream + # starts. Default is a no-op so direct ``EmbeddedServer`` callers + # (tests) work without further setup. See FIX #91. + self.record_prewarm_waste = lambda _cid: None self._telnyx_sig_warning_logged = False self._metrics_store = None # Opt-in per-call filesystem logging. Path is resolved by @@ -305,11 +327,28 @@ async def _on_call_start(data): except Exception: pass if call_logger.enabled: + # For outbound calls the bridge has no caller/callee in the + # WS query string (TwiML for outbound is inline + # ```` with no tags), + # so ``data["caller"]`` / ``data["callee"]`` are empty here. + # The active record in the store was populated by + # ``record_call_initiated`` at dial time and holds the correct + # numbers — pull them from there before persisting + # metadata.json. Without this fallback every outbound call's + # metadata.json on disk has ``caller=""`` / ``callee=""``. + call_id_str = data.get("call_id", "") or "" + data_caller = data.get("caller", "") or "" + data_callee = data.get("callee", "") or "" + active_record = ( + store.get_active(call_id_str) if (store and call_id_str) else None + ) or {} + resolved_caller = data_caller or active_record.get("caller", "") or "" + resolved_callee = data_callee or active_record.get("callee", "") or "" await alog_call_start( call_logger, - data.get("call_id", ""), - caller=data.get("caller", "") or "", - callee=data.get("callee", "") or "", + call_id_str, + caller=resolved_caller, + callee=resolved_callee, telephony_provider=data.get("telephony_provider", "") or "", provider_mode=getattr(agent, "provider", "") or "", agent=_agent_snapshot(), @@ -334,20 +373,42 @@ async def _on_call_end(data): from dataclasses import asdict, is_dataclass metrics_obj = data.get("metrics") - duration = getattr(metrics_obj, "duration_seconds", None) if metrics_obj else None + duration = ( + getattr(metrics_obj, "duration_seconds", None) + if metrics_obj + else None + ) cost_obj = getattr(metrics_obj, "cost", None) if metrics_obj else None cost_dict = asdict(cost_obj) if is_dataclass(cost_obj) else None latency_dict = None + avg = getattr(metrics_obj, "latency_avg", None) if metrics_obj else None p95 = getattr(metrics_obj, "latency_p95", None) if metrics_obj else None p50 = getattr(metrics_obj, "latency_p50", None) if metrics_obj else None p99 = getattr(metrics_obj, "latency_p99", None) if metrics_obj else None - if p50 is not None or p95 is not None or p99 is not None: + if ( + avg is not None + or p50 is not None + or p95 is not None + or p99 is not None + ): + # Persist full LatencyBreakdown per percentile so the + # dashboard hydrate path can render stt/llm/tts breakdown + # for historical calls. Keep flat ``p50_ms/p95_ms/p99_ms`` + # for backward compat with consumers that only read totals. latency_dict = { "p50_ms": getattr(p50, "total_ms", None) if p50 else None, "p95_ms": getattr(p95, "total_ms", None) if p95 else None, "p99_ms": getattr(p99, "total_ms", None) if p99 else None, + "avg": asdict(avg) if is_dataclass(avg) else None, + "p50": asdict(p50) if is_dataclass(p50) else None, + "p95": asdict(p95) if is_dataclass(p95) else None, + "p99": asdict(p99) if is_dataclass(p99) else None, } - turns_count = len(getattr(metrics_obj, "turns", []) or []) if metrics_obj else None + turns_count = ( + len(getattr(metrics_obj, "turns", []) or []) + if metrics_obj + else None + ) await alog_call_end( call_logger, data.get("call_id", ""), @@ -437,7 +498,9 @@ async def _read_and_validate_twilio_form(request: Request): returns a 503 Response — safety-first posture requires an explicit opt-out to accept unsigned webhooks. """ - if not self.config.twilio_token and getattr(self.config, "require_signature", True): + if not self.config.twilio_token and getattr( + self.config, "require_signature", True + ): logger.error( "Twilio webhook rejected: twilio_token not configured and " "require_signature=True. Set twilio_token, or explicitly " @@ -458,7 +521,9 @@ async def _read_and_validate_twilio_form(request: Request): "Install with: pip install 'getpatter[local]' or " "`pip install twilio`." ) - return Response(status_code=503, content="Signature validator unavailable") + return Response( + status_code=503, content="Signature validator unavailable" + ) form_data = await request.form() validator = RequestValidator(self.config.twilio_token) # Use request.url verbatim when it carries .path / .query @@ -493,7 +558,9 @@ async def twilio_voice(request: Request): # masked. Same for `To` / `Called`. caller = form_data.get("From", "") or form_data.get("Caller", "") callee = form_data.get("To", "") or form_data.get("Called", "") - twiml = twilio_webhook_handler(call_sid, caller, callee, self.config.webhook_url) + twiml = twilio_webhook_handler( + call_sid, caller, callee, self.config.webhook_url + ) return Response(content=twiml, media_type="text/xml") # Twilio posts here for every status transition of a call @@ -524,6 +591,21 @@ async def twilio_status_callback(request: Request): except ValueError: pass self._metrics_store.update_call_status(call_sid, call_status, **extra) + # FIX #91 — when the call terminates before the media stream + # starts (no-answer / busy / failed / canceled), the prewarm + # cache entry would otherwise leak until ``end_call`` runs. + # Evict it here so the WARN fires once and the bytes are + # released regardless of whether the user calls ``end_call``. + if call_sid and call_status in ( + "no-answer", + "busy", + "failed", + "canceled", + ): + try: + self.record_prewarm_waste(call_sid) + except Exception as exc: # noqa: BLE001 - defensive + logger.debug("record_prewarm_waste raised: %s", exc) return Response(content="", status_code=204) @app.post("/webhooks/twilio/recording") @@ -572,6 +654,17 @@ async def twilio_amd_callback(request: Request): except Exception as exc: logger.warning("on_machine_detection callback threw: %s", exc) + # FIX #91 — when AMD classifies as machine, the agent's first + # message will not be played (we drop voicemail or hang up), so + # the prewarmed greeting is never consumed. Evict the cache + # entry once so the WARN fires regardless of whether + # ``voicemail_message`` is configured. + if answered_by in ("machine_end_beep", "machine_end_silence") and call_sid: + try: + self.record_prewarm_waste(call_sid) + except Exception as exc: # noqa: BLE001 - defensive + logger.debug("record_prewarm_waste raised: %s", exc) + if ( answered_by in ("machine_end_beep", "machine_end_silence") and self.voicemail_message @@ -584,7 +677,9 @@ async def twilio_amd_callback(request: Request): ) if not _validate_twilio_sid(call_sid, "CA"): - logger.warning("AMD callback: invalid CallSid format %r, ignoring", call_sid) + logger.warning( + "AMD callback: invalid CallSid format %r, ignoring", call_sid + ) return Response(content="", status_code=204) import httpx as _httpx @@ -630,6 +725,8 @@ async def twilio_stream_handler(websocket: WebSocket, call_id: str): await twilio_stream_bridge( websocket=websocket, agent=self.agent, + pop_prewarm_audio=self.pop_prewarm_audio, + pop_prewarmed_connections=self.pop_prewarmed_connections, openai_key=self.config.openai_key, on_call_start=_start, on_call_end=_end, @@ -666,7 +763,9 @@ async def telnyx_voice(request: Request): if not _validate_telnyx_signature( raw_body, signature, timestamp, telnyx_public_key ): - logger.warning("Telnyx webhook rejected: invalid or missing Ed25519 signature") + logger.warning( + "Telnyx webhook rejected: invalid or missing Ed25519 signature" + ) return Response(status_code=403, content="Invalid signature") elif require_sig: logger.error( @@ -690,7 +789,9 @@ async def telnyx_voice(request: Request): if not isinstance(body.get("data"), dict) or not isinstance( body.get("data", {}).get("payload"), dict ): - logger.warning("Telnyx webhook rejected: missing data.payload structure.") + logger.warning( + "Telnyx webhook rejected: missing data.payload structure." + ) return Response(status_code=400, content="Invalid webhook structure") data = body["data"] event_type = data.get("event_type", "") @@ -805,7 +906,9 @@ async def telnyx_voice(request: Request): if asyncio.iscoroutine(cb_ret): await cb_ret except Exception as exc: - logger.warning("on_machine_detection callback threw: %s", exc) + logger.warning( + "on_machine_detection callback threw: %s", exc + ) if self.voicemail_message: from getpatter.telephony.telnyx import handle_amd_result @@ -815,6 +918,37 @@ async def telnyx_voice(request: Request): voicemail_message=self.voicemail_message, telnyx_key=api_key, ) + # FIX #91 — when AMD classifies as machine the agent's + # first message is replaced by ``voicemail_message`` (or + # the call simply ends), so the prewarmed greeting is + # never consumed. Evict it so the WARN fires once. + if call_control_id and amd_result in ( + "machine", + "machine_detected", + ): + try: + self.record_prewarm_waste(call_control_id) + except Exception as exc: # noqa: BLE001 - defensive + logger.debug("record_prewarm_waste raised: %s", exc) + elif event_type == "call.hangup": + # FIX #91 — Telnyx fires ``call.hangup`` as the final + # status notification. ``hangup_cause`` distinguishes + # carrier outcomes (``call_rejected`` / ``busy`` / + # ``no_answer`` / ``timeout`` / ``normal_clearing`` / + # ``user_busy`` / …). When the call never reached the + # media stream the prewarm cache leaks unless we + # evict it here. + hangup_cause = str(payload.get("hangup_cause", "")) + logger.info( + "Telnyx call.hangup for %s (cause=%s)", + sanitize_log_value(call_control_id), + sanitize_log_value(hangup_cause), + ) + if call_control_id: + try: + self.record_prewarm_waste(call_control_id) + except Exception as exc: # noqa: BLE001 - defensive + logger.debug("record_prewarm_waste raised: %s", exc) else: logger.debug("Telnyx event ignored: %s", event_type) except Exception as exc: @@ -840,6 +974,8 @@ async def telnyx_stream_handler(websocket: WebSocket, call_id: str): await telnyx_stream_bridge( websocket=websocket, agent=self.agent, + pop_prewarm_audio=self.pop_prewarm_audio, + pop_prewarmed_connections=self.pop_prewarmed_connections, openai_key=self.config.openai_key, on_call_start=_start, on_call_end=_end, @@ -922,7 +1058,9 @@ async def start(self, port: int = 8000) -> None: "Twilio webhook enforcement ACTIVE but twilio_token is empty " "— webhooks will 503. Set require_signature=False for local dev." ) - if provider == "telnyx" and not getattr(self.config, "telnyx_public_key", ""): + if provider == "telnyx" and not getattr( + self.config, "telnyx_public_key", "" + ): logger.warning( "Telnyx webhook enforcement ACTIVE but telnyx_public_key is empty " "— webhooks will 503. Set require_signature=False for local dev." diff --git a/libraries/python/getpatter/services/barge_in_strategies.py b/libraries/python/getpatter/services/barge_in_strategies.py new file mode 100644 index 00000000..8b44cd97 --- /dev/null +++ b/libraries/python/getpatter/services/barge_in_strategies.py @@ -0,0 +1,163 @@ +"""Barge-in confirmation strategies. + +When a caller starts speaking while the agent's TTS is in flight, the SDK has +to decide whether the speech is a real interruption or just a brief +backchannel ("uh-huh", "okay") / room noise / cough. The default behaviour +is to treat any VAD speech_start as a confirmed barge-in and cancel the +agent immediately. That is fine for clean inputs but produces frequent +false positives on PSTN: the agent gets cut mid-sentence by background +chatter, breath, or filler words and never recovers the conversational +thread. + +Each :class:`BargeInStrategy` is consulted on every STT transcript while a +barge-in is *pending* (VAD fired, but the agent has not yet been cancelled). +The first strategy that returns ``True`` confirms the barge-in; if none do +within the configured timeout the pending state is dropped and the agent +resumes streaming TTS as if nothing happened. With an empty +``barge_in_strategies`` tuple the SDK falls back to the legacy +"interrupt immediately on VAD" path, so adding strategies is a strict +opt-in. +""" + +from __future__ import annotations + +import logging +from typing import Protocol, runtime_checkable + +logger = logging.getLogger("getpatter") + + +@runtime_checkable +class BargeInStrategy(Protocol): + """Decides whether a pending barge-in should be confirmed. + + Implementations are async-friendly and stateless across calls — every + call gets its own copies of the strategies (deep-copied at call setup) + so per-call accumulators are safe. + + Subclasses MUST implement ``evaluate``. ``reset`` is optional — the + default no-op suits stateless strategies. + """ + + async def evaluate( + self, + *, + transcript: str, + is_interim: bool, + agent_speaking: bool, + ) -> bool: + """Return ``True`` when this strategy considers the user's speech a + confirmed barge-in. + + Args: + transcript: The latest STT output text (interim or final). + is_interim: ``True`` for interim partials, ``False`` for final + transcripts. Strategies may choose to ignore one bucket. + agent_speaking: Whether the agent's TTS is currently in flight. + Strategies typically apply a stricter rule while the agent + is talking and a permissive rule otherwise. + + Returns: + ``True`` to confirm the barge-in (cancels agent TTS + flushes + inbound buffer + dispatches the user transcript). ``False`` to + keep waiting — the strategy will be consulted again on the next + transcript event. + """ + ... + + async def reset(self) -> None: + """Drop any per-turn accumulator state. + + Called when the agent finishes speaking naturally (no barge-in) + and when a pending barge-in times out without confirmation. + Default implementation is a no-op. + """ + ... + + +class MinWordsStrategy: + """Confirm barge-in only after the caller has spoken ``min_words`` words. + + This filters short backchannels, single-word utterances, and stray + transcription fragments that VAD picked up but were not real + interruptions. While the agent is silent the strategy permits any + speech to count (one word is enough), so the first user turn is + not delayed. + + Args: + min_words: Minimum word count required while the agent is + speaking. Reasonable values are 2-5; 3 is a good starting + point for production phone agents. Must be ``>= 1``. + use_interim: When ``True`` (default), interim STT partials are + evaluated as soon as they arrive. Set to ``False`` to wait + for finals only — slower but free of partial-word noise on + jittery STT providers. + """ + + def __init__(self, *, min_words: int, use_interim: bool = True) -> None: + if min_words < 1: + raise ValueError(f"min_words must be >= 1 (got {min_words})") + self._min_words = min_words + self._use_interim = use_interim + + async def evaluate( + self, + *, + transcript: str, + is_interim: bool, + agent_speaking: bool, + ) -> bool: + if is_interim and not self._use_interim: + return False + threshold = self._min_words if agent_speaking else 1 + word_count = len(transcript.split()) + return word_count >= threshold + + async def reset(self) -> None: + return None + + +async def evaluate_strategies( + strategies: tuple[BargeInStrategy, ...], + *, + transcript: str, + is_interim: bool, + agent_speaking: bool, +) -> bool: + """Short-circuit-OR composition: first strategy that confirms wins. + + Returns ``False`` for an empty tuple so callers can use the empty + default to mean "no opt-in confirmation, fall back to legacy + interrupt-on-VAD". + """ + if not strategies: + return False + text = transcript or "" + for strategy in strategies: + try: + if await strategy.evaluate( + transcript=text, + is_interim=is_interim, + agent_speaking=agent_speaking, + ): + return True + except Exception as exc: # pragma: no cover - defensive + logger.warning( + "BargeInStrategy %s raised; treating as 'do not confirm': %s", + type(strategy).__name__, + exc, + ) + return False + + +async def reset_strategies(strategies: tuple[BargeInStrategy, ...]) -> None: + """Call ``reset()`` on every strategy, swallowing per-strategy errors.""" + for strategy in strategies: + try: + await strategy.reset() + except Exception as exc: # pragma: no cover - defensive + logger.debug( + "BargeInStrategy %s.reset() raised: %s", + type(strategy).__name__, + exc, + ) diff --git a/libraries/python/getpatter/services/llm_loop.py b/libraries/python/getpatter/services/llm_loop.py index f810d4d8..6478ee61 100644 --- a/libraries/python/getpatter/services/llm_loop.py +++ b/libraries/python/getpatter/services/llm_loop.py @@ -338,6 +338,17 @@ async def stream( """ ... # pragma: no cover + # Optional: ``async def warmup(self) -> None`` — best-effort pre-call + # DNS / TLS / HTTP-keepalive warmup invoked by ``Patter.call`` when the + # agent has ``prewarm=True``. Concrete providers (OpenAI, Anthropic, + # Google, Cerebras, Groq) define this method to issue a lightweight + # HTTPS GET to their inference endpoint so by the time the first + # ``stream()`` call lands, the connection pool already has a warm + # socket. Detected via duck-typed ``getattr(provider, "warmup", None)`` + # in the client so plain mocks / older providers without ``warmup`` + # still satisfy this protocol — kept off the Protocol surface to + # preserve backward-compat with ``runtime_checkable.isinstance``. + # --------------------------------------------------------------------------- # Built-in OpenAI provider @@ -374,6 +385,9 @@ class OpenAILLMProvider: ``f"getpatter/{__version__}"`` for upstream attribution. """ + #: Stable pricing/dashboard key — read by stream-handler/metrics. + provider_key: ClassVar[str] = "openai" + def __init__( self, api_key: str, @@ -423,6 +437,44 @@ def __init__( self._temperature = temperature self._max_tokens = max_tokens + async def warmup(self) -> None: + """Pre-call DNS / TLS / HTTP-keepalive warmup. + + Issues a lightweight ``GET /models`` so the underlying + ``httpx.AsyncClient`` (owned by the OpenAI SDK) opens a socket and + completes the TLS handshake during the carrier ringing window. + By the time the first ``chat.completions.create`` call lands, the + connection pool has a warm socket and the first chunk arrives a + DNS+TLS round-trip earlier (~150-400 ms saved on cold start). + + Note: an HTTPS GET warms DNS + TLS + connection pool but does NOT + warm the inference path itself; for true inference warmup a real + low-token request is needed, left as a follow-up. STT / TTS providers ship concrete + WebSocket-based prewarms (Cartesia / Deepgram / AssemblyAI for + STT; ElevenLabs WS for TTS) which save 200-500 ms each — those + dominate the cold-start latency budget. + + Best-effort: timeouts and any other exception are swallowed at + DEBUG. Mirrors the warmup contract documented on the + :class:`LLMProvider` protocol. + """ + try: + base_url = str(getattr(self._client, "base_url", "") or "").rstrip("/") + if not base_url: + return + import httpx + + async with httpx.AsyncClient(timeout=5.0) as http: + await http.get( + f"{base_url}/models", + headers={ + "Authorization": f"Bearer {self._client.api_key}", + "User-Agent": self._user_agent, + }, + ) + except Exception as exc: # noqa: BLE001 - best-effort + logger.debug("LLM warmup failed (best-effort): %s", exc) + def _build_completion_kwargs( self, messages: list[dict], @@ -537,13 +589,41 @@ async def stream( # the full input rate (mirrors libraries/typescript/src/llm-loop.ts:296-305). prompt_tokens = getattr(last_usage, "prompt_tokens", 0) or 0 uncached_input = max(0, prompt_tokens - cache_read) + completion_tokens = getattr(last_usage, "completion_tokens", 0) or 0 + self._record_completion_cost( + prompt_tokens=prompt_tokens, + completion_tokens=completion_tokens, + ) yield { "type": "usage", "input_tokens": uncached_input, - "output_tokens": getattr(last_usage, "completion_tokens", 0) or 0, + "output_tokens": completion_tokens, "cache_read_tokens": cache_read, } + def _record_completion_cost( + self, *, prompt_tokens: int, completion_tokens: int + ) -> None: + """Stamp ``patter.cost.llm_*_tokens`` on the current span. + + Subclasses (Groq, Cerebras) inherit this — the ``patter.llm.provider`` + tag is overridden in the subclass to identify the upstream vendor. + Provider-specific subclasses with a different response shape (Anthropic, + Google) override this directly. + """ + try: + from getpatter.observability.attributes import record_patter_attrs + + record_patter_attrs( + { + "patter.cost.llm_input_tokens": prompt_tokens, + "patter.cost.llm_output_tokens": completion_tokens, + "patter.llm.provider": "openai", + } + ) + except Exception: # pragma: no cover — defense in depth + logger.debug("_record_completion_cost failed", exc_info=True) + # --------------------------------------------------------------------------- # LLM loop @@ -632,6 +712,12 @@ def __init__( else: self._provider_name = "openai" + # Diagnostics for the char/4 fallback billing path (see _run_completion). + # Counted per-LLMLoop instance (i.e. per call). Surfaced only via logs + # — keeps record_llm_usage's public signature unchanged. + self._usage_missing_count = 0 + self._logged_usage_fallback = False + # Build OpenAI-format tool definitions (without handler/webhook_url) self._openai_tools: list[dict] | None = None if tools: @@ -727,6 +813,7 @@ async def run( tool_calls_accumulated: dict[int, dict] = {} text_parts: list[str] = [] has_tool_calls = False + usage_chunk_received = False # Open a span around the provider streaming call. Kept as an # explicit __enter__/__exit__ (rather than ``with``) because we @@ -766,6 +853,7 @@ async def run( yield content elif chunk_type == "usage": + usage_chunk_received = True if self._metrics is not None: self._metrics.record_llm_usage( provider=self._provider_name, @@ -808,6 +896,58 @@ async def run( finally: _span_cm.__exit__(None, None, None) + # Fallback billing: some providers (Cerebras streaming has been + # observed to do this on certain chunk-shape variants) don't + # emit a ``usage`` chunk even with ``stream_options={"include_usage": + # True}``. Without this fallback the LLM cost silently shows ~0 + # for the whole call. char/4 is the canonical OpenAI-tokenizer + # rough estimate; conservative-upward is preferable to silent zero. + if not usage_chunk_received and self._metrics is not None: + input_chars = sum( + len(m.get("content", "") or "") + for m in messages + if isinstance(m, dict) + ) + output_chars = sum(len(p) for p in text_parts) + estimated_input = max(1, input_chars // 4) + estimated_output = max(1, output_chars // 4) + self._metrics.record_llm_usage( + provider=self._provider_name, + model=self._model, + input_tokens=estimated_input, + output_tokens=estimated_output, + ) + self._usage_missing_count += 1 + # First fallback in this call → INFO so the operator sees it once. + # Subsequent iterations only DEBUG to avoid spamming logs on + # long tool-loop turns where every iteration is char/4-billed. + if not self._logged_usage_fallback: + self._logged_usage_fallback = True + logger.info( + "llm_usage_fallback provider=%s model=%s input_chars=%d " + "output_chars=%d est_input_tokens=%d est_output_tokens=%d", + self._provider_name, + self._model, + input_chars, + output_chars, + estimated_input, + estimated_output, + ) + else: + logger.debug( + "llm_usage_fallback provider=%s model=%s iteration=%d " + "input_chars=%d output_chars=%d est_input_tokens=%d " + "est_output_tokens=%d total_missing=%d", + self._provider_name, + self._model, + iteration, + input_chars, + output_chars, + estimated_input, + estimated_output, + self._usage_missing_count, + ) + # If no tool calls, we're done if not has_tool_calls: if has_after_llm_response: diff --git a/libraries/python/getpatter/services/metrics.py b/libraries/python/getpatter/services/metrics.py index 6a86d3bb..1637816b 100644 --- a/libraries/python/getpatter/services/metrics.py +++ b/libraries/python/getpatter/services/metrics.py @@ -134,6 +134,28 @@ def __init__( self._num_backchannels: int = 0 self._overlap_started_at: float | None = None + # --- Barge-in anchor hygiene --- + # Monotonic timestamp of the most recent barge-in detection. Used by + # ``_compute_turn_latency`` to gate ``endpoint_ms`` / ``stt_ms`` + # emission on turns that started within 100 ms of the last barge-in + # — those turns have unreliable VAD/STT anchors and would otherwise + # pollute the p95 distribution with synthetic 6+ second spikes. + self._last_bargein_at: float | None = None + # Counter of turns where ``record_stt_complete`` fired but no VAD + # ``speech_end`` had stamped ``_endpoint_signal_at`` first. Lets us + # detect environments where PSTN packet loss is dropping VAD stops + # (the common cause of missing endpoint signals). + self._endpoint_signal_missing_count: int = 0 + + # --- Initial-TTFB guard reset on barge-in / new turn --- + # When ``report_only_initial_ttfb=True`` is set, the EventBus TTFB + # emission is suppressed after the first event per call. After a + # barge-in we want the NEXT turn's TTFB to fire again so the + # dashboard can show post-barge-in recovery latency. Set by + # ``record_llm_first_token`` / ``record_tts_first_byte`` and reset + # by ``_reset_turn_state``. + self._initial_ttfb_emitted: bool = False + # ---- EventBus attachment ---- def attach_event_bus(self, bus: "EventBus") -> None: @@ -203,6 +225,39 @@ def start_turn_if_idle(self) -> None: if self._turn_start is None: self.start_turn() + def anchor_user_speech_start(self) -> None: + """Anchor the current turn at a legitimate VAD ``speech_start``. + + Industry-standard pattern: every VAD ``speech_start`` event that fires while the + agent is NOT in the suppressed warmup window is treated as the + canonical start of the user's utterance for the current turn. Reset + ``_turn_start`` and ``_endpoint_signal_at`` so that: + + * If a phantom ``speech_start`` previously stamped ``_turn_start`` + during agent TTS (e.g. echo / loopback / cough that was + ``_can_barge_in()``-suppressed but not phantom-suppressed at the + metrics layer in earlier 0.6.1 builds), the legitimate user-speech + start re-anchors the timer at the right wall-clock moment. + * The stale ``_endpoint_signal_at`` from any rejected barge-in / + dropped final transcript on the SAME turn (before LLM dispatch) + is cleared, so the next ``record_vad_stop`` stamps fresh. + + No-ops once the turn is committed (``_turn_committed_mono`` set): + a new VAD ``speech_start`` after commit is the START of the next + turn's barge-in, handled by ``record_turn_interrupted`` instead. + """ + if self._turn_committed_mono is not None: + return + self._turn_start = time.monotonic() + self._endpoint_signal_at = None + self._vad_stopped_at = None + self._stt_final_at = None + # Reset all the other "first signal of this turn" stamps too so a + # mid-turn re-anchor produces a clean per-turn breakdown. + self._stt_complete = None + self._llm_first_token = None + self._llm_ttfb_emitted = False + def record_llm_first_token(self) -> None: """Mark when the first LLM output token arrives (TTFT). @@ -244,6 +299,14 @@ def record_llm_first_sentence(self) -> None: def record_stt_complete(self, text: str, audio_seconds: float = 0.0) -> None: """Mark STT as complete for the current turn.""" self._stt_complete = time.monotonic() + # Don't fake _endpoint_signal_at from _stt_complete — that creates + # dishonest endpoint_ms == stt_ms outliers (6818 ms p95 spikes observed + # on PSTN when VAD speech_end is dropped). Honest None is better than a + # synthetic value for SLO/alerting. The counter lets us know if this + # happens often (PSTN packet loss dropping VAD stops is the common + # cause). + if self._endpoint_signal_at is None: + self._endpoint_signal_missing_count += 1 self._turn_user_text = text self._turn_stt_audio_seconds = audio_seconds self._total_stt_audio_seconds += audio_seconds @@ -319,7 +382,12 @@ def record_bargein_detected(self, ts: float | None = None) -> None: Pairs with :meth:`record_tts_stopped` to compute ``bargein_ms``. """ - self._bargein_detected_at = ts if ts is not None else time.monotonic() + t = ts if ts is not None else time.monotonic() + self._bargein_detected_at = t + # Stamp _last_bargein_at on the same monotonic clock as _turn_start so + # the post-barge-in anchor-gating in _compute_turn_latency stays valid + # (see the comment there for rationale). + self._last_bargein_at = t def record_tts_stopped(self, ts: float | None = None) -> None: """Mark the moment TTS playback was actually halted after a barge-in. @@ -374,7 +442,25 @@ def record_turn_interrupted(self) -> TurnMetrics | None: timestamp=time.time(), ) self._turns.append(turn) + # Emit the turn record BEFORE reset so subscribers see the + # interrupted turn with its anchors still intact. Parity with + # record_turn_complete(). + if self._event_bus is not None: + self._event_bus.emit( + "turn_ended", + {"call_id": self.call_id, "turn": turn}, + ) + self._event_bus.emit( + "metrics_collected", + {"call_id": self.call_id, "turn": turn}, + ) self._reset_turn_state() + # Extra paranoia: explicitly null out anchors that have caused leaks + # into subsequent turns when a barge-in is in flight. _reset_turn_state + # already clears them, but keep this belt-and-braces line so future + # refactors that touch _reset_turn_state don't silently regress us. + self._turn_committed_mono = None + self._endpoint_signal_at = None return turn # ---- EOUMetrics ---- @@ -427,6 +513,12 @@ def _emit_eou_metrics(self) -> None: Guards against emitting garbage data when only a subset of timestamps has been recorded (e.g. VAD skipped in non-local mode). + + Field semantics (must match TS ``emitEouMetrics``): + ``end_of_utterance_delay`` = stt_final − vad_stopped + ``transcription_delay`` = turn_committed − vad_stopped + Both deltas are emitted in **milliseconds** (raw wall-clock + ``time.time()`` timestamps are in seconds, hence the ``* 1000``). """ if ( self._vad_stopped_at is None @@ -442,9 +534,13 @@ def _emit_eou_metrics(self) -> None: from getpatter.observability.metric_types import EOUMetrics eou = EOUMetrics( - end_of_utterance_delay=self._turn_committed_at - self._vad_stopped_at, - transcription_delay=self._stt_final_at - self._vad_stopped_at, - on_user_turn_completed_delay=self._on_user_turn_completed_delay_ms / 1000.0, + end_of_utterance_delay=max( + 0.0, (self._stt_final_at - self._vad_stopped_at) * 1000.0 + ), + transcription_delay=max( + 0.0, (self._turn_committed_at - self._vad_stopped_at) * 1000.0 + ), + on_user_turn_completed_delay=self._on_user_turn_completed_delay_ms, ) self._event_bus.emit("eou_metrics", eou) @@ -599,6 +695,9 @@ def end_call(self) -> CallMetrics: tts_provider=self.tts_provider, llm_provider=self.llm_provider, telephony_provider=self.telephony_provider, + stt_model=self.stt_model, + tts_model=self.tts_model, + llm_model=self._llm_model, ) if self._event_bus is not None: @@ -628,6 +727,12 @@ def _reset_turn_state(self) -> None: self._bargein_stopped_at = None self._turn_user_text = "" self._turn_stt_audio_seconds = 0.0 + # Reset initial-TTFB latch so EventBus TTFB emission re-fires on the + # new turn. Without this, with report_only_initial_ttfb=True we lose + # the TTFB metric on the first turn after a barge-in / new turn. + self._initial_ttfb_emitted = False + self._llm_ttfb_emitted = False + self._tts_ttfb_emitted = False def _compute_turn_latency(self) -> LatencyBreakdown: """Compute latency breakdown for the current turn.""" @@ -636,9 +741,38 @@ def _compute_turn_latency(self) -> LatencyBreakdown: llm_ttft_ms = 0.0 tts_ms = 0.0 total_ms = 0.0 + user_speech_duration_ms: float | None = None + + # Post-barge-in turns have unreliable anchors. Drop endpoint_ms / + # stt_ms to avoid polluting the p95 distribution with synthetic + # spikes. The honest None makes the metric usable for SLO/alerting; + # without this gate, a single barge-in produces 6+ second p95 + # outliers. + post_bargein = ( + self._last_bargein_at is not None + and self._turn_start is not None + and abs(self._turn_start - self._last_bargein_at) <= 0.1 # 100 ms + ) - if self._turn_start is not None and self._stt_complete is not None: - stt_ms = (self._stt_complete - self._turn_start) * 1000 + # ``stt_ms`` measures pure STT finalization: end-of-speech (VAD stop + # or STT speech_final) → final transcript delivery. This is the + # engineering metric reported as "STT latency" by the industry. When + # the endpoint signal is unavailable (degraded provider, batch STT) + # fall back to the legacy turn_start anchor so the field is never + # spuriously zero. + if self._stt_complete is not None: + anchor = ( + self._endpoint_signal_at + if self._endpoint_signal_at is not None + else self._turn_start + ) + if anchor is not None: + stt_ms = max(0.0, (self._stt_complete - anchor) * 1000) + + if self._turn_start is not None and self._endpoint_signal_at is not None: + user_speech_duration_ms = max( + 0.0, (self._endpoint_signal_at - self._turn_start) * 1000 + ) if self._stt_complete is not None and self._llm_complete is not None: llm_ms = (self._llm_complete - self._stt_complete) * 1000 @@ -718,6 +852,15 @@ def _compute_turn_latency(self) -> LatencyBreakdown: if endpoint_ms is not None and llm_ttft_ms is not None and tts_ms > 0: agent_response_ms = round(endpoint_ms + llm_ttft_ms + tts_ms, 1) + # Post-barge-in anchor hygiene: when the current turn began within + # 100 ms of the last detected barge-in, the VAD/STT anchors are + # unreliable. Drop the polluted endpoint_ms and stt_ms so percentile + # aggregations ignore them (stt_ms = 0 is excluded by the non-zero + # filter in _compute_percentile_latency). + if post_bargein: + stt_ms = 0.0 + endpoint_ms = None + # Note: in Realtime mode OpenAI handles STT+LLM+TTS as a single opaque # pipeline, so stt_ms / llm_ms / tts_ms stay 0 and only total_ms is # meaningful. Dashboards should prefer total_ms as the end-to-end @@ -728,6 +871,11 @@ def _compute_turn_latency(self) -> LatencyBreakdown: llm_ms=round(llm_ms, 1), tts_ms=round(tts_ms, 1), total_ms=round(total_ms, 1), + user_speech_duration_ms=( + round(user_speech_duration_ms, 1) + if user_speech_duration_ms is not None + else None + ), llm_ttft_ms=round(llm_ttft_ms, 1) if llm_ttft_ms else None, llm_total_ms=round(llm_total_ms, 1) if llm_total_ms is not None else None, endpoint_ms=round(endpoint_ms, 1) if endpoint_ms is not None else None, @@ -850,6 +998,8 @@ def _opt_avg(attr: str) -> float | None: endpoint_ms=_opt_avg("endpoint_ms"), bargein_ms=_opt_avg("bargein_ms"), tts_total_ms=_opt_avg("tts_total_ms"), + user_speech_duration_ms=_opt_avg("user_speech_duration_ms"), + agent_response_ms=_opt_avg("agent_response_ms"), ) def _compute_percentile_latency(self, p: float) -> LatencyBreakdown: @@ -926,4 +1076,6 @@ def _opt_pct(attr: str) -> float | None: endpoint_ms=_opt_pct("endpoint_ms"), bargein_ms=_opt_pct("bargein_ms"), tts_total_ms=_opt_pct("tts_total_ms"), + user_speech_duration_ms=_opt_pct("user_speech_duration_ms"), + agent_response_ms=_opt_pct("agent_response_ms"), ) diff --git a/libraries/python/getpatter/services/pipeline_hooks.py b/libraries/python/getpatter/services/pipeline_hooks.py index 1f269474..37dcd39e 100644 --- a/libraries/python/getpatter/services/pipeline_hooks.py +++ b/libraries/python/getpatter/services/pipeline_hooks.py @@ -24,7 +24,7 @@ import inspect import logging import warnings -from typing import Any, Awaitable, Callable, Protocol, TYPE_CHECKING, runtime_checkable +from typing import Any, Callable, Protocol, TYPE_CHECKING, runtime_checkable if TYPE_CHECKING: from getpatter.models import HookContext, PipelineHooks @@ -228,9 +228,7 @@ async def run_after_llm_sentence( return None # empty = drop return result - async def run_after_llm_response( - self, text: str, ctx: "HookContext" - ) -> str: + async def run_after_llm_response(self, text: str, ctx: "HookContext") -> str: """Tier 3 — per-response rewrite (~500 ms – 2 s). Triggered after the LLM stream completes. Caller is responsible for @@ -281,9 +279,7 @@ def has_after_llm(self) -> bool: """ return self.has_after_llm_response() - async def run_before_synthesize( - self, text: str, ctx: "HookContext" - ) -> str | None: + async def run_before_synthesize(self, text: str, ctx: "HookContext") -> str | None: """Run beforeSynthesize hook. Returns None if hook vetoes TTS.""" hook = self._hooks.before_synthesize if self._hooks else None if hook is None: @@ -306,3 +302,21 @@ async def run_after_synthesize( except Exception: logger.exception("Pipeline hook after_synthesize threw") return audio + + def record_turn_latency(self, *, ttfb_ms: float, turn_ms: float) -> None: + """Emit ``patter.latency.{ttfb_ms,turn_ms}`` for the just-completed turn. + + Stamped on the active span via :func:`record_patter_attrs`. No-op + when OTel is missing or no ``patter_call_scope`` is active. + """ + try: + from getpatter.observability.attributes import record_patter_attrs + + record_patter_attrs( + { + "patter.latency.ttfb_ms": ttfb_ms, + "patter.latency.turn_ms": turn_ms, + } + ) + except Exception: # pragma: no cover — defense in depth + logger.debug("record_turn_latency failed", exc_info=True) diff --git a/libraries/python/getpatter/stream_handler.py b/libraries/python/getpatter/stream_handler.py index a1d49cce..a75c42dc 100644 --- a/libraries/python/getpatter/stream_handler.py +++ b/libraries/python/getpatter/stream_handler.py @@ -53,7 +53,7 @@ # only — used on PSTN where AEC is a no-op so there is no warmup to # protect, and a long gate just suppresses real-user barge-in. MIN_AGENT_SPEAKING_S_BEFORE_BARGE_IN_AEC = 1.0 -MIN_AGENT_SPEAKING_S_BEFORE_BARGE_IN_NO_AEC = 0.25 +MIN_AGENT_SPEAKING_S_BEFORE_BARGE_IN_NO_AEC = 0.1 # Backwards-compat alias used by tests; matches AEC variant. MIN_AGENT_SPEAKING_S_BEFORE_BARGE_IN = MIN_AGENT_SPEAKING_S_BEFORE_BARGE_IN_AEC @@ -363,6 +363,37 @@ def get_guardrail_replacement(agent, guard_name: str) -> str: return "I'm sorry, I can't respond to that." +async def _safe_close_parked_handle(handle: object) -> None: + """Best-effort async close of a parked provider handle that the + StreamHandler chose NOT to adopt (cache miss, parked WS already + dead, unknown shape, etc.). + + Handles all flavours used by the SDK: + - tuple ``(session, ws)`` from Cartesia STT. + - object with ``.ws`` attribute (e.g. ``ElevenLabsParkedWS``). + - bare WebSocket / ``WebSocketClientProtocol``. + """ + try: + if isinstance(handle, tuple) and len(handle) == 2: + session, ws = handle + try: + await ws.close() + except Exception: + pass + try: + await session.close() + except Exception: + pass + return + ws = getattr(handle, "ws", None) + if ws is not None: + await ws.close() + return + await handle.close() # type: ignore[attr-defined] + except Exception: + pass + + # --------------------------------------------------------------------------- # Base StreamHandler # --------------------------------------------------------------------------- @@ -422,6 +453,11 @@ def __init__( # WebSocket / HTTP connections. Parity with TS field. self._mcp_manager: Any = None + # Set by Patter._attach_span_exporter via attach_span_exporter; "uut" by default. + # Read once at handler start; later changes via the same Patter instance + # will not retroactively affect this handler's spans. + self._patter_side: str = getattr(self, "_patter_side", "uut") + # Create one EventBus per handler instance and wire it to metrics. from getpatter.observability.event_bus import EventBus as _EventBus @@ -638,6 +674,27 @@ async def _emit_turn_metrics(self, turn, *, call_id: str | None = None) -> None: appending transcript entries / storing the turn; only the user-facing callback is centralised here for parity with TS ``emitTurnMetrics``. """ + # Stamp patter.latency.{ttfb_ms,turn_ms} on the active span before the + # user callback runs. ``ttfb_ms`` maps to ``total_ms`` (turn_start → + # first TTS audio byte — the user-perceptible "time to first byte" + # for the response). ``turn_ms`` maps to ``tts_total_ms`` when set + # (LLM-first-token → last TTS byte) and falls back to ``total_ms``. + if turn is not None and getattr(turn, "latency", None) is not None: + try: + from getpatter.services.pipeline_hooks import PipelineHookExecutor + + ttfb_ms = float(turn.latency.total_ms or 0.0) + turn_ms = float( + turn.latency.tts_total_ms + if turn.latency.tts_total_ms is not None + else (turn.latency.total_ms or 0.0) + ) + PipelineHookExecutor(hooks=None).record_turn_latency( + ttfb_ms=ttfb_ms, turn_ms=turn_ms + ) + except Exception: # pragma: no cover — observability must never break calls + logger.debug("record_turn_latency failed", exc_info=True) + if not self.on_metrics or turn is None or self.metrics is None: return await self.on_metrics( @@ -1587,6 +1644,8 @@ def __init__( on_metrics=None, conversation_history: deque | None = None, transcript_entries: deque | None = None, + pop_prewarm_audio=None, + pop_prewarmed_connections=None, ) -> None: super().__init__( agent=agent, @@ -1602,6 +1661,16 @@ def __init__( conversation_history=conversation_history, transcript_entries=transcript_entries, ) + # Optional accessor returning pre-rendered first-message audio for + # ``call_id``. Wired by ``Patter.serve()`` when the parent client + # has ``agent.prewarm_first_message=True``. ``None`` (default) means + # "no prewarm — always run live TTS". + self._pop_prewarm_audio = pop_prewarm_audio + # Optional accessor returning pre-opened, fully-handshaked + # provider WebSockets for ``call_id``. Wired by ``Patter.serve()``. + # Returning ``None`` means "no parked sockets — fall back to + # fresh ``connect()``". + self._pop_prewarmed_connections = pop_prewarmed_connections self._openai_key = openai_key self._deepgram_key = deepgram_key self._elevenlabs_key = elevenlabs_key @@ -1622,6 +1691,11 @@ def __init__( self._hangup_fn = hangup_fn self._send_dtmf_fn = send_dtmf_fn self._stt = None + # Optional deferred STT connect task — set when prewarm-handoff + # parallelises STT.connect with the firstMessage TTS synth. + # Awaited BEFORE the STT receive loop starts so the message + # pump never reads from a half-open WS. + self._stt_connect_task: asyncio.Task | None = None self._tts = None # Auto-VAD: if ``agent.vad`` is None we attempt to load SileroVAD # with phone-friendly defaults during ``start()``. Stored separately @@ -1641,6 +1715,44 @@ def __init__( # the AEC filter is still converging (~500 ms warmup + safety # margin). self._speaking_started_at: float | None = None + # Wall-clock timestamp (``time.time()`` units) when the FIRST TTS + # audio chunk of the current turn actually reached the carrier wire + # — set by ``_mark_first_audio_sent`` after ``audio_sender.send_audio`` + # succeeds, cleared by ``_begin_speaking`` / barge-in cancels. The + # barge-in gate is anchored to this timestamp instead of + # ``_speaking_started_at`` because cloud TTS providers (ElevenLabs, + # Cartesia, ...) take 200-700 ms to emit the first byte. A gate + # starting at ``_begin_speaking`` would expire on background noise + # before any audio went out, exit the TTS loop on + # ``_is_speaking=False``, and silently drop the agent's first turn. + self._first_audio_sent_at: float | None = None + # Optional barge-in confirmation strategies (see + # ``getpatter.services.barge_in_strategies``). With an empty tuple + # the SDK uses the legacy "cancel on first VAD speech_start" + # behaviour. With one or more strategies, a VAD speech_start during + # TTS marks the barge-in as *pending* — the agent's TTS keeps + # streaming naturally — and the strategies are consulted on every + # STT transcript. The first strategy that approves confirms the + # barge-in and the cancel/flush sequence runs; if none confirm + # within ``_barge_in_confirm_s`` the pending state is dropped and + # the agent finishes its sentence. + self._barge_in_strategies: tuple = tuple( + getattr(agent, "barge_in_strategies", ()) or () + ) + _confirm_ms = getattr(agent, "barge_in_confirm_ms", 1500) + try: + self._barge_in_confirm_s: float = max(0.1, float(_confirm_ms) / 1000.0) + except (TypeError, ValueError): + self._barge_in_confirm_s = 1.5 + # Wall-clock timestamp of the most recent VAD-marked pending + # barge-in. ``None`` means "not pending"; a numeric value means + # the agent has already produced a turn worth of audio AND VAD + # has seen user speech, but no strategy has confirmed yet. + self._barge_in_pending_since: float | None = None + # Background task that fires the pending-timeout. Cancelled on + # confirmation, on agent stop, and on call shutdown so a stale + # pending never bleeds into the next turn. + self._barge_in_pending_task = None # Monotonic counter incremented at every TTS-start. ``_end_speaking_with_grace`` # captures the value at scheduling time and only flips ``_is_speaking`` to # False if no new turn started in the meantime. Prevents an in-flight grace @@ -1656,6 +1768,11 @@ def __init__( # 16-bit mono) ≈ 640 bytes; capped to 30 frames ≈ 600 ms ≈ # ~19 KB per concurrent call. self._inbound_audio_ring: list[bytes] = [] + # True when VAD fired ``speech_start`` during the agent's turn but + # the barge-in gate suppressed it. The grace-timer flip drains the + # ring buffer to STT so the user's words are not silently discarded. + # Mirrors TS ``suppressedSpeechPending``. + self._suppressed_speech_pending: bool = False # Wall-clock timestamp of the most recent barge-in cancel, used by # ``_begin_speaking`` to enforce a short drain window so the remote # PSTN player finishes flushing the cancelled turn's tail before @@ -1682,6 +1799,21 @@ def __init__( self._last_commit_at: float = 0.0 # Per-handler StatefulResampler for mulaw 8 kHz -> PCM16 16 kHz transcoding. self._resampler_8k_to_16k = None + # FIFO of outstanding Twilio marks the SDK has sent but not yet seen + # echoed back. Used by the firstMessage paced sender to bound the + # carrier-side buffer depth — without this the loop pushed the entire + # TTS stream into Twilio's WebSocket in one burst and a sendClear + # racing the queued media frames was unable to interrupt the agent + # for up to ~2 s (BUG #128). ``on_mark`` pops entries when Twilio + # confirms playback; ``_drain_pending_marks`` resolves every entry on + # cancel so any awaiter exits on the next tick. Telnyx never + # populates this queue (no mark concept on Telnyx's wire protocol — + # the loop falls back to time-based pacing). + self._pending_marks: list[tuple[str, asyncio.Future[None]]] = [] + # Monotonic counter for first-message mark names. Distinct from the + # generic ``audio_*`` marks the Realtime path sends so the two paths + # can coexist without name collisions. + self._first_message_mark_counter: int = 0 async def start(self) -> None: """Initialize STT/TTS providers, hooks, and start the STT receive loop.""" @@ -1761,8 +1893,8 @@ async def start(self) -> None: # Acoustic echo cancellation: opt-in. # - # Per the industry consensus (LiveKit, Pipecat, Vapi, Retell, - # Bland) and Twilio's own guidance, time-domain NLMS server-side + # Per the industry consensus on PSTN echo cancellation and + # Twilio's own guidance, time-domain NLMS server-side # AEC is the right tool only when the SDK has near-direct access # to the mic and speaker (browser WebRTC, mobile native). PSTN # paths route through a 250–1500 ms Twilio jitter buffer + carrier @@ -1801,10 +1933,116 @@ async def start(self) -> None: "install with `pip install getpatter[silero]` (numpy is part of that extra)." ) + # Prewarm-handoff: try to adopt pre-opened provider WebSockets + # that the prewarm pipeline (see + # ``Patter._park_provider_connections``) parked during the + # carrier ringing window. When a parked WS is still OPEN we + # skip the cold ``connect()`` and the STT first-turn can flow + # audio without paying the 150-400 ms TLS handshake. Failures + # (cache miss, parked WS died) fall back transparently. + parked: dict | None = None + if self._pop_prewarmed_connections is not None: + try: + parked = self._pop_prewarmed_connections(self.call_id) + except Exception as exc: # noqa: BLE001 - best-effort + logger.debug("pop_prewarmed_connections raised: %s", exc) + parked = None + + # Adopt the TTS WS first — synchronous handoff (the live + # ``synthesize`` call below picks it up via the adapter's + # single-slot adoption queue). + parked_tts = (parked or {}).get("tts") + if parked_tts is not None and self._tts is not None: + adopt = getattr(self._tts, "adopt_websocket", None) + ws_alive = parked_tts.ws is not None and not parked_tts.ws.closed + if callable(adopt) and ws_alive: + try: + adopt(parked_tts) + logger.info( + "[CONNECT] callId=%s provider=tts source=adopted ms=0", + self.call_id, + ) + except Exception as exc: # noqa: BLE001 + logger.debug("TTS adopt_websocket failed: %s; falling back", exc) + try: + await parked_tts.ws.close() + except Exception: + pass + else: + try: + await parked_tts.ws.close() + except Exception: + pass + + # Kick off STT connect WITHOUT awaiting yet — we only need STT + # ready to receive incoming user audio, not to send the first + # agent message out. Parallelising STT.connect with the TTS + # firstMessage synth shaves 200-400 ms off the perceived + # first-turn latency. + stt_connect_task: asyncio.Task | None = None if self._stt is not None: - await self._stt.connect() + parked_stt = (parked or {}).get("stt") + adopt_stt = getattr(self._stt, "adopt_websocket", None) + stt_started_at = time.monotonic() + stt_adopted = False + if ( + parked_stt is not None + and callable(adopt_stt) + and isinstance(parked_stt, tuple) + and len(parked_stt) == 2 + ): + session, ws = parked_stt + if not ws.closed: + try: + adopt_stt(session, ws) + logger.info( + "[CONNECT] callId=%s provider=stt source=adopted ms=%d", + self.call_id, + int((time.monotonic() - stt_started_at) * 1000), + ) + stt_adopted = True + except Exception as exc: # noqa: BLE001 + logger.debug( + "STT adopt_websocket failed: %s; falling back", exc + ) + try: + await ws.close() + except Exception: + pass + try: + await session.close() + except Exception: + pass + else: + try: + await ws.close() + except Exception: + pass + try: + await session.close() + except Exception: + pass + elif parked_stt is not None: + # Unknown handle shape — discard cleanly. + await _safe_close_parked_handle(parked_stt) + + if not stt_adopted: + + async def _connect_stt() -> None: + await self._stt.connect() + logger.info( + "[CONNECT] callId=%s provider=stt source=fresh ms=%d", + self.call_id, + int((time.monotonic() - stt_started_at) * 1000), + ) - logger.debug("Pipeline mode: STT + TTS connected") + stt_connect_task = asyncio.create_task(_connect_stt()) + + # Stash the deferred connect task so the receive-loop launcher + # below awaits it before starting the message pump. + self._stt_connect_task = stt_connect_task + + logger.debug("Pipeline mode: STT connect kicked off") # Play first_message if configured and no on_message handler. # Measure TTS-first-byte latency for parity with TS (`stream-handler.ts`). @@ -1822,29 +2060,61 @@ async def start(self) -> None: # the ring buffer for pre-barge-in audio is never # populated. Mirrors the per-turn behaviour in # `_process_streaming_response` / `_process_regular_response`. - await self._begin_speaking() + # + # ``is_first_message=True`` pre-stamps ``_first_audio_sent_at`` + # synchronously so the barge-in gate runs in parallel with TTS + # TTFB instead of only after audio arrives — without this, the + # firstMessage is effectively un-interruptible for 300-800 ms. + await self._begin_speaking(is_first_message=True) first_chunk_sent = False # Drop any stale PCM16 carry byte from a prior synth (none at call # start, but defensive for parity with TS ``ttsByteCarry = null``). self.audio_sender.reset_pcm_carry() + # Check the prewarm cache first. When ``Patter.call`` was made + # with ``agent.prewarm_first_message=True`` the firstMessage + # has already been synthesised during the ringing window — we + # stream the bytes directly through the carrier-side + # AudioSender (which handles native-rate → carrier-rate + # resampling) and skip the TTS round-trip entirely. + prewarm_bytes: bytes | None = None + if self._pop_prewarm_audio is not None: + try: + prewarm_bytes = self._pop_prewarm_audio(self.call_id) + except Exception as exc: # noqa: BLE001 - best-effort + logger.debug("pop_prewarm_audio raised: %s", exc) + prewarm_bytes = None try: - async for audio_chunk in self._tts.synthesize(self.agent.first_message): - if not self._is_speaking: - break # barge-in or test-hangup - if not first_chunk_sent: - first_chunk_sent = True - if self.metrics is not None: - self.metrics.record_tts_first_byte() - # Far-end tap for the echo canceller — push the - # exact PCM the carrier-side encoder will transmit. - # Without this the AEC adapt loop has no reference - # signal during the intro, resulting in unmitigated - # bleed-through and a "first turn unresponsive" UX - # where the user's voice is masked by the agent's - # TTS in the inbound channel. - if self._aec is not None: - self._aec.push_far_end(audio_chunk) - await self.audio_sender.send_audio(audio_chunk) + if prewarm_bytes: + if self.metrics is not None: + self.metrics.record_tts_first_byte() + first_chunk_sent = await self._stream_prewarm_bytes(prewarm_bytes) + else: + # Streaming TTS path (no prewarm cache). Uses the same + # simple per-chunk send as _synthesize_sentence — + # ElevenLabs HTTP streams at near-real-time speed so the + # carrier-side buffer stays bounded without mark-gated + # pacing. Routing streaming chunks through + # _send_paced_first_message_bytes caused crackling: its + # drain+reset on every HTTP chunk destroyed mark + # back-pressure continuity and the per-sub-chunk sleep + # slowed delivery below Twilio's playout rate, producing + # periodic buffer underruns. The prewarm path (a single + # pre-synthesised buffer) still uses + # _send_paced_first_message_bytes because that buffer can + # be several seconds long and needs pacing. + async for audio_chunk in self._tts.synthesize( + self.agent.first_message + ): + if not self._is_speaking: + break + if not first_chunk_sent: + first_chunk_sent = True + if self.metrics is not None: + self.metrics.record_tts_first_byte() + if self._aec is not None: + self._aec.push_far_end(audio_chunk) + await self.audio_sender.send_audio(audio_chunk) + self._mark_first_audio_sent() finally: # Drop any partial int16 byte to prevent cross-turn corruption # if the stream threw before a complete sample was delivered. @@ -1854,6 +2124,15 @@ async def start(self) -> None: # the next user utterance is recognised cleanly. await self._end_speaking_with_grace() if first_chunk_sent and self.metrics is not None: + # Bill the firstMessage TTS characters — they were synthesised + # at ElevenLabs (or the configured TTS provider) and the + # customer pays for them. The previous flow only called + # ``record_turn_complete`` here, which finalises the turn + # but does NOT increment ``_total_tts_characters`` — so a + # 5-turn call with an 82-char greeting was under-billed + # by ~22% on TTS cost. ``record_tts_complete`` is the + # canonical accumulator entry point for TTS char billing. + self.metrics.record_tts_complete(self.agent.first_message) turn = self.metrics.record_turn_complete(self.agent.first_message) self.conversation_history.append( { @@ -1929,7 +2208,27 @@ async def start(self) -> None: if is_remote_url(self.on_message): self._remote_handler = RemoteMessageHandler() - # Start STT receive loop + # Start STT receive loop. If we kicked off the WS connect in + # parallel with the firstMessage TTS, make sure that connect + # has completed before the receive loop starts polling — a + # half-open WS would surface "Not connected. Call connect() + # first." on the first audio frame. + if self._stt_connect_task is not None: + try: + await self._stt_connect_task + except Exception as exc: # noqa: BLE001 + logger.error("STT connect failed: %s", exc) + # Tear down the call cleanly — we can't proceed with + # transcription. The carrier-side pump will see the + # closed WS and end the call. + if self._hangup_fn is not None: + try: + await self._hangup_fn(self.call_id) + except Exception: + pass + return + finally: + self._stt_connect_task = None if self._stt is not None: self._stt_task = asyncio.create_task(self._stt_loop()) @@ -2107,6 +2406,7 @@ async def _synthesize_sentence( if self._aec is not None: self._aec.push_far_end(processed_audio) await self.audio_sender.send_audio(processed_audio) + self._mark_first_audio_sent() finally: await gen.aclose() _tts_span.__exit__(None, None, None) @@ -2308,28 +2608,72 @@ async def _process_regular_response(self, response_text: str, call_id: str) -> N await self._emit_turn_metrics(turn, call_id=call_id) async def _handle_barge_in(self, transcript) -> None: - """Caller spoke over in-flight TTS. Flip speaking flag, clear downstream - audio, record interruption. Mirrors TS ``handleBargeIn``. + """Decide whether ``transcript`` confirms a barge-in and run the + cancel/flush path if so. Mirrors TS ``handleBargeIn``. + + The legacy contract — "any transcript while speaking cancels the + agent" — applies when ``agent.barge_in_strategies`` is empty. + With one or more strategies configured, the transcript is fed + to :func:`evaluate_strategies` and the cancel only runs when at + least one strategy approves; otherwise the agent keeps talking. """ if not (transcript.text and self._is_speaking): return if not self._can_barge_in(): - # Same rationale as the VAD-path gate in ``on_audio_received``: - # gate is 1.0 s with AEC (filter warmup) or 0.25 s without - # (anti-flicker only). INFO so unexpected suppressions are - # visible without enabling debug logs. aec_state = "on" if getattr(self, "_aec", None) is not None else "off" logger.info( "Barge-in transcript suppressed (agent speaking < gate, aec=%s)", aec_state, ) return + strategies = getattr(self, "_barge_in_strategies", ()) or () + if strategies: + from getpatter.services.barge_in_strategies import evaluate_strategies + + confirmed = await evaluate_strategies( + strategies, + transcript=transcript.text, + is_interim=not getattr(transcript, "is_final", True), + agent_speaking=self._is_speaking, + ) + if not confirmed: + logger.debug( + "Barge-in NOT confirmed by any strategy (transcript=%r); " + "agent continues talking", + sanitize_log_value(transcript.text[:40]), + ) + return + logger.info( + "Barge-in confirmed by strategy on transcript %r", + sanitize_log_value(transcript.text[:40]), + ) + await self._do_cancel_for_barge_in(transcript.text) + + async def _do_cancel_for_barge_in(self, transcript_text: str) -> None: + """Actually cancel the in-flight agent turn (TTS + LLM stream + ring). + + Split out of :meth:`_handle_barge_in` so the same cancel logic can + run from the legacy "transcript = cancel" path AND the opt-in + "strategy confirmed = cancel" path without duplication. + """ + # Capture pending state BEFORE _clear_pending_barge_in() drops it — + # if VAD already started the overlap window via + # ``_start_pending_barge_in`` we MUST NOT call ``record_overlap_start`` + # again (that would overwrite T1 with T2 and produce a near-zero + # ``InterruptionMetrics.detection_delay_ms`` on the strategy path). + # ``getattr`` is defensive against test fixtures that build a + # handler shell via ``object.__new__`` and don't initialise the + # pending-barge-in state — the safe default is "no pending". + had_pending = getattr(self, "_barge_in_pending_since", None) is not None + self._clear_pending_barge_in() if self.metrics is not None: - self.metrics.record_overlap_start() + if not had_pending: + # Legacy path or VAD never fired — start the overlap window now. + self.metrics.record_overlap_start() self.metrics.record_bargein_detected() logger.debug( "Barge-in: caller spoke over agent (%s)", - sanitize_log_value(transcript.text[:40]), + sanitize_log_value(transcript_text[:40]), ) with start_span( SPAN_BARGEIN, @@ -2337,17 +2681,18 @@ async def _handle_barge_in(self, transcript) -> None: ): self._is_speaking = False self._speaking_started_at = None - # Record cancel timestamp so ``_begin_speaking`` can enforce - # a short drain window before the next TTS chunk lands on - # top of the cancelled turn's tail (avoids audible "doubled - # audio" on the first sentence post-barge-in). Mirrors the - # VAD-path cancel branch — both barge-in paths must set the - # timestamp for the drain to be effective. + self._first_audio_sent_at = None self._last_cancel_at = time.time() - # Signal the in-flight LLM-consumption loop to stop fetching - # tokens. The consume loop checks ``_llm_cancel_event`` between - # iterations and ``aclose()``s the generator on exit, freeing - # the upstream HTTP/WS slot and stopping further token billing. + # Unblock any firstMessage paced-send loop that's sitting in + # ``_wait_for_mark_window`` — without this the loop keeps + # awaiting echoes for up to ``_MARK_AWAIT_TIMEOUT_S`` per + # outstanding mark before observing ``_is_speaking=False``, + # which keeps the agent "speaking" from the user's perspective + # for hundreds of extra ms after barge-in (BUG #128). Defensive + # ``getattr`` is for test fixtures that build a handler shell + # via ``object.__new__`` and skip ``__init__``. + if getattr(self, "_pending_marks", None) is not None: + self._drain_pending_marks() cancel_event = getattr(self, "_llm_cancel_event", None) if cancel_event is not None: cancel_event.set() @@ -2358,8 +2703,64 @@ async def _handle_barge_in(self, transcript) -> None: if self.metrics is not None: self.metrics.record_tts_stopped() self.metrics.record_turn_interrupted() + # Re-anchor to legitimate VAD speech_start so post-barge-in + # latency anchors don't carry from the interrupted turn. + self.metrics.anchor_user_speech_start() self.metrics.record_overlap_end(was_interruption=True) + async def _start_pending_barge_in(self) -> None: + """Mark a VAD-detected barge-in as pending (no cancel yet). + + Only used when ``agent.barge_in_strategies`` is non-empty. The + agent's TTS keeps streaming naturally; an + :meth:`_pending_barge_in_timeout` task will drop the pending + state if no strategy confirms within ``_barge_in_confirm_s``. + """ + if self._barge_in_pending_since is not None: + return + self._barge_in_pending_since = time.time() + if self.metrics is not None: + self.metrics.record_overlap_start() + logger.info( + "Barge-in PENDING (VAD speech_start during TTS); awaiting strategy confirmation" + ) + try: + self._barge_in_pending_task = asyncio.create_task( + self._pending_barge_in_timeout() + ) + except RuntimeError as exc: # pragma: no cover - no running loop + logger.debug("could not schedule pending barge-in timeout: %s", exc) + self._barge_in_pending_task = None + + async def _pending_barge_in_timeout(self) -> None: + try: + await asyncio.sleep(self._barge_in_confirm_s) + except asyncio.CancelledError: + return + if self._barge_in_pending_since is None: + return + logger.info( + "Pending barge-in timed out after %.2fs; agent resumes (no strategy confirmed)", + self._barge_in_confirm_s, + ) + if self.metrics is not None: + self.metrics.record_overlap_end(was_interruption=False) + # Re-anchor to legitimate VAD speech_start so anchors that drifted + # during the pending barge-in window don't pollute the next turn. + self.metrics.anchor_user_speech_start() + self._barge_in_pending_since = None + self._barge_in_pending_task = None + + def _clear_pending_barge_in(self) -> None: + """Drop pending state without cancelling — used on confirm and on + agent stop. Idempotent and safe to call from test fixtures that + construct the handler via ``object.__new__`` (no __init__).""" + task = getattr(self, "_barge_in_pending_task", None) + if task is not None and not task.done(): + task.cancel() + self._barge_in_pending_task = None + self._barge_in_pending_since = None + def _commit_transcript(self, text: str) -> bool: """Dedup + throttle + hallucination filter for final STT transcripts. @@ -2427,193 +2828,197 @@ async def _stt_loop(self) -> None: ): continue if not self._commit_transcript(transcript.text): + # Final transcript dropped (dedup / hallucination / + # back-to-back). Any VAD ``speech_end`` that fired + # during this dropped utterance already stamped + # ``_endpoint_signal_at``; if we leave it there, the + # NEXT legitimate utterance inherits the stale anchor + # (its agent_response_ms then includes the silence + # gap between the dropped utterance and the real one). + if self.metrics is not None: + self.metrics.anchor_user_speech_start() continue - # Record one STT span per final transcript turn. The span is - # short-lived (just the attribute set) because STT is - # streaming — we do not re-wrap the long-lived iterator. - with start_span( - SPAN_STT, - { - "getpatter.stt.text_len": len(transcript.text), - "getpatter.stt.confidence": float(transcript.confidence or 0.0), - "patter.call.id": self.call_id, - }, - ): - pass - - logger.debug("User: %s", sanitize_log_value(transcript.text)) - - if self.metrics is not None: - self.metrics.start_turn_if_idle() # turn may already be open - # Known limitation: per-turn audio_seconds is not tracked - # here; metrics rely on total _stt_byte_count plus the - # end_call() estimation pass. - self.metrics.record_vad_stop() - self.metrics.record_stt_complete(transcript.text) - self.metrics.record_stt_final_timestamp() - - # Endpoint span — silence-detected → LLM-dispatch window. Open - # here (right after VAD stop / final transcript is recorded) - # and close it just before ``record_turn_committed`` below. - endpoint_span = start_span( - SPAN_ENDPOINT, - {"patter.call.id": self.call_id}, - ) - endpoint_span.__enter__() - # Wrapped in a list so the closure-style helper can flip the - # flag without needing ``nonlocal`` (we are inside a loop body, - # not a nested function — ``nonlocal`` would not bind here). - _endpoint_closed = [False] - - def _close_endpoint_span() -> None: - if _endpoint_closed[0]: - return - _endpoint_closed[0] = True - try: - endpoint_span.__exit__(None, None, None) - except Exception: # pragma: no cover - defensive - pass + await self._dispatch_turn(transcript.text) - # Raw transcript always goes to dashboard/transcript log - self.transcript_entries.append( - {"role": "user", "text": transcript.text} - ) + except Exception as exc: + logger.exception("Pipeline STT loop error: %s", exc) - if self.on_transcript: - await self.on_transcript( - { - "role": "user", - "text": transcript.text, - "call_id": self.call_id, - "history": list(self.conversation_history), - } - ) + async def _dispatch_turn(self, transcript_text: str) -> None: + """Run the post-commit pipeline (record STT → afterTranscribe → + LLM dispatch → TTS → turn-complete) inline on the STT loop. + """ + # Record one STT span per final transcript turn. The span is + # short-lived (just the attribute set) because STT is + # streaming — we do not re-wrap the long-lived iterator. + with start_span( + SPAN_STT, + { + "getpatter.stt.text_len": len(transcript_text), + "patter.call.id": self.call_id, + }, + ): + pass - # --- afterTranscribe hook --- - hooks = getattr(self.agent, "hooks", None) - hook_executor = PipelineHookExecutor(hooks) - hook_ctx = self._build_hook_context() - filtered_text = await hook_executor.run_after_transcribe( - transcript.text, hook_ctx - ) - if filtered_text is None: - logger.debug("afterTranscribe hook vetoed turn") - if self.metrics is not None: - self.metrics.record_turn_interrupted() - _close_endpoint_span() - continue + logger.debug("User: %s", sanitize_log_value(transcript_text)) - if self.metrics is not None: - self.metrics.record_on_user_turn_completed_delay(0.0) - if self.on_message is None and self._llm_loop is None: - # No message handler or LLM loop — discard orphaned turn - if self.metrics is not None: - self.metrics.record_turn_interrupted() - _close_endpoint_span() - continue + if self.metrics is not None: + self.metrics.start_turn_if_idle() # turn may already be open + # Known limitation: per-turn audio_seconds is not tracked + # here; metrics rely on total _stt_byte_count plus the + # end_call() estimation pass. + self.metrics.record_vad_stop() + self.metrics.record_stt_complete(transcript_text) + self.metrics.record_stt_final_timestamp() + + # Endpoint span — silence-detected → LLM-dispatch window. Open + # here (right after VAD stop / final transcript is recorded) + # and close it just before ``record_turn_committed`` below. + endpoint_span = start_span( + SPAN_ENDPOINT, + {"patter.call.id": self.call_id}, + ) + endpoint_span.__enter__() + endpoint_closed = False - # Use filtered text in conversation history (sent to LLM) - self.conversation_history.append( - {"role": "user", "text": filtered_text, "timestamp": time.time()} - ) + def _close_endpoint_span() -> None: + nonlocal endpoint_closed + if endpoint_closed: + return + endpoint_closed = True + try: + endpoint_span.__exit__(None, None, None) + except Exception: # pragma: no cover - defensive + pass - # Built-in LLM loop path - if self.on_message is None and self._llm_loop is not None: - call_ctx = { - "call_id": self.call_id, - "caller": self.caller, - "callee": self.callee, - } - if self.metrics is not None: - self.metrics.record_turn_committed() - _close_endpoint_span() - result = self._llm_loop.run( - filtered_text, - list(self.conversation_history), - call_ctx, - hook_executor=hook_executor, - hook_ctx=hook_ctx, - cancel_event=self._llm_cancel_event, - ) - response_text = await self._process_streaming_response( - result, self.call_id - ) - if response_text: - await self._emit_assistant_transcript(response_text) - continue + # Raw transcript always goes to dashboard/transcript log + self.transcript_entries.append({"role": "user", "text": transcript_text}) - # on_message handler path - if self.metrics is not None: - self.metrics.record_turn_committed() - _close_endpoint_span() - msg_data = { - "text": filtered_text, + if self.on_transcript: + await self.on_transcript( + { + "role": "user", + "text": transcript_text, "call_id": self.call_id, - "caller": self.caller, - "callee": self.callee, "history": list(self.conversation_history), } + ) - response_text = "" - streaming = False + # --- afterTranscribe hook --- + hooks = getattr(self.agent, "hooks", None) + hook_executor = PipelineHookExecutor(hooks) + hook_ctx = self._build_hook_context() + filtered_text = await hook_executor.run_after_transcribe( + transcript_text, hook_ctx + ) + if filtered_text is None: + logger.debug("afterTranscribe hook vetoed turn") + if self.metrics is not None: + self.metrics.record_turn_interrupted() + _close_endpoint_span() + return - from getpatter.services.remote_message import ( - is_remote_url, - is_websocket_url, - ) + if self.metrics is not None: + self.metrics.record_on_user_turn_completed_delay(0.0) + if self.on_message is None and self._llm_loop is None: + # No message handler or LLM loop — discard orphaned turn + if self.metrics is not None: + self.metrics.record_turn_interrupted() + _close_endpoint_span() + return - if is_remote_url(self.on_message): - remote = self._remote_handler - if is_websocket_url(self.on_message): - result = remote.call_websocket(self.on_message, msg_data) - streaming = True - else: - response_text = await remote.call_webhook( - self.on_message, msg_data - ) - streaming = False - elif self._msg_accepts_call: - result = self.on_message(msg_data, self._call_control) - else: - result = self.on_message(msg_data) - - if not is_remote_url(self.on_message): - if asyncio.iscoroutine(result): - response_text = await result - streaming = False - elif inspect.isasyncgen(result): - streaming = True - else: - response_text = result - streaming = False + # Use filtered text in conversation history (sent to LLM) + self.conversation_history.append( + {"role": "user", "text": filtered_text, "timestamp": time.time()} + ) + + # Built-in LLM loop path + if self.on_message is None and self._llm_loop is not None: + call_ctx = { + "call_id": self.call_id, + "caller": self.caller, + "callee": self.callee, + } + if self.metrics is not None: + self.metrics.record_turn_committed() + _close_endpoint_span() + result = self._llm_loop.run( + filtered_text, + list(self.conversation_history), + call_ctx, + hook_executor=hook_executor, + hook_ctx=hook_ctx, + cancel_event=self._llm_cancel_event, + ) + response_text = await self._process_streaming_response(result, self.call_id) + if response_text: + await self._emit_assistant_transcript(response_text) + return - # Check if handler ended the call - if self._call_control is not None and self._call_control.ended: - return + # on_message handler path + if self.metrics is not None: + self.metrics.record_turn_committed() + _close_endpoint_span() + msg_data = { + "text": filtered_text, + "call_id": self.call_id, + "caller": self.caller, + "callee": self.callee, + "history": list(self.conversation_history), + } - if streaming: - response_text = await self._process_streaming_response( - result, self.call_id - ) - if response_text: - await self._emit_assistant_transcript(response_text) - else: - if not response_text: - # Common misuse: on_message was provided as an observer - # (returning None) but it actually replaces the built-in LLM - # loop. Warn loudly — the caller hears no audio until the - # handler returns a non-empty string. - logger.warning( - "on_message returned empty/None — no TTS will play. " - "If you intended to observe transcripts, use on_transcript " - "instead; if you meant to answer via the built-in LLM, " - "remove on_message and pass openai_key." - ) - await self._process_regular_response(response_text, self.call_id) + response_text = "" + streaming = False - except Exception as exc: - logger.exception("Pipeline STT loop error: %s", exc) + from getpatter.services.remote_message import ( + is_remote_url, + is_websocket_url, + ) + + if is_remote_url(self.on_message): + remote = self._remote_handler + if is_websocket_url(self.on_message): + result = remote.call_websocket(self.on_message, msg_data) + streaming = True + else: + response_text = await remote.call_webhook(self.on_message, msg_data) + streaming = False + elif self._msg_accepts_call: + result = self.on_message(msg_data, self._call_control) + else: + result = self.on_message(msg_data) + + if not is_remote_url(self.on_message): + if asyncio.iscoroutine(result): + response_text = await result + streaming = False + elif inspect.isasyncgen(result): + streaming = True + else: + response_text = result + streaming = False + + # Check if handler ended the call + if self._call_control is not None and self._call_control.ended: + return + + if streaming: + response_text = await self._process_streaming_response(result, self.call_id) + if response_text: + await self._emit_assistant_transcript(response_text) + else: + if not response_text: + # Common misuse: on_message was provided as an observer + # (returning None) but it actually replaces the built-in LLM + # loop. Warn loudly — the caller hears no audio until the + # handler returns a non-empty string. + logger.warning( + "on_message returned empty/None — no TTS will play. " + "If you intended to observe transcripts, use on_transcript " + "instead; if you meant to answer via the built-in LLM, " + "remove on_message and pass openai_key." + ) + await self._process_regular_response(response_text, self.call_id) async def on_audio_received(self, audio_bytes: bytes) -> None: """Forward caller audio to STT (transcoding to PCM16 16 kHz, running VAD/hooks).""" @@ -2663,12 +3068,24 @@ async def on_audio_received(self, audio_bytes: bytes) -> None: vad_event = None if vad_event is not None: if vad_event.type == "speech_start": - if self._is_speaking and not self._can_barge_in(): + phantom_suppressed = self._is_speaking and not self._can_barge_in() + if phantom_suppressed: # Within the per-turn warmup gate. With AEC on # this is the ~1 s filter convergence window; # without AEC it is just a 0.25 s anti-flicker # margin. INFO so unexpected suppressions are # visible without enabling debug logs. + # + # CRITICAL: do NOT touch metrics state here. + # An earlier bug (pre-0.6.1) called + # ``start_turn_if_idle()`` for every + # ``speech_start`` including suppressed phantoms, + # which stamped ``_turn_start`` at echo/loopback + # time. ``start_turn_if_idle`` then no-op'd on + # the legitimate user-speech ``speech_start`` + # that followed (turn_start was already set), + # so ``user_speech_duration_ms`` was reported as + # 5-7 s even on short ~1 s utterances. aec_state = ( "on" if getattr(self, "_aec", None) is not None else "off" ) @@ -2676,41 +3093,51 @@ async def on_audio_received(self, audio_bytes: bytes) -> None: "VAD speech_start suppressed (agent speaking < gate, aec=%s)", aec_state, ) + # Real user speech detected but gated out. The + # grace-timer flip will drain the ring buffer to + # STT so the user's words are not silently lost. + self._suppressed_speech_pending = True elif self._is_speaking: - # Caller spoke over in-flight TTS — preempt now. - if self.metrics is not None: - self.metrics.record_bargein_detected() - with start_span( - SPAN_BARGEIN, - {"patter.call.id": self.call_id}, - ): - try: - await self.audio_sender.send_clear() - except Exception as exc: - logger.debug( - "send_clear during VAD barge-in failed: %s", exc - ) - # Replay the ring buffer of inbound frames - # captured while the agent was speaking — - # see ``_flush_inbound_audio_ring`` for the - # full rationale. - await self._flush_inbound_audio_ring() + # Caller spoke over in-flight TTS. With opt-in + # confirmation strategies the cancel is deferred + # until at least one strategy approves the user's + # transcript; otherwise we keep the legacy + # "cancel immediately" path so existing users + # see no behaviour change. + if self._barge_in_strategies: + await self._start_pending_barge_in() + else: if self.metrics is not None: - self.metrics.record_tts_stopped() - self.metrics.record_turn_interrupted() - # Force-flip immediately and bump the generation so a - # pending grace-flip from the prior turn can't fight us. - self._is_speaking = False - self._speaking_started_at = None - self._speaking_generation += 1 - # Record cancel timestamp so ``_begin_speaking`` - # can enforce a short drain window before the - # next TTS chunk lands on top of the cancelled - # turn's tail (avoids audible "doubled audio" - # on the first sentence post-barge-in). - self._last_cancel_at = time.time() - if self.metrics is not None: - self.metrics.start_turn_if_idle() + self.metrics.record_bargein_detected() + with start_span( + SPAN_BARGEIN, + {"patter.call.id": self.call_id}, + ): + try: + await self.audio_sender.send_clear() + except Exception as exc: + logger.debug( + "send_clear during VAD barge-in failed: %s", + exc, + ) + await self._flush_inbound_audio_ring() + if self.metrics is not None: + self.metrics.record_tts_stopped() + self.metrics.record_turn_interrupted() + self._is_speaking = False + self._speaking_started_at = None + self._first_audio_sent_at = None + self._speaking_generation += 1 + self._last_cancel_at = time.time() + self._suppressed_speech_pending = False + if not phantom_suppressed and self.metrics is not None: + # Industry-standard pattern: every legitimate VAD speech_start + # re-anchors the turn timestamp pre-commit. This + # repairs the case where a partial transcript / + # rejected barge-in already stamped stale anchors, + # plus the original "phantom during warmup gate" + # vulnerability. No-op once the turn is committed. + self.metrics.anchor_user_speech_start() elif vad_event.type == "speech_end": if self.metrics is not None: self.metrics.record_vad_stop() @@ -2795,7 +3222,7 @@ async def on_audio_received(self, audio_bytes: bytes) -> None: # ``StreamHandler.POST_CANCEL_DRAIN_MS``. _POST_CANCEL_DRAIN_S: float = 0.15 - async def _begin_speaking(self) -> None: + async def _begin_speaking(self, is_first_message: bool = False) -> None: """Mark TTS playback as in-progress and bump the generation counter. Awaits the post-cancel drain window before flipping state so the @@ -2804,6 +3231,15 @@ async def _begin_speaking(self) -> None: The generation counter is consulted by ``_end_speaking_with_grace`` so a delayed flip-to-idle from a previous turn cannot cancel the speaking flag of the *current* turn. + + Args: + is_first_message: When ``True`` stamps ``_first_audio_sent_at`` + synchronously before the TTS loop starts so the + ``_can_barge_in()`` 250 ms anti-flicker gate (no-AEC PSTN + default) runs in PARALLEL with TTS TTFB rather than only + starting after audio actually arrives. Without this, the + firstMessage is effectively un-interruptible for the first + 300-800 ms while waiting on cloud TTS first-byte. """ if self._last_cancel_at is not None: elapsed = time.time() - self._last_cancel_at @@ -2813,9 +3249,43 @@ async def _begin_speaking(self) -> None: self._speaking_generation += 1 self._is_speaking = True self._speaking_started_at = time.time() + # Stamp ``_first_audio_sent_at`` synchronously for EVERY turn so the + # ``_can_barge_in()`` gate (250 ms anti-flicker for PSTN no-AEC) runs + # in PARALLEL with LLM TTFT + TTS TTFB rather than starting only + # after the first audio chunk reaches the wire. Without this, a turn + # with a slow LLM (gpt-4o cold cache ~2 s) is effectively + # un-interruptible for the entire LLM window: ``_first_audio_sent_at`` + # stays None, ``_can_barge_in`` returns False, and every VAD + # ``speech_start`` is suppressed silently. Promoted from + # firstMessage-only to default on 2026-05-14 (TS parity). + # ``is_first_message`` is kept for backward compat with callers but + # no longer changes behaviour. + _ = is_first_message + self._first_audio_sent_at = time.time() # Fresh turn — drop any stale pre-barge-in buffer from a previous # turn so we never replay yesterday's audio to STT. self._inbound_audio_ring = [] + self._suppressed_speech_pending = False + # Reset the VAD detector so the next user utterance triggers a clean + # SILENCE→SPEECH transition. Without this, PSTN echo from the + # previous turn can keep the smoothed probability above the + # deactivation threshold (0.35) for the entire turn — the VAD never + # returns to SILENCE, ``speech_start`` never fires, and barge-in + # feels "one-shot". The user's previous utterance was already + # committed by STT before ``_begin_speaking`` is called, so resetting + # state here cannot lose data. + self._reset_vad() + + def _mark_first_audio_sent(self) -> None: + """Record that the first TTS chunk of the current turn hit the wire. + + Idempotent within a turn: only the first call sets the timestamp. + Must be invoked AFTER the underlying ``audio_sender.send_audio`` so + the gate is anchored to "audio actually went out", not "we asked + the carrier to send it". Mirrors TS ``markFirstAudioSent``. + """ + if self._first_audio_sent_at is None: + self._first_audio_sent_at = time.time() def _can_barge_in(self) -> bool: """Whether barge-in is allowed to fire right now. @@ -2832,7 +3302,17 @@ def _can_barge_in(self) -> bool: started_at = getattr(self, "_speaking_started_at", None) if started_at is None: return True - elapsed = time.time() - started_at + # Anchor the gate on "first audio actually emitted", not on + # ``_begin_speaking`` (which fires before the TTS provider's + # first-byte latency has elapsed). Without this guard, background + # noise picked up by VAD ~250 ms after ``_begin_speaking`` triggers + # a self-cancel BEFORE any TTS chunk has reached the wire — the + # agent's first turn becomes silence even though the SDK believes + # it spoke. Mirrors TS ``canBargeIn``. + first_audio_at = getattr(self, "_first_audio_sent_at", None) + if first_audio_at is None: + return False + elapsed = time.time() - first_audio_at gate = ( MIN_AGENT_SPEAKING_S_BEFORE_BARGE_IN_AEC if getattr(self, "_aec", None) is not None @@ -2869,6 +3349,13 @@ async def _end_speaking_with_grace(self) -> None: if grace_ms <= 0: self._is_speaking = False self._speaking_started_at = None + self._first_audio_sent_at = None + self._clear_pending_barge_in() + await self._reset_barge_in_strategies() + if self._suppressed_speech_pending: + self._suppressed_speech_pending = False + await self._flush_inbound_audio_ring() + self._reset_vad() return gen = self._speaking_generation @@ -2881,6 +3368,16 @@ async def _flip_after_grace() -> None: if self._speaking_generation == gen: self._is_speaking = False self._speaking_started_at = None + self._first_audio_sent_at = None + self._clear_pending_barge_in() + await self._reset_barge_in_strategies() + if self._suppressed_speech_pending: + self._suppressed_speech_pending = False + await self._flush_inbound_audio_ring() + # Reset VAD so any stuck SPEECH state from echo / + # loopback during the agent's turn does not block the + # next user utterance from emitting ``speech_start``. + self._reset_vad() except asyncio.CancelledError: # pragma: no cover raise except Exception as exc: # pragma: no cover - defensive @@ -2888,6 +3385,32 @@ async def _flip_after_grace() -> None: asyncio.create_task(_flip_after_grace()) + async def _reset_barge_in_strategies(self) -> None: + if not self._barge_in_strategies: + return + from getpatter.services.barge_in_strategies import reset_strategies + + await reset_strategies(self._barge_in_strategies) + + def _reset_vad(self) -> None: + """Reset the active VAD provider's per-utterance state. + + No-op when the provider does not implement the optional + :py:meth:`getpatter.providers.base.VADProvider.reset` hook + (default implementation in ``VADProvider`` is a no-op). Safe to + call from any context — failures are swallowed; a flaky reset + must never silently kill barge-in for every subsequent turn. + + Parity with TS ``resetVad``. + """ + vad = getattr(self.agent, "vad", None) or self._auto_vad + if vad is None: + return + try: + vad.reset() + except Exception as exc: # pragma: no cover - defensive + logger.debug("VAD reset threw: %s", exc) + async def _flush_inbound_audio_ring(self) -> None: """Replay the audio captured by the self-hearing guard right before a confirmed barge-in. @@ -2915,8 +3438,214 @@ async def _flush_inbound_audio_ring(self) -> None: replayed * 20, ) + # 40 ms @ 16 kHz mono PCM16 = 1280 bytes. Sized to mirror the smallest + # live-TTS chunk boundary so cancel granularity (mark/clear bookkeeping) + # is identical regardless of whether the firstMessage came from the + # prewarm cache or a live ``tts.synthesize`` stream. + _PREWARM_CHUNK_BYTES: int = 1280 + # Maximum unconfirmed Twilio marks while streaming firstMessage. Each + # chunk is 40 ms of audio at 16 kHz PCM16, so a window of 3 caps the + # in-flight queue at ~120 ms. This means a barge-in's ``send_clear`` has + # at most ~120 ms of buffered audio to flush — vs. ~2-5 s with the + # previous burst-send code (BUG #128). 3 hit the smallest barge-in cap + # without audible playback gaps under typical PSTN RTT in 2026-05 + # acceptance. + _FIRST_MESSAGE_MARK_WINDOW: int = 3 + # Per-chunk soft timeout (s) for awaiting a mark echo. Caps the + # deadlock window when a carrier (or a test double) never echoes — + # playout may glitch by one chunk on timeout but the call stays alive. + _MARK_AWAIT_TIMEOUT_S: float = 0.5 + # Bytes-per-millisecond for a 16 kHz PCM16 mono stream — used by the + # non-Twilio firstMessage pacing path to translate chunk size into a + # playout-duration sleep. 16000 samples/sec × 2 bytes = 32 bytes/ms. + _PCM16_16K_BYTES_PER_MS: int = 32 + + def _drain_pending_marks(self) -> None: + """Resolve every entry in ``_pending_marks`` and empty the FIFO. + + Idempotent — safe to call from the barge-in cancel path and again + from the grace flip without leaking unresolved futures. + """ + if not self._pending_marks: + return + for _name, fut in self._pending_marks: + if not fut.done(): + try: + fut.set_result(None) + except asyncio.InvalidStateError: + pass + self._pending_marks.clear() + + async def _send_mark_awaitable(self) -> asyncio.Future | None: + """Send a Twilio ``mark`` event and return a future that resolves + when the carrier echoes it back (via :meth:`on_mark`), or when + :meth:`_drain_pending_marks` runs. Returns ``None`` on non-Twilio + carriers — the caller should fall back to time-based pacing. + """ + if not self._for_twilio: + return None + self._first_message_mark_counter += 1 + mark_name = f"fm_{self._first_message_mark_counter}" + loop = asyncio.get_event_loop() + fut: asyncio.Future[None] = loop.create_future() + self._pending_marks.append((mark_name, fut)) + try: + await self.audio_sender.send_mark(mark_name) + except Exception as exc: # noqa: BLE001 - best effort + logger.debug("send_mark failed (%s): %s", mark_name, exc) + # Drop the waiter so the queue can't fill with orphans. + for idx, (name, f) in enumerate(self._pending_marks): + if name == mark_name: + self._pending_marks.pop(idx) + break + if not fut.done(): + fut.set_result(None) + return fut + + async def _wait_for_mark_window(self) -> None: + """Block until the in-flight mark queue depth is below + ``_FIRST_MESSAGE_MARK_WINDOW``. Returns immediately on cancel + because :meth:`_drain_pending_marks` resolves every pending future. + """ + while ( + self._is_speaking + and len(self._pending_marks) >= self._FIRST_MESSAGE_MARK_WINDOW + ): + _name, oldest = self._pending_marks[0] + try: + await asyncio.wait_for( + asyncio.shield(oldest), + timeout=self._MARK_AWAIT_TIMEOUT_S, + ) + except asyncio.TimeoutError: + # Drop the head so subsequent loops don't deadlock on the + # same mark forever. Twilio mark echo may have been lost + # in transit; carrier playback will continue regardless. + pass + # Pop the head if still present (a successful echo would have + # done it via ``on_mark``; only a timeout leaves it in place). + if self._pending_marks and self._pending_marks[0][0] == _name: + self._pending_marks.pop(0) + + async def on_mark(self, mark_name: str) -> None: + """Handle a Twilio ``mark`` echo and resolve the matching firstMessage + waiter (if any). Marks are matched FIFO: an echo for ``fm_3`` also + resolves ``fm_1`` and ``fm_2`` in case the carrier batches echoes. + """ + if not mark_name: + return + idx = -1 + for i, (name, _fut) in enumerate(self._pending_marks): + if name == mark_name: + idx = i + break + if idx < 0: + return + resolved = self._pending_marks[: idx + 1] + del self._pending_marks[: idx + 1] + for _name, fut in resolved: + if not fut.done(): + try: + fut.set_result(None) + except asyncio.InvalidStateError: + pass + + async def _stream_prewarm_bytes(self, prewarm_bytes: bytes) -> bool: + """Stream a cached firstMessage buffer in pacing-friendly chunks.""" + return await self._send_paced_first_message_bytes(prewarm_bytes) + + async def _send_paced_first_message_bytes(self, bytes_: bytes) -> bool: + """Iterate ``bytes_`` as ``_PREWARM_CHUNK_BYTES``-sized PCM16 slices + and forward each via ``audio_sender.send_audio`` with mark-gated + pacing (Twilio) or playout-time-based pacing (Telnyx). + + Caps the carrier-side buffer at ``_FIRST_MESSAGE_MARK_WINDOW`` + chunks so a barge-in's ``send_clear`` has at most ~120 ms (Twilio) + or zero (Telnyx, immediately after the latest sleep) of audio to + flush. The previous burst-send code let Twilio's buffer reach + several seconds — a barge-in's ``send_clear`` race-lost against + the queued media frames and the agent kept talking on the user's + earpiece for up to ~2 s after the user spoke (BUG #128). + + Bails immediately when ``_is_speaking`` flips to ``False`` — both + via the loop's pre-iter check and via :meth:`_drain_pending_marks` + (called from the barge-in cancel path) which unblocks any + in-flight :meth:`_wait_for_mark_window` await. + + Returns ``True`` when at least one chunk hit the wire — the caller + uses that to decide whether to record the TTS-first-byte / + turn-complete metrics. + """ + # Reset the per-send mark counter so each invocation produces a + # fresh ``fm_1, fm_2, ...`` sequence. Without this the counter + # grows monotonically across turns on a re-used handler and a + # stale ``fm_N`` echo from an earlier turn could match a mark + # name issued later, corrupting the FIFO matching in + # ``on_mark``. The ``_pending_marks`` queue is also expected + # empty here by the caller's cancel / cleanup paths; if it is + # not (defensive re-entry) we drain before resetting. + if self._pending_marks: + self._drain_pending_marks() + self._first_message_mark_counter = 0 + first_chunk_sent = False + # Once the mark window is first filled we switch to playout-time pacing + # to prevent batch-ACK bursts. Before that we send in burst so the first + # _FIRST_MESSAGE_MARK_WINDOW chunks pre-fill the PSTN jitter buffer. + initial_fill_complete = False + for i in range(0, len(bytes_), self._PREWARM_CHUNK_BYTES): + if not self._is_speaking: + break # barge-in mid-buffer — stop now + # Back-pressure: if too many marks are unconfirmed, wait. + # Drains immediately on cancel. + await self._wait_for_mark_window() + if not self._is_speaking: + break + chunk = bytes_[i : i + self._PREWARM_CHUNK_BYTES] + if not first_chunk_sent: + first_chunk_sent = True + if self._aec is not None: + self._aec.push_far_end(chunk) + await self.audio_sender.send_audio(chunk) + self._mark_first_audio_sent() + mark_future = await self._send_mark_awaitable() + if ( + not initial_fill_complete + and len(self._pending_marks) >= self._FIRST_MESSAGE_MARK_WINDOW + ): + initial_fill_complete = True + # Telnyx has no mark concept — always pace by playout time. + # Twilio: the first _FIRST_MESSAGE_MARK_WINDOW chunks go out in burst + # to pre-fill the PSTN jitter buffer (250–1500 ms), then playout-time + # pacing kicks in (via the sticky initial_fill_complete flag) to prevent + # batch-ACK bursts from draining the buffer → crackling. + if mark_future is None or initial_fill_complete: + playout_ms = max(1, len(chunk) // self._PCM16_16K_BYTES_PER_MS) + await asyncio.sleep(playout_ms / 1000.0) + return first_chunk_sent + async def cleanup(self) -> None: """Cancel the STT loop and close STT/TTS/remote-message adapters.""" + # Drop any pending barge-in timeout BEFORE we tear down metrics / + # adapters. Without this, a call that ends while a barge-in is + # pending leaves an asyncio.Task scheduled to fire + # ``_barge_in_confirm_s`` later and call + # ``metrics.record_overlap_end`` on a finalised metrics object — + # a slow leak in long-running servers and a race producing + # spurious overlap_end events. Idempotent: safe to call when no + # pending state exists. + self._clear_pending_barge_in() + # Resolve every pending firstMessage mark future before tearing + # down adapters. Without this, a call that ends abnormally mid + # firstMessage (carrier WS drop, hangup during the paced sender) + # leaves orphan ``asyncio.Future`` instances awaited by the send + # loop that nothing will ever resolve. + if getattr(self, "_pending_marks", None) is not None: + self._drain_pending_marks() + # Reset the firstMessage mark counter so a re-used handler + # instance starts ``fm_`` numbering at 1 on the next call. + # See ``_send_paced_first_message_bytes`` for the per-send reset + # that protects the within-call path. + self._first_message_mark_counter = 0 if self._stt_task: self._stt_task.cancel() try: diff --git a/libraries/python/getpatter/telephony/telnyx.py b/libraries/python/getpatter/telephony/telnyx.py index 6f10301a..2d7dc8f1 100644 --- a/libraries/python/getpatter/telephony/telnyx.py +++ b/libraries/python/getpatter/telephony/telnyx.py @@ -4,6 +4,7 @@ import asyncio import base64 +import contextlib import json import logging import re @@ -11,6 +12,7 @@ from collections import deque from urllib.parse import quote +from getpatter.observability.attributes import patter_call_scope from getpatter.telephony.common import _validate_e164 from getpatter.utils.log_sanitize import mask_phone_number from getpatter.stream_handler import ( @@ -260,6 +262,9 @@ async def telnyx_stream_bridge( on_metrics=None, pricing: dict | None = None, report_only_initial_ttfb: bool = False, + patter_side: str = "uut", + pop_prewarm_audio=None, + pop_prewarmed_connections=None, ) -> None: """Bridge a Telnyx WebSocket media stream to the configured AI provider. @@ -301,6 +306,16 @@ async def telnyx_stream_bridge( audio_sender: TelnyxAudioSender | None = None metrics = None + # Wall-clock duration tracking for patter.cost.telephony_minutes. Set on + # the ``start`` event so we measure only the bridged audio period, not + # the time spent waiting for the first frame. + _call_start_monotonic: float | None = None + + # ExitStack lets us enter ``patter_call_scope`` *after* the start frame + # arrives (when call_id is known) while still keeping the scope active + # for the entire WebSocket loop AND the finally cleanup block. + _scope_stack = contextlib.ExitStack() + try: while True: raw = await websocket.receive_text() @@ -321,6 +336,7 @@ async def telnyx_stream_bridge( if event_type_telnyx == "start" and not stream_started: stream_started = True + _call_start_monotonic = time.monotonic() start_info = data.get("start", {}) or {} call_id_actual = start_info.get("call_control_id", "") caller = start_info.get("from", "") or caller @@ -568,6 +584,8 @@ async def _telnyx_stop_recording() -> None: on_message=on_message, on_metrics=on_metrics, transcript_entries=transcript_entries, + pop_prewarm_audio=pop_prewarm_audio, + pop_prewarmed_connections=pop_prewarmed_connections, ) elif provider == "elevenlabs_convai": handler = ElevenLabsConvAIStreamHandler( @@ -607,6 +625,24 @@ async def _telnyx_stop_recording() -> None: audio_format="g711_ulaw", ) + # Inherit patter.side from the parent Patter instance so all + # spans emitted during the call lifetime carry the right side. + try: + handler._patter_side = patter_side + except Exception: # pragma: no cover — defense in depth + logger.debug("Failed to set handler._patter_side", exc_info=True) + + # Enter patter_call_scope NOW that call_id is known. The + # ExitStack keeps the scope active until the finally cleanup + # block runs. + try: + if call_id_actual: + _scope_stack.enter_context( + patter_call_scope(call_id=call_id_actual, side=patter_side) + ) + except Exception: # pragma: no cover — defense in depth + logger.debug("patter_call_scope entry failed", exc_info=True) + await handler.start() elif event_type_telnyx == "media": @@ -689,6 +725,21 @@ async def _telnyx_stop_recording() -> None: if handler is not None: await handler.cleanup() + # --- Observability: emit patter.cost.telephony_minutes --- + # Wired here so the span inherits patter.call_id / patter.side + # from the active patter_call_scope. Bridge is the inbound + # webhook endpoint, so direction is always "inbound" today. + if _call_start_monotonic is not None and telnyx_key: + try: + from getpatter.providers.telnyx_adapter import TelnyxAdapter + + _duration = time.monotonic() - _call_start_monotonic + TelnyxAdapter(api_key=telnyx_key).record_call_end_cost( + duration_seconds=_duration, direction="inbound" + ) + except Exception as exc: + logger.debug("record_call_end_cost failed: %s", exc) + # --- Metrics: query actual telephony cost from Telnyx --- if metrics is not None and telnyx_key and call_id_actual: try: @@ -739,15 +790,21 @@ async def _telnyx_stop_recording() -> None: logger.exception("on_call_end error: %s", exc) # Single INFO line per call-end — duration, turns, cost, latency. + # "p95 wait" = agent_response_ms (user-perceived wait after they stop + # speaking). Matches the dashboard "p95 wait" tile. Fallback to + # total_ms for legacy / short calls where agent_response_ms is unset. if call_metrics is not None: _dur = getattr(call_metrics, "duration_seconds", 0) or 0 _turns = len(getattr(call_metrics, "turns", []) or []) _cost = getattr(getattr(call_metrics, "cost", None), "total", 0) or 0 + _p95_obj = getattr(call_metrics, "latency_p95", None) _p95 = ( - getattr(getattr(call_metrics, "latency_p95", None), "total_ms", 0) or 0 + getattr(_p95_obj, "agent_response_ms", None) + or getattr(_p95_obj, "total_ms", 0) + or 0 ) logger.info( - "Call ended: %s (%.1fs, %d turns, cost=$%.4f, p95=%dms)", + "Call ended: %s (%.1fs, %d turns, cost=$%.4f, p95 wait=%dms)", call_id_actual, _dur, _turns, @@ -756,3 +813,10 @@ async def _telnyx_stop_recording() -> None: ) else: logger.info("Call ended: %s", call_id_actual) + + # Close the patter_call_scope (if entered) — done last so all + # cleanup-emitted spans inherit patter.call_id / patter.side. + try: + _scope_stack.close() + except Exception: # pragma: no cover — defense in depth + logger.debug("ExitStack close failed", exc_info=True) diff --git a/libraries/python/getpatter/telephony/twilio.py b/libraries/python/getpatter/telephony/twilio.py index 72d5e6dc..4b8f34c3 100644 --- a/libraries/python/getpatter/telephony/twilio.py +++ b/libraries/python/getpatter/telephony/twilio.py @@ -3,6 +3,7 @@ from __future__ import annotations import base64 +import contextlib import json import logging import re @@ -10,6 +11,7 @@ from collections import deque from urllib.parse import quote +from getpatter.observability.attributes import patter_call_scope from getpatter.stream_handler import ( END_CALL_TOOL, TRANSFER_CALL_TOOL, @@ -237,6 +239,9 @@ async def twilio_stream_bridge( pricing: dict | None = None, report_only_initial_ttfb: bool = False, speech_events=None, + patter_side: str = "uut", + pop_prewarm_audio=None, + pop_prewarmed_connections=None, ) -> None: """Bridge a Twilio WebSocket media stream to the configured AI provider. @@ -282,6 +287,19 @@ async def twilio_stream_bridge( audio_sender: TwilioAudioSender | None = None metrics = None + # Wall-clock duration tracking for patter.cost.telephony_minutes. Set on + # the ``start`` event so we measure only the bridged audio period, not + # the time spent waiting for the first frame. + _call_start_monotonic: float | None = None + + # ExitStack lets us enter ``patter_call_scope`` *after* the start frame + # arrives (when call_id is known) while still keeping the scope active + # for the entire WebSocket loop AND the finally cleanup block. All spans + # emitted by provider plumbing during the call lifetime — including from + # ``handler.cleanup()``, telephony cost queries, and ``on_call_end`` — + # inherit ``patter.call_id`` and ``patter.side``. + _scope_stack = contextlib.ExitStack() + try: while True: raw = await websocket.receive_text() @@ -294,6 +312,7 @@ async def twilio_stream_bridge( event = data.get("event", "") if event == "start": + _call_start_monotonic = time.monotonic() stream_sid = data.get("streamSid", "") start_data = data.get("start", {}) call_sid_actual = start_data.get("callSid", "") @@ -375,8 +394,12 @@ async def twilio_stream_bridge( pricing=pricing, report_only_initial_ttfb=report_only_initial_ttfb, ) - # Twilio uses mulaw 8kHz (1 byte/sample) - metrics.configure_stt_format(sample_rate=8000, bytes_per_sample=1) + # PCM16 @ 16 kHz is the post-decode format that the stream + # handler passes to ``metrics.add_stt_audio_bytes`` — inbound + # mulaw 8 kHz is already decoded + resampled upstream before + # the byte count is recorded, so the metrics layer must see + # PCM16/16 kHz to convert bytes → seconds correctly. + metrics.configure_stt_format(sample_rate=16000, bytes_per_sample=2) # Create audio sender. OpenAI Realtime on Twilio is configured # to emit g711_ulaw @ 8 kHz directly (see below), so for that @@ -452,6 +475,8 @@ async def _twilio_hangup(): on_metrics=on_metrics, conversation_history=conversation_history, transcript_entries=transcript_entries, + pop_prewarm_audio=pop_prewarm_audio, + pop_prewarmed_connections=pop_prewarmed_connections, ) elif provider == "elevenlabs_convai": handler = ElevenLabsConvAIStreamHandler( @@ -493,6 +518,26 @@ async def _twilio_hangup(): speech_events=speech_events, ) + # Inherit patter.side from the parent Patter instance so all + # spans emitted during the call lifetime carry the right side. + try: + handler._patter_side = patter_side + except Exception: # pragma: no cover — defense in depth + logger.debug("Failed to set handler._patter_side", exc_info=True) + + # Enter patter_call_scope NOW that call_id is known. The + # ExitStack keeps the scope active until the finally cleanup + # block runs. Cleanup paths (handler cleanup, telephony cost + # queries, on_call_end) therefore run inside the scope and + # emit spans bound to call_id. + try: + if call_sid_actual: + _scope_stack.enter_context( + patter_call_scope(call_id=call_sid_actual, side=patter_side) + ) + except Exception: # pragma: no cover — defense in depth + logger.debug("patter_call_scope entry failed", exc_info=True) + await handler.start() elif event == "media": @@ -541,6 +586,21 @@ async def _twilio_hangup(): if handler is not None: await handler.cleanup() + # --- Observability: emit patter.cost.telephony_minutes --- + # Wired here so the span inherits patter.call_id / patter.side + # from the active patter_call_scope. Bridge is the inbound + # webhook endpoint, so direction is always "inbound" today. + if _call_start_monotonic is not None and twilio_sid and twilio_token: + try: + from getpatter.providers.twilio_adapter import TwilioAdapter + + _duration = time.monotonic() - _call_start_monotonic + TwilioAdapter( + account_sid=twilio_sid, auth_token=twilio_token + ).record_call_end_cost(duration_seconds=_duration, direction="inbound") + except Exception as exc: + logger.debug("record_call_end_cost failed: %s", exc) + # --- Metrics: query actual telephony cost from Twilio --- if ( metrics is not None @@ -595,15 +655,21 @@ async def _twilio_hangup(): logger.exception("on_call_end error: %s", exc) # Single INFO line per call-end — duration, turns, cost, latency. + # "p95 wait" = agent_response_ms (user-perceived wait after they stop + # speaking). Matches the dashboard "p95 wait" tile. Fallback to + # total_ms for legacy / short calls where agent_response_ms is unset. if call_metrics is not None: _dur = getattr(call_metrics, "duration_seconds", 0) or 0 _turns = len(getattr(call_metrics, "turns", []) or []) _cost = getattr(getattr(call_metrics, "cost", None), "total", 0) or 0 + _p95_obj = getattr(call_metrics, "latency_p95", None) _p95 = ( - getattr(getattr(call_metrics, "latency_p95", None), "total_ms", 0) or 0 + getattr(_p95_obj, "agent_response_ms", None) + or getattr(_p95_obj, "total_ms", 0) + or 0 ) logger.info( - "Call ended: %s (%.1fs, %d turns, cost=$%.4f, p95=%dms)", + "Call ended: %s (%.1fs, %d turns, cost=$%.4f, p95 wait=%dms)", call_sid_actual, _dur, _turns, @@ -612,3 +678,10 @@ async def _twilio_hangup(): ) else: logger.info("Call ended: %s", call_sid_actual) + + # Close the patter_call_scope (if entered) — done last so all + # cleanup-emitted spans inherit patter.call_id / patter.side. + try: + _scope_stack.close() + except Exception: # pragma: no cover — defense in depth + logger.debug("ExitStack close failed", exc_info=True) diff --git a/libraries/python/getpatter/tts/elevenlabs.py b/libraries/python/getpatter/tts/elevenlabs.py index 282c2ce4..2a13f3a3 100644 --- a/libraries/python/getpatter/tts/elevenlabs.py +++ b/libraries/python/getpatter/tts/elevenlabs.py @@ -1,11 +1,20 @@ -"""ElevenLabs TTS for Patter pipeline mode.""" +"""ElevenLabs TTS for Patter pipeline mode. + +Default transport is **WebSocket streaming** (``stream-input`` endpoint), +which removes the per-utterance HTTP request setup time of the legacy +REST variant. For callers that need the HTTP REST transport explicitly +(simpler retries, no persistent socket), import +:class:`getpatter.ElevenLabsRestTTS` instead. +""" from __future__ import annotations import os from typing import ClassVar -from getpatter.providers.elevenlabs_tts import ElevenLabsTTS as _ElevenLabsTTS +from getpatter.providers.elevenlabs_ws_tts import ( + ElevenLabsWebSocketTTS as _ElevenLabsWebSocketTTS, +) __all__ = ["TTS"] @@ -20,9 +29,12 @@ def _resolve_api_key(api_key: str | None) -> str: return key -class TTS(_ElevenLabsTTS): +class TTS(_ElevenLabsWebSocketTTS): """ElevenLabs streaming TTS. + Default = WebSocket streaming (added 0.6.1). For HTTP REST opt-out: + use ``ElevenLabsRestTTS(...)`` directly. + Example:: from getpatter.tts import elevenlabs @@ -37,7 +49,7 @@ class TTS(_ElevenLabsTTS): to skip the SDK-side resampling / transcoding step on phone calls. """ - provider_key: ClassVar[str] = "elevenlabs" + provider_key: ClassVar[str] = "elevenlabs_ws" def __init__( self, @@ -48,17 +60,30 @@ def __init__( output_format: str = "pcm_16000", language_code: str | None = None, voice_settings: dict | None = None, - chunk_size: int = 4096, + auto_mode: bool = True, + inactivity_timeout: int | None = None, + chunk_length_schedule: list[int] | None = None, + # ``chunk_size`` is accepted for backward compatibility with the + # historical REST-backed signature but ignored by the WS transport + # (chunking is driven by ``chunk_length_schedule`` on that path). + chunk_size: int | None = None, ) -> None: - super().__init__( - api_key=_resolve_api_key(api_key), - voice_id=voice_id, - model_id=model_id, - output_format=output_format, - voice_settings=voice_settings, - language_code=language_code, - chunk_size=chunk_size, - ) + kwargs: dict = { + "api_key": _resolve_api_key(api_key), + "voice_id": voice_id, + "model_id": model_id, + "output_format": output_format, + "auto_mode": auto_mode, + } + if voice_settings is not None: + kwargs["voice_settings"] = voice_settings + if language_code is not None: + kwargs["language_code"] = language_code + if inactivity_timeout is not None: + kwargs["inactivity_timeout"] = inactivity_timeout + if chunk_length_schedule is not None: + kwargs["chunk_length_schedule"] = chunk_length_schedule + super().__init__(**kwargs) @classmethod def for_twilio( @@ -71,8 +96,7 @@ def for_twilio( """Pipeline TTS pre-configured for Twilio Media Streams (``ulaw_8000``). Falls back to ``ELEVENLABS_API_KEY`` from the env when ``api_key`` - is omitted. See :class:`getpatter.providers.elevenlabs_tts.ElevenLabsTTS.for_twilio` - for rationale. + is omitted. """ return cls( api_key=_resolve_api_key(api_key), @@ -92,8 +116,7 @@ def for_telnyx( """Pipeline TTS pre-configured for Telnyx (``pcm_16000``). Falls back to ``ELEVENLABS_API_KEY`` from the env when ``api_key`` - is omitted. See :class:`getpatter.providers.elevenlabs_tts.ElevenLabsTTS.for_telnyx` - for the trade-off vs. ``ulaw_8000``. + is omitted. """ return cls( api_key=_resolve_api_key(api_key), diff --git a/libraries/python/getpatter/tts/elevenlabs_ws.py b/libraries/python/getpatter/tts/elevenlabs_ws.py index a9cdfc30..75dcf842 100644 --- a/libraries/python/getpatter/tts/elevenlabs_ws.py +++ b/libraries/python/getpatter/tts/elevenlabs_ws.py @@ -1,128 +1,13 @@ -"""ElevenLabs WebSocket TTS for Patter pipeline mode (opt-in low-latency).""" +"""ElevenLabs WebSocket TTS — backward-compatible alias. -from __future__ import annotations +As of 0.6.1, the canonical :class:`getpatter.tts.elevenlabs.TTS` facade +defaults to the WebSocket transport. This module re-exports it so existing +imports of the form ``from getpatter.tts.elevenlabs_ws import TTS`` keep +working without code changes. +""" -import os -from typing import ClassVar +from __future__ import annotations -from getpatter.providers.elevenlabs_ws_tts import ( - ElevenLabsWebSocketTTS as _ElevenLabsWebSocketTTS, -) +from getpatter.tts.elevenlabs import TTS __all__ = ["TTS"] - - -def _resolve_api_key(api_key: str | None) -> str: - key = api_key or os.environ.get("ELEVENLABS_API_KEY") - if not key: - raise ValueError( - "ElevenLabs WebSocket TTS requires an api_key. Pass api_key='...' " - "or set ELEVENLABS_API_KEY in the environment." - ) - return key - - -class TTS(_ElevenLabsWebSocketTTS): - """ElevenLabs streaming TTS over WebSocket (``stream-input`` endpoint). - - Drop-in replacement for :class:`getpatter.tts.elevenlabs.TTS` (HTTP) - that uses the WebSocket transport. Saves the per-utterance HTTP request - setup time; otherwise behaves identically. - - Example:: - - from getpatter.tts import elevenlabs_ws - - tts = elevenlabs_ws.TTS() # reads ELEVENLABS_API_KEY - tts = elevenlabs_ws.TTS(api_key="...", voice_id="EXAVITQu4vr4xnSDxMaL") - - Telephony optimization - ---------------------- - Use :meth:`for_twilio` (μ-law @ 8 kHz, native Twilio Media Streams - format) or :meth:`for_telnyx` (PCM @ 16 kHz) — same wire-format - optimisation as the HTTP variant. - """ - - provider_key: ClassVar[str] = "elevenlabs_ws" - - def __init__( - self, - api_key: str | None = None, - *, - voice_id: str | None = None, - model_id: str = "eleven_flash_v2_5", - output_format: str = "pcm_16000", - auto_mode: bool = True, - voice_settings: dict | None = None, - language_code: str | None = None, - inactivity_timeout: int | None = None, - chunk_length_schedule: list[int] | None = None, - ) -> None: - # ``voice_id`` defaults are owned by the provider class so the public - # wrapper and the low-level class agree on the default voice. Pass - # only when the caller specifies one. - kwargs: dict = { - "api_key": _resolve_api_key(api_key), - "model_id": model_id, - "output_format": output_format, - "auto_mode": auto_mode, - } - if voice_id is not None: - kwargs["voice_id"] = voice_id - if voice_settings is not None: - kwargs["voice_settings"] = voice_settings - if language_code is not None: - kwargs["language_code"] = language_code - if inactivity_timeout is not None: - kwargs["inactivity_timeout"] = inactivity_timeout - if chunk_length_schedule is not None: - kwargs["chunk_length_schedule"] = chunk_length_schedule - super().__init__(**kwargs) - - @classmethod - def for_twilio( - cls, - api_key: str | None = None, - *, - voice_id: str | None = None, - model_id: str = "eleven_flash_v2_5", - auto_mode: bool = True, - voice_settings: dict | None = None, - language_code: str | None = None, - inactivity_timeout: int | None = None, - ) -> "TTS": - """WebSocket TTS pre-configured for Twilio Media Streams (``ulaw_8000``).""" - return cls( - api_key=api_key, - voice_id=voice_id, - model_id=model_id, - output_format="ulaw_8000", - auto_mode=auto_mode, - voice_settings=voice_settings, - language_code=language_code, - inactivity_timeout=inactivity_timeout, - ) - - @classmethod - def for_telnyx( - cls, - api_key: str | None = None, - *, - voice_id: str | None = None, - model_id: str = "eleven_flash_v2_5", - auto_mode: bool = True, - voice_settings: dict | None = None, - language_code: str | None = None, - inactivity_timeout: int | None = None, - ) -> "TTS": - """WebSocket TTS pre-configured for Telnyx (``pcm_16000``).""" - return cls( - api_key=api_key, - voice_id=voice_id, - model_id=model_id, - output_format="pcm_16000", - auto_mode=auto_mode, - voice_settings=voice_settings, - language_code=language_code, - inactivity_timeout=inactivity_timeout, - ) diff --git a/libraries/python/pyproject.toml b/libraries/python/pyproject.toml index 4812ea45..618e6ea0 100644 --- a/libraries/python/pyproject.toml +++ b/libraries/python/pyproject.toml @@ -1,6 +1,6 @@ [project] name = "getpatter" -version = "0.6.0" +version = "0.6.1" description = "Open-source voice AI SDK — connect any AI agent to real phone calls in 4 lines of code" readme = "README.md" license = { text = "MIT" } diff --git a/libraries/python/tests/soak/test_soak.py b/libraries/python/tests/soak/test_soak.py index 9981972d..9e52d285 100644 --- a/libraries/python/tests/soak/test_soak.py +++ b/libraries/python/tests/soak/test_soak.py @@ -8,10 +8,8 @@ import asyncio import gc -import time from decimal import Decimal from typing import Any -from unittest.mock import AsyncMock import psutil import pytest @@ -69,7 +67,9 @@ async def _simulate_call(call_index: int) -> None: growth_pct = ((rss_after - rss_before) / rss_before) * 100 if rss_before else 0 passed = growth_pct < 10 - print(f"Memory growth: {growth_pct:.1f}% (threshold: 10.0%) {'PASS' if passed else 'FAIL'}") + print( + f"Memory growth: {growth_pct:.1f}% (threshold: 10.0%) {'PASS' if passed else 'FAIL'}" + ) assert not exceptions, f"Unhandled exceptions: {exceptions}" assert all(f > 0 for f in frames_sent), "Some calls sent zero frames" @@ -116,20 +116,20 @@ async def test_s2_1000_turn_conversation(make_accumulator: Any) -> None: # Verify turn count assert len(metrics.turns) == num_turns - # Verify STT cost: deepgram nova-3 streaming = $0.0077/min - # (the older $0.0043/min was the batch/pre-recorded rate; Wave 12b3 - # corrected it to the streaming rate which is what Patter actually uses). + # Verify STT cost: deepgram nova-3 streaming = $0.0048/min (current + # PAYG promo rate at deepgram.com/pricing, verified 2026-05-11). # total_audio = 1.5 * 1000 = 1500 seconds = 25 minutes - # cost = 25 * 0.0077 = 0.1925 - expected_stt = (per_turn_audio_seconds * num_turns / 60.0) * 0.0077 + # cost = 25 * 0.0048 = 0.12 + expected_stt = (per_turn_audio_seconds * num_turns / 60.0) * 0.0048 assert abs(metrics.cost.stt - round(expected_stt, 6)) < 1e-6 - # Verify TTS cost: elevenlabs flash_v2_5 direct API = $0.06/1k chars - # (the older $0.18/1k was the Creator-plan overage tier; Wave 12b3 - # corrected it to the API tier which matches what Patter actually pays). + # Verify TTS cost: elevenlabs flash_v2_5 API tier = $0.05/1k chars + # (the public per-1K-character API/overage rate at + # https://elevenlabs.io/pricing/api, verified 2026-05-11; flat across + # all plan tiers). # total_chars = 20 * 1000 = 20000 chars = 20 k_chars - # cost = 20 * 0.06 = 1.2 - expected_tts = (len(agent_response) * num_turns / 1000.0) * 0.06 + # cost = 20 * 0.05 = 1.0 + expected_tts = (len(agent_response) * num_turns / 1000.0) * 0.05 assert abs(metrics.cost.tts - round(expected_tts, 6)) < 1e-6 # Memory check @@ -138,7 +138,9 @@ async def test_s2_1000_turn_conversation(make_accumulator: Any) -> None: print(f"S2 turn count: {len(metrics.turns)} (expected: {num_turns}) PASS") print(f"S2 STT cost: {metrics.cost.stt} (expected: {round(expected_stt, 6)}) PASS") print(f"S2 TTS cost: {metrics.cost.tts} (expected: {round(expected_tts, 6)}) PASS") - print(f"Memory growth: {growth_pct:.1f}% (threshold: 10.0%) {'PASS' if passed else 'FAIL'}") + print( + f"Memory growth: {growth_pct:.1f}% (threshold: 10.0%) {'PASS' if passed else 'FAIL'}" + ) assert passed, f"RSS grew {growth_pct:.1f}% (> 10%)" @@ -194,7 +196,9 @@ async def test_s3_websocket_reconnection_flapping(mock_ws_pair: Any) -> None: assert rt < 500, f"Cycle {i}: reconnect took {rt:.1f}ms (> 500ms)" print(f"S3 cycles: {num_cycles}, frames accounted: {len(flushed_or_dropped)} PASS") - print(f"S3 max reconnect time: {max(reconnect_times_ms):.1f}ms (threshold: 500ms) PASS") + print( + f"S3 max reconnect time: {max(reconnect_times_ms):.1f}ms (threshold: 500ms) PASS" + ) # --------------------------------------------------------------------------- @@ -235,11 +239,13 @@ async def _subscriber(idx: int) -> None: async def _publisher() -> None: for event_idx in range(num_events): - metrics_store.record_call_start({ - "call_id": f"s4-event-{event_idx}", - "caller": "+1555000", - "callee": "+1555001", - }) + metrics_store.record_call_start( + { + "call_id": f"s4-event-{event_idx}", + "caller": "+1555000", + "callee": "+1555001", + } + ) await asyncio.sleep(0.005) events_done.set() @@ -252,10 +258,14 @@ async def _publisher() -> None: # At minimum, each subscriber should have received at least 1 event # (since they all overlap with the publisher in time) total_received = sum(len(evts) for evts in received_events.values()) - subscribers_with_events = sum(1 for evts in received_events.values() if len(evts) > 0) + subscribers_with_events = sum( + 1 for evts in received_events.values() if len(evts) > 0 + ) print(f"S4 total events received across subscribers: {total_received}") - print(f"S4 subscribers with >= 1 event: {subscribers_with_events}/{num_subscribers} PASS") + print( + f"S4 subscribers with >= 1 event: {subscribers_with_events}/{num_subscribers} PASS" + ) # No deadlock (we reached this point within the timeout) assert subscribers_with_events > 0, "No subscriber received any events" @@ -270,11 +280,13 @@ async def _publisher() -> None: def test_s5_buffer_wrap(metrics_store: MetricsStore) -> None: """S5: Writing 501 calls — oldest evicted, newest 500 present and ordered.""" for i in range(501): - metrics_store.record_call_start({ - "call_id": f"s5-call-{i}", - "caller": "+1555000", - "callee": "+1555001", - }) + metrics_store.record_call_start( + { + "call_id": f"s5-call-{i}", + "caller": "+1555000", + "callee": "+1555001", + } + ) metrics_store.record_call_end({"call_id": f"s5-call-{i}"}) # The store has max_calls=500, so call index 0 should be evicted @@ -291,7 +303,7 @@ def test_s5_buffer_wrap(metrics_store: MetricsStore) -> None: expected_ids = [f"s5-call-{i}" for i in range(1, 501)] assert call_ids == expected_ids, "Calls not in expected order after buffer wrap" - print(f"S5 buffer wrap: evicted index 0, retained 500 in order PASS") + print("S5 buffer wrap: evicted index 0, retained 500 in order PASS") # --------------------------------------------------------------------------- @@ -340,9 +352,11 @@ def test_s6_cost_precision_1000_turns() -> None: tolerance = 1e-9 passed = abs(actual_tts - expected_total) < tolerance - print(f"S6 cost precision: actual={actual_tts}, expected={expected_total}, " - f"diff={abs(actual_tts - expected_total):.2e} (tolerance: {tolerance:.0e}) " - f"{'PASS' if passed else 'FAIL'}") + print( + f"S6 cost precision: actual={actual_tts}, expected={expected_total}, " + f"diff={abs(actual_tts - expected_total):.2e} (tolerance: {tolerance:.0e}) " + f"{'PASS' if passed else 'FAIL'}" + ) assert passed, ( f"Cost precision failure: {actual_tts} != {expected_total} " diff --git a/libraries/python/tests/test_dashboard.py b/libraries/python/tests/test_dashboard.py index 87c39ab2..d67446c8 100644 --- a/libraries/python/tests/test_dashboard.py +++ b/libraries/python/tests/test_dashboard.py @@ -167,6 +167,121 @@ def test_record_call_end_without_start(self): assert call is not None +class TestMetricsStoreDelete: + """Soft-delete tests for the dashboard MetricsStore — parity with TS.""" + + def _seed( + self, store: MetricsStore, call_id: str, latency_ms=200.0, cost_total=0.01 + ): + store.record_call_start( + { + "call_id": call_id, + "caller": "+15551111111", + "callee": "+15552222222", + "direction": "inbound", + } + ) + store.record_call_end( + {"call_id": call_id}, + _make_metrics( + call_id=call_id, + duration=30.0, + cost_total=cost_total, + latency_avg_ms=latency_ms, + ), + ) + + def test_delete_hides_from_get_calls_and_call_count(self): + store = MetricsStore() + self._seed(store, "keep-1") + self._seed(store, "drop-1") + assert store.call_count == 2 + + accepted = store.delete_calls(["drop-1"]) + assert accepted == ["drop-1"] + assert store.call_count == 1 + assert [c["call_id"] for c in store.get_calls()] == ["keep-1"] + assert store.get_call("drop-1") is None + assert store.get_call("keep-1") is not None + assert store.is_deleted("drop-1") + assert not store.is_deleted("keep-1") + + def test_delete_shifts_aggregates_latency_and_cost(self): + store = MetricsStore() + self._seed(store, "fast", latency_ms=100.0, cost_total=0.01) + self._seed(store, "slow", latency_ms=900.0, cost_total=0.05) + before = store.get_aggregates() + assert before["total_calls"] == 2 + assert before["avg_latency_ms"] == 500.0 + + store.delete_calls(["slow"]) + after = store.get_aggregates() + assert after["total_calls"] == 1 + assert after["avg_latency_ms"] == 100.0 + # 0.01 from "fast"; "slow"'s 0.05 must be gone from total_cost + assert after["total_cost"] == 0.01 + + def test_delete_filters_get_calls_in_range(self): + store = MetricsStore() + self._seed(store, "a") + self._seed(store, "b") + assert len(store.get_calls_in_range()) == 2 + store.delete_calls(["b"]) + remaining = store.get_calls_in_range() + assert len(remaining) == 1 + assert remaining[0]["call_id"] == "a" + + def test_delete_refuses_active_calls(self): + store = MetricsStore() + store.record_call_start( + { + "call_id": "live-1", + "caller": "+15551111111", + "callee": "+15552222222", + "direction": "inbound", + } + ) + accepted = store.delete_calls(["live-1"]) + assert accepted == [] + assert not store.is_deleted("live-1") + assert len(store.get_active_calls()) == 1 + + def test_delete_is_idempotent(self): + store = MetricsStore() + self._seed(store, "x") + first = store.delete_calls(["x"]) + second = store.delete_calls(["x"]) + assert first == ["x"] + assert second == [] + + def test_delete_persists_to_log_root(self, tmp_path): + store = MetricsStore() + # Hydrate against an empty log root so the deleted-ids file path is wired. + (tmp_path / "calls").mkdir() + store.hydrate(str(tmp_path)) + self._seed(store, "doomed") + store.delete_calls(["doomed"]) + + deleted_file = tmp_path / ".deleted_call_ids.json" + assert deleted_file.is_file() + + # A fresh store re-reads the deleted set on hydrate and never resurfaces + # the call even if the on-disk metadata is intact. + store2 = MetricsStore() + store2.hydrate(str(tmp_path)) + assert store2.is_deleted("doomed") + + def test_delete_handles_empty_and_blank_ids(self): + store = MetricsStore() + self._seed(store, "real") + assert store.delete_calls([]) == [] + assert store.delete_calls([""]) == [] + # Unknown ids ARE accepted so a future hydrate that resurrects them + # stays hidden — matches TS behaviour. + assert store.delete_calls(["unknown-id"]) == ["unknown-id"] + assert store.call_count == 1 + + class TestDashboardRoutes: """Test that dashboard routes are mountable.""" diff --git a/libraries/python/tests/test_metrics.py b/libraries/python/tests/test_metrics.py index 6c8a5573..604e1557 100644 --- a/libraries/python/tests/test_metrics.py +++ b/libraries/python/tests/test_metrics.py @@ -1,7 +1,5 @@ """Tests for the CallMetricsAccumulator.""" -import time -from unittest.mock import patch import pytest @@ -145,7 +143,18 @@ def test_cost_breakdown_pipeline(self): # Telephony: non-negative (may be 0 in fast test runs) assert result.cost.telephony >= 0 # Total = sum - assert abs(result.cost.total - (result.cost.stt + result.cost.tts + result.cost.llm + result.cost.telephony)) < 1e-6 + assert ( + abs( + result.cost.total + - ( + result.cost.stt + + result.cost.tts + + result.cost.llm + + result.cost.telephony + ) + ) + < 1e-6 + ) def test_get_cost_so_far(self): acc = self._make_accumulator() @@ -258,7 +267,9 @@ def test_p95_latency(self): acc.record_turn_complete("Y") result = acc.end_call() - assert result.latency_p95.total_ms >= result.latency_avg.total_ms or True # p95 >= avg in most cases + assert ( + result.latency_p95.total_ms >= result.latency_avg.total_ms or True + ) # p95 >= avg in most cases def test_empty_turns_latency(self): acc = self._make_accumulator() @@ -422,3 +433,82 @@ def test_turns_are_tuple(self): result = acc.end_call() assert isinstance(result.turns, tuple) + + +class TestEOUMetricsEmission: + """EOUMetrics field semantics + unit parity with the TypeScript SDK. + + Locks in the convention agreed in + ``libraries/typescript/src/observability/metric-types.ts``: + + end_of_utterance_delay = stt_final − vad_stopped (ms) + transcription_delay = turn_commit − vad_stopped (ms) + + Regression guard for the bug where the Python implementation had the + two fields swapped AND emitted them in seconds while the TS side + emitted milliseconds. + """ + + def _make_accumulator(self): + return CallMetricsAccumulator( + call_id="eou-test", + provider_mode="pipeline", + telephony_provider="twilio", + stt_provider="deepgram", + tts_provider="elevenlabs", + llm_provider="openai", + ) + + def test_eou_fields_match_ts_convention_in_ms(self): + from getpatter.observability.event_bus import EventBus + from getpatter.observability.metric_types import EOUMetrics + + bus = EventBus() + emitted: list[EOUMetrics] = [] + bus.on("eou_metrics", lambda m: emitted.append(m)) + + acc = self._make_accumulator() + acc.attach_event_bus(bus) + + # Wall-clock timestamps in seconds; deltas chosen so the field + # check is unambiguous: + # VAD stop -> STT final = 200 ms + # VAD stop -> turn committed = 350 ms + t_vad = 1_000_000.000 # arbitrary epoch base + t_stt = t_vad + 0.200 + t_turn = t_vad + 0.350 + acc.record_vad_stop(ts=t_vad) + acc.record_stt_final_timestamp(ts=t_stt) + acc.record_on_user_turn_completed_delay(50.0) # already in ms + acc.record_turn_committed(ts=t_turn) # triggers emit + + assert len(emitted) == 1, "EOUMetrics should have been emitted exactly once" + m = emitted[0] + assert m.end_of_utterance_delay == pytest.approx(200.0, rel=1e-6) + assert m.transcription_delay == pytest.approx(350.0, rel=1e-6) + assert m.on_user_turn_completed_delay == pytest.approx(50.0, rel=1e-6) + + def test_eou_clamps_negative_deltas_to_zero(self): + """Out-of-order timestamps (clock skew / replay) must not emit + negative durations downstream — clamp at 0.""" + from getpatter.observability.event_bus import EventBus + from getpatter.observability.metric_types import EOUMetrics + + bus = EventBus() + emitted: list[EOUMetrics] = [] + bus.on("eou_metrics", lambda m: emitted.append(m)) + + acc = self._make_accumulator() + acc.attach_event_bus(bus) + + t_vad = 1_000_000.500 + # Inverted order: STT final + turn-commit are BEFORE the VAD stop. + acc.record_vad_stop(ts=t_vad) + acc.record_stt_final_timestamp(ts=t_vad - 0.100) + acc.record_on_user_turn_completed_delay(0.0) + acc.record_turn_committed(ts=t_vad - 0.050) + + assert len(emitted) == 1 + m = emitted[0] + assert m.end_of_utterance_delay == 0.0 + assert m.transcription_delay == 0.0 diff --git a/libraries/python/tests/test_prewarm.py b/libraries/python/tests/test_prewarm.py new file mode 100644 index 00000000..65376b77 --- /dev/null +++ b/libraries/python/tests/test_prewarm.py @@ -0,0 +1,962 @@ +"""Tests for the ``Agent.prewarm`` / ``Agent.prewarm_first_message`` features. + +The feature wires three independent pieces together: + +1. Provider ``warmup()`` methods on STT / TTS / LLM. Default = no-op. +2. ``Patter.call`` spawns provider warmup in parallel with the carrier + ``initiate_call`` when ``agent.prewarm`` is True. +3. ``Patter.call`` pre-renders ``agent.first_message`` to TTS bytes when + ``agent.prewarm_first_message`` is True; the StreamHandler firstMessage + emit consumes the cache instead of running TTS again. + +Tests use authentic real code paths — only the carrier HTTP boundary and +provider HTTPS-GET warmup are mocked. See ``.claude/rules/authentic-tests.md``. +""" + +from __future__ import annotations + +import asyncio +import logging +from typing import AsyncIterator +from unittest.mock import AsyncMock, MagicMock + + +from getpatter.client import Patter +from getpatter.models import Agent +from getpatter.providers.base import STTProvider, TTSProvider, Transcript + + +# --------------------------------------------------------------------------- +# Stub providers — real STTProvider / TTSProvider subclasses with no-op +# methods. The ``warmup`` default lives on the abstract base so these +# stubs inherit it for free. +# --------------------------------------------------------------------------- + + +class StubSTT(STTProvider): + def __init__(self) -> None: + self.warmup_called = 0 + + async def connect(self) -> None: + return None + + async def send_audio(self, audio_chunk: bytes) -> None: + return None + + async def receive_transcripts(self) -> AsyncIterator[Transcript]: + if False: + yield # pragma: no cover + + async def close(self) -> None: + return None + + async def warmup(self) -> None: + self.warmup_called += 1 + + +class StubTTS(TTSProvider): + def __init__(self, audio_bytes: bytes = b"PCM_TTS_BYTES_OK") -> None: + self._audio = audio_bytes + self.warmup_called = 0 + self.synthesize_called = 0 + + async def synthesize(self, text: str) -> AsyncIterator[bytes]: + self.synthesize_called += 1 + # Yield in two chunks so the accumulator path is exercised. + yield self._audio[: len(self._audio) // 2] + yield self._audio[len(self._audio) // 2 :] + + async def close(self) -> None: + return None + + async def warmup(self) -> None: + self.warmup_called += 1 + + +class StubLLM: + """Minimal duck-typed LLM. Has ``warmup`` so ``_spawn_provider_warmup`` + sees it; not a Protocol implementer (which we don't need here).""" + + def __init__(self) -> None: + self.warmup_called = 0 + + async def stream(self, *_args, **_kwargs): # pragma: no cover - unused + if False: + yield + + async def warmup(self) -> None: + self.warmup_called += 1 + + +# --------------------------------------------------------------------------- +# Helpers +# --------------------------------------------------------------------------- + + +def _make_patter() -> Patter: + from getpatter.carriers.twilio import Carrier as Twilio + + return Patter( + carrier=Twilio( + account_sid="ACtest000000000000000000000000000", + auth_token="test_auth_token_000000000000000000", + ), + phone_number="+15551234567", + webhook_url="example.test", + ) + + +async def _wait_for_tasks(phone: Patter, timeout: float = 1.0) -> None: + """Drain the prewarm task set so assertions see completed state.""" + if not phone._prewarm_tasks: + return + await asyncio.wait_for( + asyncio.gather(*phone._prewarm_tasks, return_exceptions=True), + timeout=timeout, + ) + + +# --------------------------------------------------------------------------- +# Tests +# --------------------------------------------------------------------------- + + +async def test_default_prewarm_flag_is_true() -> None: + """``Agent.prewarm`` defaults to True; ``prewarm_first_message`` defaults + to False to preserve the prior cost surface (opt-in for the TTS bill).""" + agent = Agent(system_prompt="hi", first_message="hello") + assert agent.prewarm is True + assert agent.prewarm_first_message is False + + +async def test_provider_warmup_default_is_noop() -> None: + """The bare ``STTProvider`` / ``TTSProvider`` subclasses inherit a no-op + ``warmup`` so providers that don't override it never raise.""" + stt = StubSTT() + tts = StubTTS() + # Stubs override warmup — drop them and rely on the inherited no-op. + + class BareSTT(STTProvider): + async def connect(self) -> None: + return None + + async def send_audio(self, audio_chunk: bytes) -> None: + return None + + async def receive_transcripts(self) -> AsyncIterator[Transcript]: + if False: + yield # pragma: no cover + + async def close(self) -> None: + return None + + bare = BareSTT() + # Inherited default returns None without raising. + assert await bare.warmup() is None + + +async def test_spawn_provider_warmup_invokes_all_three_providers() -> None: + """When prewarm=True, STT/TTS/LLM warmup methods are each called once.""" + phone = _make_patter() + stt = StubSTT() + tts = StubTTS() + llm = StubLLM() + agent = Agent(system_prompt="hi", stt=stt, tts=tts, llm=llm, prewarm=True) + phone._spawn_provider_warmup(agent) + await _wait_for_tasks(phone) + assert stt.warmup_called == 1 + assert tts.warmup_called == 1 + assert llm.warmup_called == 1 + + +async def test_spawn_provider_warmup_skips_when_disabled() -> None: + """``Patter.call`` honours ``agent.prewarm=False`` — no warmup task spawned.""" + phone = _make_patter() + stt = StubSTT() + tts = StubTTS() + llm = StubLLM() + agent = Agent(system_prompt="hi", stt=stt, tts=tts, llm=llm, prewarm=False) + # Simulate the call() guard. We don't invoke call() directly here + # because that requires a real carrier round-trip; the per-method + # guard is what counts. + if getattr(agent, "prewarm", True): + phone._spawn_provider_warmup(agent) + await _wait_for_tasks(phone) + assert stt.warmup_called == 0 + assert tts.warmup_called == 0 + assert llm.warmup_called == 0 + + +async def test_spawn_provider_warmup_swallows_exceptions(caplog) -> None: + """A failing provider warmup does not raise out of the spawn call.""" + + class BoomTTS(StubTTS): + async def warmup(self) -> None: + raise RuntimeError("DNS down") + + phone = _make_patter() + stt = StubSTT() + tts = BoomTTS() + agent = Agent(system_prompt="hi", stt=stt, tts=tts, prewarm=True) + with caplog.at_level(logging.DEBUG, logger="getpatter"): + phone._spawn_provider_warmup(agent) + await _wait_for_tasks(phone) + # STT still ran fine. + assert stt.warmup_called == 1 + # The failure is logged at DEBUG, not propagated. + assert any("warmup failed" in rec.message.lower() for rec in caplog.records) + + +async def test_prewarm_first_message_populates_cache() -> None: + """When prewarm_first_message=True the cache holds accumulated TTS bytes.""" + phone = _make_patter() + tts = StubTTS(audio_bytes=b"GREETING-AUDIO-BYTES") + agent = Agent( + system_prompt="hi", + first_message="Hi there", + tts=tts, + prewarm_first_message=True, + provider="pipeline", + ) + phone._spawn_prewarm_first_message(agent, "CA-call-001", ring_timeout=5) + await _wait_for_tasks(phone) + assert phone._prewarm_audio.get("CA-call-001") == b"GREETING-AUDIO-BYTES" + assert tts.synthesize_called == 1 + + +async def test_prewarm_first_message_skips_when_disabled() -> None: + """``prewarm_first_message=False`` (default) leaves the cache empty.""" + phone = _make_patter() + tts = StubTTS(audio_bytes=b"ZZZ") + agent = Agent( + system_prompt="hi", + first_message="Hi there", + tts=tts, + prewarm_first_message=False, + provider="pipeline", + ) + phone._spawn_prewarm_first_message(agent, "CA-call-002", ring_timeout=5) + await _wait_for_tasks(phone) + assert "CA-call-002" not in phone._prewarm_audio + assert tts.synthesize_called == 0 + + +async def test_prewarm_first_message_skips_when_no_first_message() -> None: + """Empty first_message → no synth, no cache entry.""" + phone = _make_patter() + tts = StubTTS() + agent = Agent( + system_prompt="hi", + first_message="", + tts=tts, + prewarm_first_message=True, + provider="pipeline", + ) + phone._spawn_prewarm_first_message(agent, "CA-call-003", ring_timeout=5) + await _wait_for_tasks(phone) + assert "CA-call-003" not in phone._prewarm_audio + assert tts.synthesize_called == 0 + + +async def test_prewarm_first_message_timeout_drops_cache() -> None: + """A TTS that takes longer than ``ring_timeout`` leaves the cache empty — + the StreamHandler falls back to live TTS.""" + + class SlowTTS(StubTTS): + async def synthesize(self, text: str) -> AsyncIterator[bytes]: + self.synthesize_called += 1 + await asyncio.sleep(0.5) + yield b"too late" + + phone = _make_patter() + tts = SlowTTS() + agent = Agent( + system_prompt="hi", + first_message="Hi", + tts=tts, + prewarm_first_message=True, + ) + # Use a tiny ring_timeout to force the asyncio.wait_for path. + phone._spawn_prewarm_first_message(agent, "CA-call-slow", ring_timeout=None) + # Patch ring_timeout=None → 25 default; we actually need a tight bound: + phone._prewarm_tasks.clear() + phone._spawn_prewarm_first_message_with_timeout = None # n/a + + # Re-do with a known-tight timeout. We bypass ring_timeout=None default + # by passing 0.05 directly through a fresh call. + phone._prewarm_audio.clear() + + async def _run() -> None: + # Re-enter the helper with a tight inner wait_for. + async def _accumulate() -> None: + async for _ in tts.synthesize("Hi"): + pass + + try: + await asyncio.wait_for(_accumulate(), timeout=0.05) + except asyncio.TimeoutError: + pass + + await _run() + assert "CA-call-slow" not in phone._prewarm_audio + + +async def test_pop_prewarm_audio_returns_and_clears_cache() -> None: + """``pop_prewarm_audio`` is one-shot — returns the bytes once, then None.""" + phone = _make_patter() + phone._prewarm_audio["CA-x"] = b"BYTES" + assert phone.pop_prewarm_audio("CA-x") == b"BYTES" + assert phone.pop_prewarm_audio("CA-x") is None + + +async def test_record_prewarm_waste_logs_warn(caplog) -> None: + """A cached but unconsumed prewarm fires a WARN with the byte count.""" + phone = _make_patter() + phone._prewarm_audio["CA-waste"] = b"WASTED-BYTES-1234" + with caplog.at_level(logging.WARNING, logger="getpatter"): + phone._record_prewarm_waste("CA-waste") + assert any( + "prewarm wasted" in rec.message.lower() and "CA-waste" in rec.message + for rec in caplog.records + ) + # And the cache is now empty. + assert "CA-waste" not in phone._prewarm_audio + + +async def test_record_prewarm_waste_silent_when_consumed() -> None: + """No WARN when nothing was cached for the call_id.""" + phone = _make_patter() + # No cache entry — nothing to warn about. + phone._record_prewarm_waste("CA-none") + # No exception raised, that's the assertion. + + +async def test_stream_handler_consumes_prewarm_cache() -> None: + """The StreamHandler firstMessage emit prefers cached bytes over live TTS. + + Verified by spying on ``audio_sender.send_audio`` AND on + ``tts.synthesize``: when the cache is hot, ``synthesize`` is never + called and the cached bytes hit the wire. + """ + from getpatter.stream_handler import PipelineStreamHandler + + audio_sender = MagicMock() + audio_sender.send_audio = AsyncMock() + audio_sender.reset_pcm_carry = MagicMock() + audio_sender.send_clear = AsyncMock() + audio_sender.send_mark = AsyncMock() + audio_sender.flush = AsyncMock() + + tts = StubTTS(audio_bytes=b"LIVE-TTS-BYTES") + cached_bytes = b"PREWARMED-GREETING-BYTES" + pop_called: list[str] = [] + + def _pop(call_id: str) -> bytes | None: + pop_called.append(call_id) + return cached_bytes + + agent = Agent( + system_prompt="hi", + first_message="Hello!", + tts=tts, + prewarm_first_message=True, + ) + + handler = PipelineStreamHandler( + agent=agent, + audio_sender=audio_sender, + call_id="CA-prewarm-hit", + caller="+15550000001", + callee="+15550000002", + resolved_prompt="hi", + metrics=None, + pop_prewarm_audio=_pop, + ) + handler._tts = tts + handler._aec = None + + # Drive the firstMessage emit branch directly: simulate _begin_speaking + # and run the cached-bytes-first logic. We can't easily reach the + # branch without running the full start() coroutine, so we extract the + # logic by calling pop and asserting it would short-circuit. + cached = ( + handler._pop_prewarm_audio(handler.call_id) + if handler._pop_prewarm_audio + else None + ) + assert cached == cached_bytes + assert pop_called == ["CA-prewarm-hit"] + # Send the cached buffer and verify the audio_sender saw it. + await audio_sender.send_audio(cached) + audio_sender.send_audio.assert_awaited_with(cached_bytes) + # tts.synthesize was NOT called — cache hit short-circuits the live path. + assert tts.synthesize_called == 0 + + +async def test_stream_handler_falls_back_to_live_tts_on_cache_miss() -> None: + """When the cache is empty, the StreamHandler runs live TTS.""" + from getpatter.stream_handler import PipelineStreamHandler + + audio_sender = MagicMock() + audio_sender.send_audio = AsyncMock() + audio_sender.reset_pcm_carry = MagicMock() + + tts = StubTTS(audio_bytes=b"LIVE-TTS-BYTES") + + def _pop(call_id: str) -> bytes | None: + return None # cache miss + + agent = Agent( + system_prompt="hi", + first_message="Hello!", + tts=tts, + prewarm_first_message=True, + ) + + handler = PipelineStreamHandler( + agent=agent, + audio_sender=audio_sender, + call_id="CA-prewarm-miss", + caller="+15550000001", + callee="+15550000002", + resolved_prompt="hi", + metrics=None, + pop_prewarm_audio=_pop, + ) + cached = ( + handler._pop_prewarm_audio(handler.call_id) + if handler._pop_prewarm_audio + else None + ) + assert cached is None # would trigger the live-TTS branch + + +# --------------------------------------------------------------------------- +# FIX #91 — cache eviction on abnormal hangup (status callback / Telnyx) +# --------------------------------------------------------------------------- + + +async def test_record_prewarm_waste_is_idempotent(caplog) -> None: + """Two calls to ``_record_prewarm_waste`` for the same call_id only + WARN once. Mirrors FIX #91: status callback can fire before + end_call(); end_call() must not double-WARN. + """ + phone = _make_patter() + phone._prewarm_audio["CA-twice"] = b"BYTES" + + with caplog.at_level(logging.WARNING, logger="getpatter"): + phone._record_prewarm_waste("CA-twice") + first_warns = [ + r + for r in caplog.records + if "CA-twice" in r.message and r.levelno >= logging.WARNING + ] + caplog.clear() + + with caplog.at_level(logging.WARNING, logger="getpatter"): + phone._record_prewarm_waste("CA-twice") + second_warns = [ + r + for r in caplog.records + if "CA-twice" in r.message and r.levelno >= logging.WARNING + ] + + assert len(first_warns) == 1 + assert len(second_warns) == 0 + assert "CA-twice" not in phone._prewarm_audio + + +async def test_status_callback_evicts_prewarm_on_no_answer(caplog) -> None: + """Twilio status callback with CallStatus=no-answer evicts cache and WARNs once. + + Authentic test: real FastAPI app, real route, real waste recorder. + """ + from fastapi.testclient import TestClient + + phone = _make_patter() + phone._prewarm_audio["CAtest_noans001"] = b"GREETING-WASTED" + # Build a real EmbeddedServer so the real route runs end-to-end. Wire + # the waste-recorder closure exactly the way ``serve()`` does. + from getpatter.local_config import LocalConfig + from getpatter.server import EmbeddedServer + + config = LocalConfig( + telephony_provider="twilio", + webhook_url="example.test", + twilio_sid="ACtest000000000000000000000000000", + twilio_token="", # no token → unsigned form parsing path + require_signature=False, + ) + agent = Agent(system_prompt="hi", first_message="hello") + server = EmbeddedServer(config=config, agent=agent) + server.record_prewarm_waste = phone._record_prewarm_waste + + app = server._create_app() + client = TestClient(app) + + with caplog.at_level(logging.WARNING, logger="getpatter"): + resp = client.post( + "/webhooks/twilio/status", + data={"CallSid": "CAtest_noans001", "CallStatus": "no-answer"}, + ) + + assert resp.status_code == 204 + assert "CAtest_noans001" not in phone._prewarm_audio + waste_warns = [ + r + for r in caplog.records + if "CAtest_noans001" in r.message and "wasted" in r.message.lower() + ] + assert len(waste_warns) == 1 + + +async def test_status_callback_evicts_prewarm_on_busy_failed_canceled() -> None: + """All four abnormal terminations evict the cache (busy/failed/canceled/no-answer).""" + from fastapi.testclient import TestClient + + from getpatter.local_config import LocalConfig + from getpatter.server import EmbeddedServer + + config = LocalConfig( + telephony_provider="twilio", + webhook_url="example.test", + twilio_sid="ACtest000000000000000000000000000", + twilio_token="", # no token → unsigned form parsing path + require_signature=False, + ) + agent = Agent(system_prompt="hi", first_message="hello") + + for status in ("no-answer", "busy", "failed", "canceled"): + phone = _make_patter() + sid = f"CAtest_{status.replace('-', '')}" + phone._prewarm_audio[sid] = b"BYTES" + server = EmbeddedServer(config=config, agent=agent) + server.record_prewarm_waste = phone._record_prewarm_waste + app = server._create_app() + client = TestClient(app) + resp = client.post( + "/webhooks/twilio/status", + data={"CallSid": sid, "CallStatus": status}, + ) + assert resp.status_code == 204 + assert sid not in phone._prewarm_audio, f"{status} did not evict cache" + + +async def test_status_callback_does_not_evict_on_completed() -> None: + """``completed`` is a normal hangup — the cache was already drained + by the StreamHandler at firstMessage emit. Eviction here would + double-fire the WARN. Verified by a status-callback that arrives + AFTER the StreamHandler consumed the cache.""" + from fastapi.testclient import TestClient + + from getpatter.local_config import LocalConfig + from getpatter.server import EmbeddedServer + + phone = _make_patter() + # Simulate normal consumption — pop drains the cache. + phone._prewarm_audio["CAtest_done001"] = b"BYTES" + phone.pop_prewarm_audio("CAtest_done001") + assert "CAtest_done001" not in phone._prewarm_audio + + config = LocalConfig( + telephony_provider="twilio", + webhook_url="example.test", + twilio_sid="ACtest000000000000000000000000000", + twilio_token="", # no token → unsigned form parsing path + require_signature=False, + ) + agent = Agent(system_prompt="hi", first_message="hello") + server = EmbeddedServer(config=config, agent=agent) + server.record_prewarm_waste = phone._record_prewarm_waste + app = server._create_app() + client = TestClient(app) + resp = client.post( + "/webhooks/twilio/status", + data={"CallSid": "CAtest_done001", "CallStatus": "completed"}, + ) + assert resp.status_code == 204 + # No cache to evict; idempotent guard prevents double-WARN even if + # the eviction path was hit (it isn't, for ``completed``). + + +# --------------------------------------------------------------------------- +# FIX #92 — race start-vs-prewarm task (orphan bytes guard) +# --------------------------------------------------------------------------- + + +async def test_prewarm_orphan_bytes_dropped_when_consumer_polled_first(caplog) -> None: + """The classic race: prewarm task takes 500 ms; start arrives after + 100 ms; pop_prewarm_audio returns None, StreamHandler falls back to + live TTS. The prewarm task finishes 400 ms later — its bytes must + NOT land in ``_prewarm_audio`` (orphan bytes leak otherwise). + """ + + class SlowTTS(StubTTS): + async def synthesize(self, text: str): + self.synthesize_called += 1 + # Yield slowly so the consumer polls before we accumulate. + await asyncio.sleep(0.2) + yield self._audio[: len(self._audio) // 2] + await asyncio.sleep(0.05) + yield self._audio[len(self._audio) // 2 :] + + phone = _make_patter() + tts = SlowTTS(audio_bytes=b"LATE-BYTES-AAAA") + agent = Agent( + system_prompt="hi", + first_message="Hi", + tts=tts, + prewarm_first_message=True, + provider="pipeline", + ) + phone._spawn_prewarm_first_message(agent, "CA-race", ring_timeout=5) + # Simulate the carrier ``start`` arriving BEFORE synth finishes. + await asyncio.sleep(0.05) + cached = phone.pop_prewarm_audio("CA-race") + assert cached is None # cache miss → live-TTS fallback path + + with caplog.at_level(logging.WARNING, logger="getpatter"): + await _wait_for_tasks(phone, timeout=2.0) + + # The synth task finished but dropped its bytes instead of orphaning. + assert "CA-race" not in phone._prewarm_audio + orphan_warns = [ + r + for r in caplog.records + if "orphaned" in r.message.lower() and "CA-race" in r.message + ] + assert len(orphan_warns) == 1 + + +async def test_pop_prewarm_audio_marks_consumed_on_cache_hit() -> None: + """A normal cache hit must also mark the call_id as consumed so a + follow-up race-finishing synth task drops its bytes.""" + phone = _make_patter() + phone._prewarm_audio["CA-hit"] = b"BYTES" + out = phone.pop_prewarm_audio("CA-hit") + assert out == b"BYTES" + assert "CA-hit" in phone._prewarm_consumed + + +# --------------------------------------------------------------------------- +# FIX #93 — disconnect() cancels in-flight synth tasks and clears cache +# --------------------------------------------------------------------------- + + +async def test_disconnect_cancels_in_flight_prewarm_and_clears_cache() -> None: + """disconnect() must cancel still-running prewarm tasks AND clear + both ``_prewarm_audio`` and the consumed set so a subsequent + ``serve()`` does not see stale state.""" + + class VerySlowTTS(StubTTS): + async def synthesize(self, text: str): + self.synthesize_called += 1 + try: + await asyncio.sleep(10.0) + except asyncio.CancelledError: + raise + yield b"never-emitted" + + phone = _make_patter() + tts = VerySlowTTS() + agent = Agent( + system_prompt="hi", + first_message="hello", + tts=tts, + prewarm_first_message=True, + provider="pipeline", + ) + phone._spawn_prewarm_first_message(agent, "CA-disco", ring_timeout=30) + # Pre-seed cache + consumed set to verify they're cleared. + phone._prewarm_audio["CA-leftover"] = b"STALE" + phone._prewarm_consumed.add("CA-leftover") + assert phone._prewarm_tasks # confirm a task is in flight + + await phone.disconnect() + + assert phone._prewarm_audio == {} + assert phone._prewarm_consumed == set() + assert phone._prewarm_tasks == set() + assert phone._prewarm_ttl_tasks == {} + + +# --------------------------------------------------------------------------- +# FIX #94 — Realtime/ConvAI + prewarm_first_message warns and skips +# --------------------------------------------------------------------------- + + +async def test_prewarm_skipped_for_realtime_provider(caplog) -> None: + """Realtime / ConvAI never consume the cache — refuse to spawn the + prewarm task and emit a WARN.""" + phone = _make_patter() + tts = StubTTS() + agent = Agent( + system_prompt="hi", + first_message="hi", + tts=tts, + prewarm_first_message=True, + # Default provider is openai_realtime; named explicitly here for clarity. + provider="openai_realtime", + ) + with caplog.at_level(logging.WARNING, logger="getpatter"): + phone._spawn_prewarm_first_message(agent, "CA-realtime", ring_timeout=5) + await _wait_for_tasks(phone) + + assert tts.synthesize_called == 0 + assert "CA-realtime" not in phone._prewarm_audio + warn_msgs = [r for r in caplog.records if "only supported in pipeline" in r.message] + assert len(warn_msgs) == 1 + + +async def test_prewarm_skipped_for_convai_provider(caplog) -> None: + """Same guard for ElevenLabs ConvAI.""" + phone = _make_patter() + tts = StubTTS() + agent = Agent( + system_prompt="hi", + first_message="hi", + tts=tts, + prewarm_first_message=True, + provider="elevenlabs_convai", + ) + with caplog.at_level(logging.WARNING, logger="getpatter"): + phone._spawn_prewarm_first_message(agent, "CA-convai", ring_timeout=5) + await _wait_for_tasks(phone) + + assert tts.synthesize_called == 0 + assert "CA-convai" not in phone._prewarm_audio + + +# --------------------------------------------------------------------------- +# FIX #96 — bounded cache (size cap + TTL eviction) +# --------------------------------------------------------------------------- + + +async def test_prewarm_cache_size_cap(caplog) -> None: + """When the cache reaches ``_PREWARM_CACHE_MAX`` concurrent entries, + the next prewarm spawn is refused with a WARN. Live TTS still works + — only the optimisation is skipped.""" + from getpatter.client import _PREWARM_CACHE_MAX + + phone = _make_patter() + # Pre-fill the cache to the cap. + for i in range(_PREWARM_CACHE_MAX): + phone._prewarm_audio[f"CA-fill-{i:04d}"] = b"X" + + tts = StubTTS() + agent = Agent( + system_prompt="hi", + first_message="hi", + tts=tts, + prewarm_first_message=True, + provider="pipeline", + ) + + with caplog.at_level(logging.WARNING, logger="getpatter"): + phone._spawn_prewarm_first_message(agent, "CA-overflow", ring_timeout=5) + + # No new task spawned, no synth invoked. + assert tts.synthesize_called == 0 + assert "CA-overflow" not in phone._prewarm_audio + full_warns = [ + r + for r in caplog.records + if "cache full" in r.message.lower() and "CA-overflow" in r.message + ] + assert len(full_warns) == 1 + + +async def test_prewarm_ttl_eviction_after_ring_timeout_grace(caplog) -> None: + """A prewarmed entry that the carrier never ``start``s must evict + automatically ``ring_timeout + grace`` seconds after the synth + completes — no leak even when the status callback never fires.""" + + # Patch grace to a tiny value so the test runs in <1 s. The + # production constant remains the documented ring_timeout + 5 s. + import getpatter.client as client_mod + + original_grace = client_mod._PREWARM_TTL_GRACE_S + client_mod._PREWARM_TTL_GRACE_S = 0.1 + try: + phone = _make_patter() + tts = StubTTS(audio_bytes=b"TTL-BYTES") + agent = Agent( + system_prompt="hi", + first_message="hi", + tts=tts, + prewarm_first_message=True, + provider="pipeline", + ) + # ring_timeout=0.05 → TTL fires at 0.15 s after synth completes. + phone._spawn_prewarm_first_message(agent, "CA-ttl", ring_timeout=1) + await _wait_for_tasks(phone, timeout=1.0) + # Synth completed, cache is hot. + assert phone._prewarm_audio.get("CA-ttl") == b"TTL-BYTES" + + with caplog.at_level(logging.WARNING, logger="getpatter"): + # Wait for TTL eviction (1s ring + 0.1s grace = 1.1s). + await asyncio.sleep(1.3) + + assert "CA-ttl" not in phone._prewarm_audio + ttl_warns = [ + r + for r in caplog.records + if "ttl" in r.message.lower() and "CA-ttl" in r.message + ] + assert len(ttl_warns) == 1 + finally: + client_mod._PREWARM_TTL_GRACE_S = original_grace + # Ensure no dangling TTL task survives this test. + for t in list(phone._prewarm_ttl_tasks.values()): + t.cancel() + + +async def test_prewarm_ttl_cancelled_on_normal_consumption() -> None: + """When the StreamHandler pops the cache normally, the TTL eviction + task must be cancelled so it never fires a spurious WARN.""" + import getpatter.client as client_mod + + original_grace = client_mod._PREWARM_TTL_GRACE_S + client_mod._PREWARM_TTL_GRACE_S = 0.05 + try: + phone = _make_patter() + tts = StubTTS(audio_bytes=b"NORMAL-BYTES") + agent = Agent( + system_prompt="hi", + first_message="hi", + tts=tts, + prewarm_first_message=True, + provider="pipeline", + ) + phone._spawn_prewarm_first_message(agent, "CA-normal", ring_timeout=1) + await _wait_for_tasks(phone, timeout=1.0) + + # Normal consumption — should cancel the TTL. + out = phone.pop_prewarm_audio("CA-normal") + assert out == b"NORMAL-BYTES" + # TTL handle should have been removed and the underlying task + # cancelled. + assert "CA-normal" not in phone._prewarm_ttl_tasks + # Wait past the would-be eviction time to confirm no spurious + # ``add to _prewarm_audio`` happens (it can't, since the synth + # task already completed; this guards against future regressions + # where a ttl reschedule would pop a freshly-orphaned entry). + await asyncio.sleep(0.2) + assert "CA-normal" not in phone._prewarm_audio + finally: + client_mod._PREWARM_TTL_GRACE_S = original_grace + + +# --------------------------------------------------------------------------- +# FIX #97 regression — prewarm bytes must be chunked, not single-shot +# --------------------------------------------------------------------------- + + +async def test_stream_prewarm_bytes_chunks_buffer() -> None: + """``_stream_prewarm_bytes`` must split a multi-second prewarm + buffer into multiple ``audio_sender.send_audio`` calls, matching the + live-TTS chunk boundary. + + Catches the regression where a single ``send_audio(prewarm_bytes)`` + flooded Twilio's mark/clear bookkeeping with a multi-second buffer: + a ``send_clear`` issued mid-buffer would have nothing to clear, + producing the "agent keeps talking after barge-in" UX bug on the + first turn. + """ + import time as _time + + from getpatter.stream_handler import PipelineStreamHandler + + audio_sender = MagicMock() + audio_sender.send_audio = AsyncMock() + audio_sender.reset_pcm_carry = MagicMock() + + agent = Agent(system_prompt="hi", first_message="Hello!") + handler = PipelineStreamHandler( + agent=agent, + audio_sender=audio_sender, + call_id="CA-chunk-test", + caller="+15550000001", + callee="+15550000002", + resolved_prompt="hi", + metrics=None, + ) + handler._aec = None + handler._is_speaking = True + # Mark the first-audio gate so subsequent _mark_first_audio_sent calls + # are cheap no-ops; we don't care about that side effect here. + handler._first_audio_sent_at = _time.time() + + # 5 s of PCM16 @ 16 kHz mono = 5 * 16000 * 2 = 160_000 bytes. + prewarm_bytes = b"\x00\x01" * (5 * 16000) + assert len(prewarm_bytes) == 160_000 + + first_chunk_sent = await handler._stream_prewarm_bytes(prewarm_bytes) + + assert first_chunk_sent is True + # 160_000 / 1280 = 125 chunks. Anything ≥ 100 proves the buffer was + # split — we don't pin the exact count to keep the test robust to + # future chunk-size tweaks, but it's nowhere near 1. + assert audio_sender.send_audio.await_count >= 100, ( + f"prewarm buffer must be chunked; " + f"got {audio_sender.send_audio.await_count} send_audio call(s) " + f"— regression of FIX #97 (single-shot multi-second send)" + ) + # All chunks together must equal the full buffer (no bytes lost). + sent = b"".join(call.args[0] for call in audio_sender.send_audio.await_args_list) + assert sent == prewarm_bytes + # Every chunk except the last must be exactly PREWARM_CHUNK_BYTES bytes. + chunks = [call.args[0] for call in audio_sender.send_audio.await_args_list] + for chunk in chunks[:-1]: + assert len(chunk) == handler._PREWARM_CHUNK_BYTES + # The last chunk is at most PREWARM_CHUNK_BYTES. + assert len(chunks[-1]) <= handler._PREWARM_CHUNK_BYTES + + +async def test_stream_prewarm_bytes_stops_on_barge_in_mid_buffer() -> None: + """A barge-in mid-prewarm flips ``_is_speaking`` False and the + chunking loop must observe that and stop sending more audio. This is + the whole point of chunking — granularity for cancel. + """ + import time as _time + + from getpatter.stream_handler import PipelineStreamHandler + + audio_sender = MagicMock() + sent_chunks: list[bytes] = [] + + chunks_seen = 0 + + async def _send_audio(chunk: bytes) -> None: + nonlocal chunks_seen + sent_chunks.append(chunk) + chunks_seen += 1 + # After two chunks, simulate a barge-in flipping the gate. + if chunks_seen == 2: + handler._is_speaking = False + + audio_sender.send_audio = AsyncMock(side_effect=_send_audio) + + agent = Agent(system_prompt="hi", first_message="Hello!") + handler = PipelineStreamHandler( + agent=agent, + audio_sender=audio_sender, + call_id="CA-bargein-mid", + caller="+15550000001", + callee="+15550000002", + resolved_prompt="hi", + metrics=None, + ) + handler._aec = None + handler._is_speaking = True + handler._first_audio_sent_at = _time.time() + + # Long enough buffer that more than 2 chunks would be sent without + # barge-in interruption. + prewarm_bytes = b"\x00\x01" * (5 * 16000) + await handler._stream_prewarm_bytes(prewarm_bytes) + + # Exactly 2 chunks were sent; the loop broke on the third iteration + # before audio_sender.send_audio was called. + assert len(sent_chunks) == 2 + assert audio_sender.send_audio.await_count == 2 diff --git a/libraries/python/tests/test_prewarm_handoff.py b/libraries/python/tests/test_prewarm_handoff.py new file mode 100644 index 00000000..5db38687 --- /dev/null +++ b/libraries/python/tests/test_prewarm_handoff.py @@ -0,0 +1,236 @@ +"""Tests for the prewarm-handoff (FIX A) — keep parked WSs OPEN and adopt +them at call connect, instead of close-and-reopen which doesn't warm +TLS on Node ``ws`` (Python ``websockets`` has the same issue at the +TCP / TLS level). + +Coverage: + 1. ``Patter._park_provider_connections`` invokes + ``open_parked_connection`` on the configured STT / TTS adapters. + 2. The parked WS stays OPEN past the historic 250 ms idle window. + 3. ``pop_prewarmed_connections`` returns the parked handles and + removes them from the cache (consume-once semantics). + 4. ``close_prewarmed_connections`` (and ``_record_prewarm_waste``) + drains parked sockets cleanly. + 5. A handle whose underlying WS died between park and adopt is + dropped silently. + +Tests use authentic real code paths — only the carrier HTTP boundary +and provider WS open are mocked. See +``.claude/rules/authentic-tests.md``. +""" + +from __future__ import annotations + +import asyncio +from typing import AsyncIterator + +from getpatter.client import Patter +from getpatter.models import Agent +from getpatter.providers.base import STTProvider, TTSProvider, Transcript + + +class FakeWS: + """Minimal stand-in for the per-provider WS handles used in + parking tests. Mirrors the public surface the SDK reads — + ``closed`` and ``close()``.""" + + def __init__(self) -> None: + self.closed = False + + async def close(self) -> None: + self.closed = True + + +class StubSession: + """aiohttp.ClientSession-shaped stub used as the first half of + Cartesia STT's ``(session, ws)`` parked-handle tuple.""" + + def __init__(self) -> None: + self.closed = False + + async def close(self) -> None: + self.closed = True + + +class StubSTTWithPark(STTProvider): + def __init__(self) -> None: + self.park_calls = 0 + self.adopt_calls = 0 + self.parked_session: StubSession | None = None + self.parked_ws: FakeWS | None = None + + async def connect(self) -> None: # pragma: no cover - unused in handoff tests + return None + + async def send_audio(self, audio_chunk: bytes) -> None: # pragma: no cover + return None + + async def receive_transcripts( + self, + ) -> AsyncIterator[Transcript]: # pragma: no cover + if False: + yield # pragma: no cover + + async def close(self) -> None: + return None + + async def open_parked_connection(self) -> tuple[StubSession, FakeWS]: + self.park_calls += 1 + self.parked_session = StubSession() + self.parked_ws = FakeWS() + return self.parked_session, self.parked_ws + + def adopt_websocket( + self, session: StubSession, ws: FakeWS + ) -> None: # pragma: no cover - drained via pop in tests + self.adopt_calls += 1 + + +class StubParkedTTS: + """Mimic of ``ElevenLabsParkedWS``: object with ``.ws`` attribute.""" + + def __init__(self) -> None: + self.ws = FakeWS() + self.bos_sent = True + + +class StubTTSWithPark(TTSProvider): + def __init__(self) -> None: + self.park_calls = 0 + self.adopt_calls = 0 + self.parked_handle: StubParkedTTS | None = None + + async def synthesize(self, text: str) -> AsyncIterator[bytes]: # pragma: no cover + if False: + yield b"" + + async def close(self) -> None: + return None + + async def open_parked_connection(self) -> StubParkedTTS: + self.park_calls += 1 + self.parked_handle = StubParkedTTS() + return self.parked_handle + + def adopt_websocket( + self, parked: StubParkedTTS + ) -> None: # pragma: no cover - drained via pop in tests + self.adopt_calls += 1 + + +def _make_patter() -> Patter: + from getpatter.carriers.twilio import Carrier as Twilio + + return Patter( + carrier=Twilio( + account_sid="ACtest000000000000000000000000000", + auth_token="test_auth_token_000000000000000000", + ), + phone_number="+15551234567", + webhook_url="example.test", + ) + + +async def _drain(phone: Patter, timeout: float = 1.0) -> None: + if phone._prewarm_tasks: + await asyncio.wait_for( + asyncio.gather(*phone._prewarm_tasks, return_exceptions=True), + timeout=timeout, + ) + + +async def test_park_provider_connections_calls_open_on_stt_and_tts() -> None: + phone = _make_patter() + stt = StubSTTWithPark() + tts = StubTTSWithPark() + agent = Agent(system_prompt="p", provider="pipeline", stt=stt, tts=tts) + phone._park_provider_connections(agent, "CAtest1") + await _drain(phone) + assert stt.park_calls == 1 + assert tts.park_calls == 1 + + +async def test_parked_ws_stays_open_past_historic_idle_window() -> None: + phone = _make_patter() + stt = StubSTTWithPark() + tts = StubTTSWithPark() + agent = Agent(system_prompt="p", provider="pipeline", stt=stt, tts=tts) + phone._park_provider_connections(agent, "CAtest2") + await _drain(phone) + # Sleep well past the historic 250 ms warmup-then-close window. + await asyncio.sleep(0.4) + assert stt.parked_ws is not None and not stt.parked_ws.closed + assert tts.parked_handle is not None and not tts.parked_handle.ws.closed + + +async def test_pop_prewarmed_connections_consume_once() -> None: + phone = _make_patter() + stt = StubSTTWithPark() + tts = StubTTSWithPark() + agent = Agent(system_prompt="p", provider="pipeline", stt=stt, tts=tts) + phone._park_provider_connections(agent, "CAtest3") + await _drain(phone) + slot = phone.pop_prewarmed_connections("CAtest3") + assert slot is not None + assert slot["stt"] == (stt.parked_session, stt.parked_ws) + assert slot["tts"] is tts.parked_handle + # Second pop returns None — slot already drained. + assert phone.pop_prewarmed_connections("CAtest3") is None + + +async def test_close_prewarmed_connections_drains_sockets() -> None: + phone = _make_patter() + stt = StubSTTWithPark() + tts = StubTTSWithPark() + agent = Agent(system_prompt="p", provider="pipeline", stt=stt, tts=tts) + phone._park_provider_connections(agent, "CAtest4") + await _drain(phone) + assert stt.parked_ws is not None and not stt.parked_ws.closed + phone.close_prewarmed_connections("CAtest4") + # Closes are scheduled asynchronously via create_task — drain them. + for _ in range(5): + await asyncio.sleep(0) + assert stt.parked_ws.closed is True + assert tts.parked_handle is not None and tts.parked_handle.ws.closed is True + # Slot drained. + assert phone.pop_prewarmed_connections("CAtest4") is None + + +async def test_record_prewarm_waste_drains_parked_sockets() -> None: + phone = _make_patter() + stt = StubSTTWithPark() + tts = StubTTSWithPark() + agent = Agent(system_prompt="p", provider="pipeline", stt=stt, tts=tts) + phone._park_provider_connections(agent, "CAtest5") + await _drain(phone) + phone._record_prewarm_waste("CAtest5") + for _ in range(5): + await asyncio.sleep(0) + assert stt.parked_ws is not None and stt.parked_ws.closed is True + assert tts.parked_handle is not None and tts.parked_handle.ws.closed is True + + +async def test_park_skipped_when_neither_provider_supports_parking() -> None: + phone = _make_patter() + + # Adapters without ``open_parked_connection`` must not allocate a slot. + class MinimalSTT(STTProvider): + async def connect(self) -> None: + return None + + async def send_audio(self, _ac: bytes) -> None: + return None + + async def receive_transcripts( + self, + ) -> AsyncIterator[Transcript]: # pragma: no cover + if False: + yield # pragma: no cover + + async def close(self) -> None: + return None + + agent = Agent(system_prompt="p", provider="pipeline", stt=MinimalSTT()) + phone._park_provider_connections(agent, "CAtest6") + # No slot was created — pop returns None. + assert phone.pop_prewarmed_connections("CAtest6") is None diff --git a/libraries/python/tests/test_pricing.py b/libraries/python/tests/test_pricing.py index 4f9b08cb..c03b1868 100644 --- a/libraries/python/tests/test_pricing.py +++ b/libraries/python/tests/test_pricing.py @@ -43,10 +43,12 @@ def test_empty_overrides(self): class TestCalculateSTTCost: def test_deepgram_cost(self): pricing = merge_pricing(None) - # 60 seconds = 1 minute at $0.0077/min (Nova-3 streaming monolingual, - # the Patter default). Previous $0.0043/min was the batch rate. + # 60 seconds = 1 minute at $0.0048/min (Nova-3 streaming monolingual, + # the Patter default). Current Pay-As-You-Go promotional rate per + # deepgram.com/pricing (verified 2026-05-11). The legacy $0.0077/min + # was the launch-era standard rate. cost = calculate_stt_cost("deepgram", 60.0, pricing) - assert abs(cost - 0.0077) < 1e-6 + assert abs(cost - 0.0048) < 1e-6 def test_whisper_cost(self): pricing = merge_pricing(None) @@ -68,10 +70,10 @@ def test_unknown_provider(self): class TestCalculateTTSCost: def test_elevenlabs_cost(self): pricing = merge_pricing(None) - # 1000 characters at $0.06/1k = $0.06 (eleven_flash_v2_5 default; - # previous $0.18 was the Creator plan overage rate). + # 1000 characters at $0.05/1k = $0.05 (eleven_flash_v2_5 default). + # Source: https://elevenlabs.io/pricing/api (verified 2026-05-11). cost = calculate_tts_cost("elevenlabs", 1000, pricing) - assert abs(cost - 0.06) < 1e-6 + assert abs(cost - 0.05) < 1e-6 def test_openai_tts_cost(self): pricing = merge_pricing(None) @@ -223,6 +225,33 @@ def test_telnyx_cost(self): cost = calculate_telephony_cost("telnyx", 600.0, pricing) assert abs(cost - 0.07) < 1e-6 + def test_telnyx_inbound_rate(self): + """US inbound DID / local termination billed at $0.0035/min. + + Verified against https://telnyx.com/pricing/elastic-sip (2026-05-11). + """ + pricing = merge_pricing(None) + cost = calculate_telephony_cost("telnyx_inbound", 60.0, pricing) + assert abs(cost - 0.0035) < 1e-6 + + def test_telnyx_outbound_rate(self): + """US outbound Pay-As-You-Go billed at $0.007/min (mid-range).""" + pricing = merge_pricing(None) + cost = calculate_telephony_cost("telnyx_outbound", 60.0, pricing) + assert abs(cost - 0.007) < 1e-6 + + def test_telnyx_legacy_falls_back_to_outbound(self): + """Legacy flat ``telnyx`` key remains at $0.007/min (outbound-safe). + + Users who override ``pricing={"telnyx": {...}}`` without a + direction-aware split keep the previous behaviour. Direction-aware + billing is opt-in via the new ``telnyx_inbound`` / ``telnyx_outbound`` + keys. + """ + pricing = merge_pricing(None) + cost = calculate_telephony_cost("telnyx", 60.0, pricing) + assert abs(cost - 0.007) < 1e-6 + def test_zero_duration(self): pricing = merge_pricing(None) cost = calculate_telephony_cost("twilio", 0.0, pricing) @@ -286,21 +315,22 @@ class TestModelAwareSttPricing: def test_deepgram_default_is_nova3_streaming(self): pricing = merge_pricing(None) - # 60 seconds at $0.0077/min = $0.0077 (nova-3 default) - assert calculate_stt_cost("deepgram", 60.0, pricing) == pytest.approx(0.0077) + # 60 seconds at $0.0048/min = $0.0048 (nova-3 default, current PAYG + # promotional rate per deepgram.com/pricing, verified 2026-05-11). + assert calculate_stt_cost("deepgram", 60.0, pricing) == pytest.approx(0.0048) def test_deepgram_multilingual_uses_nested_rate(self): pricing = merge_pricing(None) - # nova-3-multilingual is $0.0092/min + # nova-3-multilingual is $0.0058/min (PAYG promo rate) cost = calculate_stt_cost( "deepgram", 60.0, pricing, model="nova-3-multilingual" ) - assert cost == pytest.approx(0.0092) + assert cost == pytest.approx(0.0058) def test_deepgram_unknown_model_falls_back_to_default(self): pricing = merge_pricing(None) cost = calculate_stt_cost("deepgram", 60.0, pricing, model="some-future-model") - assert cost == pytest.approx(0.0077) + assert cost == pytest.approx(0.0048) def test_whisper_per_model_rates(self): pricing = merge_pricing(None) @@ -321,15 +351,16 @@ class TestModelAwareTtsPricing: def test_elevenlabs_default_is_flash_v2_5(self): pricing = merge_pricing(None) - # 1000 chars at $0.06/1k = $0.06 - assert calculate_tts_cost("elevenlabs", 1000, pricing) == pytest.approx(0.06) + # 1000 chars at $0.05/1k = $0.05 (verified vs elevenlabs.io/pricing/api) + assert calculate_tts_cost("elevenlabs", 1000, pricing) == pytest.approx(0.05) def test_elevenlabs_multilingual_v2_per_model_rate(self): pricing = merge_pricing(None) cost = calculate_tts_cost( "elevenlabs", 1000, pricing, model="eleven_multilingual_v2" ) - assert cost == pytest.approx(0.18) + # Multilingual v2 / v3 share the $0.10/1k tier per the public API page. + assert cost == pytest.approx(0.10) def test_openai_tts_hd_per_model_rate(self): pricing = merge_pricing(None) @@ -360,10 +391,10 @@ def test_models_dict_merges_shallowly(self): assert calculate_tts_cost( "elevenlabs", 1000, pricing, model="eleven_flash_v2_5" ) == pytest.approx(0.04) - # Untouched — still original $0.18 + # Untouched — still original $0.10 assert calculate_tts_cost( "elevenlabs", 1000, pricing, model="eleven_multilingual_v2" - ) == pytest.approx(0.18) + ) == pytest.approx(0.10) def test_user_can_register_brand_new_model(self): pricing = merge_pricing( @@ -400,17 +431,20 @@ class TestLLMCostBilling: def test_cerebras_default_model_is_billed(self): """Cerebras default ``gpt-oss-120b`` must produce a non-zero bill.""" cost = calculate_llm_cost("cerebras", "gpt-oss-120b", 1000, 1000) - # Real rate-card math, no mock: 1000 in @ $0.85/M + 1000 out @ $1.20/M + # Real rate-card math (2026-05-11 docs): 1000 in @ $0.35/M + 1000 out @ $0.75/M. + # Updated from $0.85/$1.20 — see CHANGELOG 0.6.1 pricing correction. assert cost == pytest.approx( - (1000 / 1_000_000) * 0.85 + (1000 / 1_000_000) * 1.20 + (1000 / 1_000_000) * 0.35 + (1000 / 1_000_000) * 0.75 ) assert cost > 0.0 def test_cerebras_llama_3_1_8b_is_billed(self): """``llama3.1-8b`` (deprecating 2026-05-27 but still supported) must bill.""" cost = calculate_llm_cost("cerebras", "llama3.1-8b", 1000, 1000) + # Real rate-card math (2026-05-11 docs): 1000 in @ $0.10/M + 1000 out @ $0.10/M. + # Output was $0.20/M previously — corrected to match Cerebras docs. assert cost == pytest.approx( - (1000 / 1_000_000) * 0.10 + (1000 / 1_000_000) * 0.20 + (1000 / 1_000_000) * 0.10 + (1000 / 1_000_000) * 0.10 ) assert cost > 0.0 diff --git a/libraries/python/tests/unit/test_barge_in_strategies.py b/libraries/python/tests/unit/test_barge_in_strategies.py new file mode 100644 index 00000000..d1130b2e --- /dev/null +++ b/libraries/python/tests/unit/test_barge_in_strategies.py @@ -0,0 +1,219 @@ +"""Unit tests for ``getpatter.services.barge_in_strategies``.""" + +from __future__ import annotations + +import pytest + +from getpatter.services.barge_in_strategies import ( + BargeInStrategy, + MinWordsStrategy, + evaluate_strategies, + reset_strategies, +) + + +class TestMinWordsStrategy: + def test_init_rejects_min_words_below_one(self) -> None: + with pytest.raises(ValueError): + MinWordsStrategy(min_words=0) + with pytest.raises(ValueError): + MinWordsStrategy(min_words=-3) + + async def test_one_word_confirms_when_agent_silent(self) -> None: + s = MinWordsStrategy(min_words=3) + # Agent not speaking → 1 word is enough; we don't want to delay + # the very first user turn just because the strategy is configured. + assert ( + await s.evaluate(transcript="hi", is_interim=False, agent_speaking=False) + is True + ) + + async def test_below_threshold_during_agent_speech_does_not_confirm(self) -> None: + s = MinWordsStrategy(min_words=3) + assert ( + await s.evaluate(transcript="okay", is_interim=False, agent_speaking=True) + is False + ) + assert ( + await s.evaluate(transcript="uh huh", is_interim=False, agent_speaking=True) + is False + ) + + async def test_meets_threshold_during_agent_speech_confirms(self) -> None: + s = MinWordsStrategy(min_words=3) + assert ( + await s.evaluate( + transcript="please stop talking", + is_interim=False, + agent_speaking=True, + ) + is True + ) + assert ( + await s.evaluate( + transcript="hold on a moment please", + is_interim=False, + agent_speaking=True, + ) + is True + ) + + async def test_use_interim_false_ignores_partials(self) -> None: + s = MinWordsStrategy(min_words=2, use_interim=False) + # An interim with enough words is not enough — still wait for final. + assert ( + await s.evaluate( + transcript="please stop", is_interim=True, agent_speaking=True + ) + is False + ) + # The final partial of the same utterance confirms. + assert ( + await s.evaluate( + transcript="please stop", is_interim=False, agent_speaking=True + ) + is True + ) + + async def test_word_count_uses_whitespace_split(self) -> None: + s = MinWordsStrategy(min_words=2) + # Multiple spaces, leading/trailing whitespace, tabs collapse correctly. + assert ( + await s.evaluate( + transcript=" hello world ", + is_interim=False, + agent_speaking=True, + ) + is True + ) + assert ( + await s.evaluate( + transcript="\thello\n", + is_interim=False, + agent_speaking=True, + ) + is False + ) + + async def test_empty_transcript_does_not_confirm_during_agent_speech(self) -> None: + s = MinWordsStrategy(min_words=2) + assert ( + await s.evaluate(transcript="", is_interim=False, agent_speaking=True) + is False + ) + + async def test_protocol_runtime_check(self) -> None: + # Sanity: MinWordsStrategy structurally satisfies the Protocol. + assert isinstance(MinWordsStrategy(min_words=2), BargeInStrategy) + + +class _RecordingStrategy: + """Test double that records every call and returns a configurable result.""" + + def __init__(self, *, returns: bool) -> None: + self._returns = returns + self.calls: list[dict] = [] + self.resets = 0 + + async def evaluate( + self, *, transcript: str, is_interim: bool, agent_speaking: bool + ) -> bool: + self.calls.append( + { + "transcript": transcript, + "is_interim": is_interim, + "agent_speaking": agent_speaking, + } + ) + return self._returns + + async def reset(self) -> None: + self.resets += 1 + + +class TestEvaluateStrategies: + async def test_empty_tuple_returns_false(self) -> None: + assert ( + await evaluate_strategies( + (), transcript="anything", is_interim=False, agent_speaking=True + ) + is False + ) + + async def test_first_true_short_circuits(self) -> None: + a = _RecordingStrategy(returns=True) + b = _RecordingStrategy(returns=False) + result = await evaluate_strategies( + (a, b), + transcript="please stop", + is_interim=False, + agent_speaking=True, + ) + assert result is True + assert len(a.calls) == 1 + # Short-circuit: ``b`` MUST NOT be invoked once ``a`` confirmed. + assert b.calls == [] + + async def test_all_false_returns_false(self) -> None: + a = _RecordingStrategy(returns=False) + b = _RecordingStrategy(returns=False) + result = await evaluate_strategies( + (a, b), transcript="okay", is_interim=False, agent_speaking=True + ) + assert result is False + assert len(a.calls) == 1 + assert len(b.calls) == 1 + + async def test_strategy_exception_is_swallowed(self) -> None: + class _Boom: + async def evaluate( + self, + *, + transcript: str, + is_interim: bool, + agent_speaking: bool, + ) -> bool: + raise RuntimeError("boom") + + async def reset(self) -> None: + return None + + ok = _RecordingStrategy(returns=True) + # The crashing strategy must not abort the loop — the next strategy + # should still get its turn and confirm. + result = await evaluate_strategies( + (_Boom(), ok), + transcript="please stop talking", + is_interim=False, + agent_speaking=True, + ) + assert result is True + assert len(ok.calls) == 1 + + +class TestResetStrategies: + async def test_resets_each_strategy(self) -> None: + a = _RecordingStrategy(returns=False) + b = _RecordingStrategy(returns=False) + await reset_strategies((a, b)) + assert a.resets == 1 + assert b.resets == 1 + + async def test_swallows_per_strategy_errors(self) -> None: + class _Boom: + async def evaluate( + self, + *, + transcript: str, + is_interim: bool, + agent_speaking: bool, + ) -> bool: + return False + + async def reset(self) -> None: + raise RuntimeError("boom") + + ok = _RecordingStrategy(returns=False) + # Must not raise even though the first strategy's reset blew up. + await reset_strategies((_Boom(), ok)) + assert ok.resets == 1 diff --git a/libraries/python/tests/unit/test_barge_in_two_stage.py b/libraries/python/tests/unit/test_barge_in_two_stage.py new file mode 100644 index 00000000..646c89e6 --- /dev/null +++ b/libraries/python/tests/unit/test_barge_in_two_stage.py @@ -0,0 +1,354 @@ +"""Two-stage barge-in regression tests. + +Cover the wiring between :class:`PipelineStreamHandler` and the opt-in +``barge_in_strategies`` confirmation pipeline: + +* a sub-threshold transcript while the agent is speaking does NOT cancel + the agent (legacy behaviour cancelled unconditionally on any + transcript while ``_is_speaking``); +* a transcript that meets the strategy's threshold confirms the + barge-in and runs the cancel path (LLM cancel event set, sendClear + fired, ``_is_speaking`` flipped to False); +* the pending timeout drops the pending state and emits an + ``overlap_end(was_interruption=False)`` metric. +""" + +from __future__ import annotations + +import asyncio +import time +from unittest.mock import AsyncMock, MagicMock + +import pytest + +from getpatter.providers.base import Transcript +from getpatter.services.barge_in_strategies import MinWordsStrategy +from getpatter.stream_handler import PipelineStreamHandler + + +def _make_handler(*, strategies, confirm_s: float = 0.05) -> PipelineStreamHandler: + """Build a minimally-wired handler suitable for unit-testing the + barge-in decision path. Uses ``object.__new__`` to skip the heavy + __init__ and only sets the fields the barge-in path actually + touches.""" + h = object.__new__(PipelineStreamHandler) + h._is_speaking = True + h._aec = None # PSTN default — anti-flicker gate + h._speaking_started_at = time.time() - 1.0 # past the 0.25 s gate + h._first_audio_sent_at = time.time() - 1.0 # post first byte + h._speaking_generation = 0 + h._last_cancel_at = None + h._llm_cancel_event = asyncio.Event() + h.metrics = MagicMock() + h.metrics.record_overlap_start = MagicMock() + h.metrics.record_overlap_end = MagicMock() + h.metrics.record_bargein_detected = MagicMock() + h.metrics.record_tts_stopped = MagicMock() + h.metrics.record_turn_interrupted = MagicMock() + h.call_id = "test-call" + h.audio_sender = MagicMock() + h.audio_sender.send_clear = AsyncMock() + h._inbound_audio_ring = [] + h._stt = None + h._barge_in_strategies = tuple(strategies) + h._barge_in_confirm_s = confirm_s + h._barge_in_pending_since = None + h._barge_in_pending_task = None + return h + + +class TestBargeInTwoStageConfirmation: + async def test_sub_threshold_transcript_does_not_cancel(self) -> None: + h = _make_handler(strategies=[MinWordsStrategy(min_words=3)]) + + await h._handle_barge_in( + Transcript(text="okay", is_final=True, speech_final=True) + ) + + assert h._is_speaking is True, ( + "single-word backchannel must NOT cancel the agent when MinWords=3" + ) + assert not h._llm_cancel_event.is_set() + h.audio_sender.send_clear.assert_not_called() + h.metrics.record_turn_interrupted.assert_not_called() + + async def test_meets_threshold_confirms_and_cancels(self) -> None: + h = _make_handler(strategies=[MinWordsStrategy(min_words=3)]) + + await h._handle_barge_in( + Transcript( + text="please stop talking now", + is_final=True, + speech_final=True, + ) + ) + + assert h._is_speaking is False + assert h._llm_cancel_event.is_set() + h.audio_sender.send_clear.assert_awaited_once() + h.metrics.record_turn_interrupted.assert_called_once() + h.metrics.record_overlap_end.assert_called_once() + + async def test_legacy_path_cancels_immediately_with_no_strategies(self) -> None: + h = _make_handler(strategies=[]) + + await h._handle_barge_in( + Transcript(text="okay", is_final=True, speech_final=True) + ) + + # Without any strategies the legacy contract holds: the very first + # transcript while ``_is_speaking`` cancels the agent. + assert h._is_speaking is False + assert h._llm_cancel_event.is_set() + + +class TestPendingBargeInLifecycle: + async def test_pending_timeout_clears_state_and_records_overlap_end(self) -> None: + h = _make_handler(strategies=[MinWordsStrategy(min_words=3)], confirm_s=0.03) + + await h._start_pending_barge_in() + assert h._barge_in_pending_since is not None + assert h._barge_in_pending_task is not None + h.metrics.record_overlap_start.assert_called_once() + + # Wait past the timeout. + await asyncio.sleep(0.08) + + assert h._barge_in_pending_since is None + assert h._barge_in_pending_task is None + # Timeout emits overlap_end(was_interruption=False) — distinguishing + # genuine cancels from "agent kept talking, false positive". + h.metrics.record_overlap_end.assert_called_once() + called_with = h.metrics.record_overlap_end.call_args + # The timeout path passes was_interruption=False (positional or + # keyword); accept either. + if called_with.kwargs: + assert called_with.kwargs.get("was_interruption") is False + else: + assert called_with.args == (False,) or called_with.args[0] is False + + async def test_clear_pending_cancels_timeout_task(self) -> None: + h = _make_handler(strategies=[MinWordsStrategy(min_words=3)], confirm_s=10) + + await h._start_pending_barge_in() + task = h._barge_in_pending_task + assert task is not None and not task.done() + + h._clear_pending_barge_in() + # Yield once so the cancellation propagates. + await asyncio.sleep(0) + + assert h._barge_in_pending_since is None + assert h._barge_in_pending_task is None + assert task.cancelled() or task.done() + + async def test_confirmation_clears_pending(self) -> None: + h = _make_handler(strategies=[MinWordsStrategy(min_words=2)], confirm_s=10) + await h._start_pending_barge_in() + assert h._barge_in_pending_since is not None + + await h._handle_barge_in( + Transcript(text="please stop", is_final=True, speech_final=True) + ) + + # The cancel path must also drop pending state. + assert h._barge_in_pending_since is None + assert h._barge_in_pending_task is None + assert h._is_speaking is False + + +class TestBargeInIdempotency: + async def test_double_start_pending_is_noop(self) -> None: + h = _make_handler(strategies=[MinWordsStrategy(min_words=3)], confirm_s=10) + + await h._start_pending_barge_in() + first_since = h._barge_in_pending_since + first_task = h._barge_in_pending_task + + # Second call must not overwrite or restart the timer. + await h._start_pending_barge_in() + + assert h._barge_in_pending_since == first_since + assert h._barge_in_pending_task is first_task + # overlap_start was called once — strategy is idempotent. + h.metrics.record_overlap_start.assert_called_once() + + +@pytest.mark.parametrize("min_words", [2, 3, 5]) +async def test_min_words_threshold_is_honoured_end_to_end(min_words: int) -> None: + h = _make_handler(strategies=[MinWordsStrategy(min_words=min_words)]) + + # Below threshold: keep agent talking + below = "word " * (min_words - 1) + await h._handle_barge_in( + Transcript(text=below.strip(), is_final=True, speech_final=True) + ) + assert h._is_speaking is True + + # At threshold: confirm + at = "word " * min_words + await h._handle_barge_in( + Transcript(text=at.strip(), is_final=True, speech_final=True) + ) + assert h._is_speaking is False + + +class TestBargeInOverlapStartPreserved: + """Regression tests: ``InterruptionMetrics.detection_delay_ms`` must + measure VAD-T1 → strategy-confirm-T2, not T2 → T2 (~0). When VAD has + already started the overlap window via ``_start_pending_barge_in``, + the strategy-confirm path MUST NOT overwrite T1 with another + ``record_overlap_start`` call. + """ + + async def test_strategy_confirm_does_not_restart_overlap_window(self) -> None: + """VAD speech_start stamps T1, strategy confirm preserves T1.""" + h = _make_handler(strategies=[MinWordsStrategy(min_words=3)], confirm_s=10) + + # Stage 1: VAD fires speech_start during TTS → pending. + await h._start_pending_barge_in() + assert h._barge_in_pending_since is not None + h.metrics.record_overlap_start.assert_called_once() + + # Stage 2: STT delivers a confirming transcript ~200 ms later. + await h._handle_barge_in( + Transcript( + text="please stop talking now", + is_final=True, + speech_final=True, + ) + ) + + # Cancel ran (agent stopped, sendClear fired) — but + # record_overlap_start MUST still have been called only once. + # If it were called twice, the second call would overwrite T1 + # with T2 and ``record_overlap_end`` (called inside the cancel + # path) would compute detection_delay = T2 - T2 ≈ 0. + assert h._is_speaking is False + h.metrics.record_overlap_start.assert_called_once() + h.metrics.record_bargein_detected.assert_called_once() + h.metrics.record_overlap_end.assert_called_once() + + async def test_legacy_path_still_records_overlap_start_once(self) -> None: + """Without strategies (no VAD pending phase), the legacy + cancel path is the SOLE caller of record_overlap_start — + confirms backward compat. + """ + h = _make_handler(strategies=[]) + + await h._handle_barge_in( + Transcript(text="okay", is_final=True, speech_final=True) + ) + + assert h._is_speaking is False + h.metrics.record_overlap_start.assert_called_once() + h.metrics.record_bargein_detected.assert_called_once() + h.metrics.record_overlap_end.assert_called_once() + + async def test_detection_delay_ms_via_real_metrics(self) -> None: + """End-to-end: drive a real CallMetricsAccumulator through the + VAD → strategy-confirm flow, time-shift T1 by 200 ms, and + assert the emitted InterruptionMetrics.detection_delay matches + ~200 ms — NOT ~0. + + Catches the regression where ``_do_cancel_for_barge_in`` called + ``record_overlap_start()`` a second time, overwriting T1 and + producing detection_delay ≈ 0. + """ + from getpatter.observability.event_bus import EventBus + from getpatter.observability.metric_types import InterruptionMetrics + from getpatter.services.metrics import CallMetricsAccumulator + + h = _make_handler(strategies=[MinWordsStrategy(min_words=3)], confirm_s=10) + # Replace the MagicMock metrics with a real accumulator wired to + # an EventBus we can inspect. + bus = EventBus() + emitted: list[InterruptionMetrics] = [] + bus.on("interruption", lambda m: emitted.append(m)) + real_metrics = CallMetricsAccumulator( + call_id="test-call", + provider_mode="pipeline", + telephony_provider="twilio", + stt_provider="deepgram", + tts_provider="elevenlabs", + llm_provider="openai", + pricing=None, + report_only_initial_ttfb=False, + ) + real_metrics.attach_event_bus(bus) + h.metrics = real_metrics + + # Stage 1: VAD fires speech_start at T1. + t1 = time.time() - 0.200 # 200 ms ago + real_metrics.record_overlap_start(ts=t1) + h._barge_in_pending_since = t1 + # Manually set pending state so _do_cancel_for_barge_in observes it. + + # Stage 2: STT delivers the confirming transcript NOW. + await h._handle_barge_in( + Transcript( + text="please stop talking now", + is_final=True, + speech_final=True, + ) + ) + + assert h._is_speaking is False + assert len(emitted) == 1, "exactly one interruption metric expected" + # detection_delay is in seconds; we expect ~0.2 s (200 ms), + # NOT ~0 s. Allow a generous upper bound for CI scheduling jitter. + delay = emitted[0].detection_delay + assert 0.150 <= delay <= 0.500, ( + f"detection_delay must reflect VAD→confirm window (~200 ms), " + f"got {delay:.4f} s — likely the second record_overlap_start " + f"overwrote T1, regressing FIX #88" + ) + + +class TestCleanupClearsPendingBargeIn: + """Regression: ``PipelineStreamHandler.cleanup`` must drop any + pending barge-in timeout task before tearing down adapters. A leaked + task fires ``record_overlap_end`` on a finalised metrics object + ``barge_in_confirm_ms`` later — slow leak in long-running servers. + """ + + async def test_cleanup_cancels_pending_barge_in_task(self) -> None: + h = _make_handler(strategies=[MinWordsStrategy(min_words=3)], confirm_s=10) + # Stub the handler's cleanup-time fields so the rest of cleanup() + # is a no-op — only the pending-barge-in path matters here. + h._stt_task = None + h._stt = None + h._tts = None + h._remote_handler = None + h._resampler_8k_to_16k = None + + await h._start_pending_barge_in() + task = h._barge_in_pending_task + assert task is not None and not task.done() + # Reset the mock so we can spot any spurious call after cleanup. + h.metrics.record_overlap_end.reset_mock() + + await h.cleanup() + + # Yield to let any leaked timeout task wake up — if the bug + # regresses, the task would NOT be cancelled and would call + # record_overlap_end after the handler is gone. + await asyncio.sleep(0) + assert h._barge_in_pending_since is None + assert h._barge_in_pending_task is None + assert task.cancelled() or task.done() + # No spurious overlap_end fired during/after cleanup. + h.metrics.record_overlap_end.assert_not_called() + + async def test_cleanup_is_idempotent_without_pending_state(self) -> None: + """Backward-compat: legacy callers (no strategies, no pending + state) must observe identical cleanup behaviour.""" + h = _make_handler(strategies=[]) + h._stt_task = None + h._stt = None + h._tts = None + h._remote_handler = None + h._resampler_8k_to_16k = None + + await h.cleanup() # should not raise + h.metrics.record_overlap_end.assert_not_called() diff --git a/libraries/python/tests/unit/test_dashboard_store_unit.py b/libraries/python/tests/unit/test_dashboard_store_unit.py index 6b9b9c71..42b801f3 100644 --- a/libraries/python/tests/unit/test_dashboard_store_unit.py +++ b/libraries/python/tests/unit/test_dashboard_store_unit.py @@ -4,7 +4,6 @@ import asyncio import time -from unittest.mock import MagicMock import pytest @@ -32,8 +31,12 @@ def _make_call_metrics(call_id: str) -> CallMetrics: duration_seconds=30.0, turns=(), cost=CostBreakdown(stt=0.01, tts=0.02, llm=0.03, telephony=0.005, total=0.065), - latency_avg=LatencyBreakdown(stt_ms=50.0, llm_ms=100.0, tts_ms=30.0, total_ms=180.0), - latency_p95=LatencyBreakdown(stt_ms=60.0, llm_ms=120.0, tts_ms=40.0, total_ms=220.0), + latency_avg=LatencyBreakdown( + stt_ms=50.0, llm_ms=100.0, tts_ms=30.0, total_ms=180.0 + ), + latency_p95=LatencyBreakdown( + stt_ms=60.0, llm_ms=120.0, tts_ms=40.0, total_ms=220.0 + ), provider_mode="pipeline", ) @@ -375,3 +378,130 @@ def test_after_calls(self) -> None: store.record_call_end({"call_id": "c1"}) store.record_call_end({"call_id": "c2"}) assert store.call_count == 2 + + +# --------------------------------------------------------------------------- +# Bug C regression — record_call_end must not duplicate after +# update_call_status already moved the row to completed. +# --------------------------------------------------------------------------- + + +@pytest.mark.unit +class TestRecordCallEndDeduplication: + """Twilio's statusCallback for ``CallStatus=completed`` invokes + ``update_call_status`` (which moves the row from active to completed), + and shortly after the WS ``stop`` frame invokes ``record_call_end`` + for the same call_id. Before the fix the second call appended a + duplicate row with ``started_at=0`` and empty caller/callee, masking + the original entry in ``get_calls`` (newest-first ordering, the + dashboard's mergeCalls de-dup keeps the first match). + """ + + def test_updates_existing_entry_instead_of_duplicating(self) -> None: + store = MetricsStore() + store.record_call_initiated( + { + "call_id": "CA-dup", + "caller": "+15551112222", + "callee": "+15553334444", + "direction": "outbound", + } + ) + store.record_call_start({"call_id": "CA-dup"}) + # Twilio statusCallback path moves the call to completed first. + store.update_call_status("CA-dup", "completed", duration_seconds=42.0) + assert len(store.get_active_calls()) == 0 + assert store.call_count == 1 + intermediate = store.get_calls()[0] + assert intermediate["caller"] == "+15551112222" + assert intermediate["callee"] == "+15553334444" + started_at_before = intermediate["started_at"] + assert started_at_before > 0 + + # Then the WS stop handler fires record_call_end. ``data["caller"]`` + # is empty here because outbound TwiML carries no Stream + # parameters. + store.record_call_end( + { + "call_id": "CA-dup", + "caller": "", + "callee": "", + "transcript": [], + }, + metrics=_make_call_metrics("CA-dup"), + ) + + # No duplicate row. + assert store.call_count == 1 + final_entry = store.get_calls()[0] + assert final_entry["call_id"] == "CA-dup" + # caller/callee preserved from the original update_call_status path. + assert final_entry["caller"] == "+15551112222" + assert final_entry["callee"] == "+15553334444" + # started_at preserved (not re-zeroed). + assert final_entry["started_at"] == started_at_before + # Metrics are now populated by record_call_end. + assert final_entry["metrics"]["cost"]["total"] == pytest.approx(0.065) + assert final_entry["status"] == "completed" + + def test_call_stays_in_24h_window_after_end(self) -> None: + # End-to-end check that mirrors the real bug: the dashboard SPA + # filters calls by [now - 24h, now] using ``startedAtMs``. With the + # duplicate bug ``started_at`` was 0 → call dropped off the slice. + store = MetricsStore() + store.record_call_initiated( + { + "call_id": "CA-window", + "caller": "+15551112222", + "callee": "+15553334444", + "direction": "outbound", + } + ) + store.record_call_start({"call_id": "CA-window"}) + store.update_call_status("CA-window", "completed", duration_seconds=5.0) + store.record_call_end( + {"call_id": "CA-window", "transcript": []}, + metrics=_make_call_metrics("CA-window"), + ) + now = time.time() + in_window = store.get_calls_in_range(from_ts=now - 86400, to_ts=now + 60) + assert any(c["call_id"] == "CA-window" for c in in_window) + + +# --------------------------------------------------------------------------- +# Bug A regression — call detail route returns active record for live calls +# (the route lookup is in routes.py — these tests verify get_active is +# wired and that a freshly-started call exposes its turns through the same +# accessor the route now falls back to). +# --------------------------------------------------------------------------- + + +@pytest.mark.unit +class TestActiveCallDetail: + """get_active exposes the live record so the dashboard route can fall + back to it while the call is in flight (otherwise the live transcript + pane stays empty until the call ends). + """ + + def test_get_active_returns_record_with_turns(self) -> None: + store = MetricsStore() + store.record_call_start(_make_call_data("CA-live")) + store.record_turn( + { + "call_id": "CA-live", + "turn": { + "turn_index": 0, + "user_text": "hello", + "agent_text": "hi there", + }, + } + ) + active = store.get_active("CA-live") + assert active is not None + assert active["call_id"] == "CA-live" + assert len(active["turns"]) == 1 + assert active["turns"][0]["user_text"] == "hello" + + def test_get_active_returns_none_for_unknown_call(self) -> None: + store = MetricsStore() + assert store.get_active("missing") is None diff --git a/libraries/python/tests/unit/test_first_message_pacing.py b/libraries/python/tests/unit/test_first_message_pacing.py new file mode 100644 index 00000000..c820b9fa --- /dev/null +++ b/libraries/python/tests/unit/test_first_message_pacing.py @@ -0,0 +1,313 @@ +"""Unit tests for the firstMessage mark-gated paced sender (BUG #128). + +Pre-fix the firstMessage TTS chunks were pushed into the carrier WebSocket +as fast as the TTS provider yielded them. A barge-in mid-buffer issued +``send_clear``, but the WebSocket queue between the SDK and the carrier +held several seconds of media frames already, and the agent kept talking +on the user's earpiece until that drained. + +Post-fix the loop sends a mark after every chunk and awaits the oldest +mark once ``_FIRST_MESSAGE_MARK_WINDOW`` chunks are unconfirmed; +``_drain_pending_marks`` (called from the cancel path) resolves every +pending future so the waiting loop exits on the next tick. On Telnyx +(no mark concept) the loop falls back to a playout-time-based sleep so +the carrier buffer never grows beyond one chunk. +""" + +from __future__ import annotations + +import asyncio +import time + +import pytest + +from getpatter.stream_handler import AudioSender, PipelineStreamHandler + + +CHUNK_BYTES = 1280 # mirrors PipelineStreamHandler._PREWARM_CHUNK_BYTES + + +class _RecordingAudioSender(AudioSender): + """In-memory AudioSender that records every call for inspection.""" + + def __init__(self) -> None: + self.audio_chunks: list[bytes] = [] + self.marks: list[str] = [] + self.clears: int = 0 + + async def send_audio(self, pcm_audio: bytes) -> None: + self.audio_chunks.append(pcm_audio) + + async def send_clear(self) -> None: + self.clears += 1 + + async def send_mark(self, mark_name: str) -> None: + self.marks.append(mark_name) + + +def _make_handler( + *, for_twilio: bool = True +) -> tuple[PipelineStreamHandler, _RecordingAudioSender]: + """Build a PipelineStreamHandler shell without exercising __init__. + + Tests need only the paced-sender / on_mark / cancel surface — we don't + want to mock 30 unrelated dependencies (STT/TTS/metrics/etc.). + """ + handler = PipelineStreamHandler.__new__(PipelineStreamHandler) + sender = _RecordingAudioSender() + handler.audio_sender = sender + handler._is_speaking = True + handler._speaking_started_at = time.time() + handler._first_audio_sent_at = time.time() + handler._aec = None + handler._for_twilio = for_twilio + handler._pending_marks = [] + handler._first_message_mark_counter = 0 + handler.call_id = "call-test" + handler.metrics = None + return handler, sender + + +def _mark_first_audio_sent_noop(self: PipelineStreamHandler) -> None: + """No-op replacement for the real ``_mark_first_audio_sent`` so we don't + need to wire the per-turn metrics accumulator into the test fixture. + """ + return None + + +@pytest.fixture(autouse=True) +def _patch_mark_first(monkeypatch: pytest.MonkeyPatch) -> None: + monkeypatch.setattr( + PipelineStreamHandler, + "_mark_first_audio_sent", + _mark_first_audio_sent_noop, + ) + + +@pytest.fixture(autouse=True) +def _instant_playout_sleep(monkeypatch: pytest.MonkeyPatch) -> None: + """Replace asyncio.sleep with sleep(0) so the playout pacing in + ``_send_paced_first_message_bytes`` yields once instead of waiting + 40 ms per chunk. This lets mark-gating tests advance through multiple + chunks in a handful of ``asyncio.sleep(0)`` iterations without + waiting real time, while leaving ``asyncio.wait_for`` timeouts in + ``_wait_for_mark_window`` unaffected (they use the event-loop clock, + not asyncio.sleep). + """ + _real_sleep = asyncio.sleep + + async def _zero(secs: float) -> None: + await _real_sleep(0) + + monkeypatch.setattr(asyncio, "sleep", _zero) + + +@pytest.mark.unit +class TestFirstMessageMarkGatedPacing: + """BUG #128 regression coverage: firstMessage must be cancellable.""" + + async def test_caps_in_flight_at_window_and_bails_on_barge_in(self) -> None: + handler, sender = _make_handler(for_twilio=True) + # 4 chunks. Window=3, so chunks 1–3 send back-to-back and chunk 4 + # blocks on _wait_for_mark_window until either a mark echoes OR + # _drain_pending_marks (called from cancel) resolves the futures. + bytes_ = b"\x00" * (CHUNK_BYTES * 4) + + task = asyncio.create_task(handler._send_paced_first_message_bytes(bytes_)) + + # Yield enough so the loop sends the first three chunks and enters + # the window wait. + for _ in range(20): + await asyncio.sleep(0) + + assert len(sender.audio_chunks) == 3 + assert sender.marks == ["fm_1", "fm_2", "fm_3"] + assert len(handler._pending_marks) == 3 + + # Simulate the cancel side of a confirmed barge-in. ``send_clear`` is + # the canonical signal; ``_drain_pending_marks`` unblocks the + # waiting loop so it sees ``_is_speaking=False`` on the next tick. + handler._is_speaking = False + handler._drain_pending_marks() + await sender.send_clear() + + sent = await task + + assert sent is True + assert sender.clears == 1 + # Chunk 4 must NOT have hit the wire. + assert len(sender.audio_chunks) == 3 + + async def test_echoed_mark_slides_window_and_next_chunk_goes_out(self) -> None: + handler, sender = _make_handler(for_twilio=True) + bytes_ = b"\x00" * (CHUNK_BYTES * 4) + + task = asyncio.create_task(handler._send_paced_first_message_bytes(bytes_)) + + for _ in range(20): + await asyncio.sleep(0) + assert len(sender.audio_chunks) == 3 + assert sender.marks == ["fm_1", "fm_2", "fm_3"] + + # Twilio echoes chunk 1 → loop should advance to chunk 4. + await handler.on_mark("fm_1") + for _ in range(20): + await asyncio.sleep(0) + + assert len(sender.audio_chunks) == 4 + assert sender.marks == ["fm_1", "fm_2", "fm_3", "fm_4"] + + # Drain the rest so the loop completes naturally. + await handler.on_mark("fm_2") + await handler.on_mark("fm_3") + await handler.on_mark("fm_4") + await task + assert handler._pending_marks == [] + + async def test_telnyx_paces_via_playout_time_and_bails_on_cancel(self) -> None: + handler, sender = _make_handler(for_twilio=False) + # 4 chunks. Telnyx never sends marks — every iteration awaits a + # real ``asyncio.sleep`` keyed to chunk playout duration. + bytes_ = b"\x00" * (CHUNK_BYTES * 4) + + task = asyncio.create_task(handler._send_paced_first_message_bytes(bytes_)) + + # Yield enough so at least the first chunk hits the wire. + for _ in range(5): + await asyncio.sleep(0) + sent_before_cancel = len(sender.audio_chunks) + assert sent_before_cancel >= 1 + # Telnyx must never accumulate marks. + assert sender.marks == [] + assert handler._pending_marks == [] + + # Cancel mid-loop. + handler._is_speaking = False + handler._drain_pending_marks() + await sender.send_clear() + await task + + assert sender.clears == 1 + # No further chunks may go out after cancel. + assert len(sender.audio_chunks) == sent_before_cancel + + +@pytest.mark.unit +class TestOnMarkResolvesWaiters: + """``on_mark`` matches the FIFO entry and resolves all earlier ones too.""" + + async def test_echo_for_later_mark_resolves_earlier_waiters(self) -> None: + handler, _sender = _make_handler(for_twilio=True) + + # Manually queue three marks (skipping send_audio so we test the + # matching logic in isolation). + await handler._send_mark_awaitable() + await handler._send_mark_awaitable() + await handler._send_mark_awaitable() + assert [name for name, _ in handler._pending_marks] == ["fm_1", "fm_2", "fm_3"] + + await handler.on_mark("fm_2") + # fm_1 and fm_2 are drained; fm_3 stays pending. + assert [name for name, _ in handler._pending_marks] == ["fm_3"] + + +@pytest.mark.unit +class TestCleanupDrainsPendingMarks: + """Cleanup on abnormal call end (carrier WS drop / hangup mid + firstMessage) must resolve every pending mark future so the paced + send loop never leaves orphan ``asyncio.Future`` instances. + """ + + async def test_cleanup_drains_pending_marks(self) -> None: + handler, _sender = _make_handler(for_twilio=True) + # Wire enough stubs so PipelineStreamHandler.cleanup() does not + # crash. The actual stt/tts/remote_handler tear-down branches + # short-circuit on ``None``. + handler._barge_in_pending_task = None + handler._barge_in_pending_since = None + handler._stt_task = None + handler._stt = None + handler._tts = None + handler._remote_handler = None + handler._resampler_8k_to_16k = None + + # Queue three marks via the public send path then trigger + # cleanup to mimic an abnormal end mid-send. + await handler._send_mark_awaitable() + await handler._send_mark_awaitable() + await handler._send_mark_awaitable() + pending_futures = [fut for _name, fut in handler._pending_marks] + assert len(pending_futures) == 3 + assert all(not fut.done() for fut in pending_futures) + + await handler.cleanup() + + # Every queued future is resolved and the queue is empty. + assert handler._pending_marks == [] + assert all(fut.done() for fut in pending_futures) + + +@pytest.mark.unit +class TestFirstMessageMarkCounterReset: + """The ``_first_message_mark_counter`` must reset at the top of each + paced send AND on cleanup so a re-used handler instance never reuses + a stale ``fm_`` name across turns. + """ + + async def test_send_paced_resets_counter_between_consecutive_sends(self) -> None: + """Each ``_send_paced_first_message_bytes`` invocation re-starts + the ``fm_`` numbering at 1 — without the reset, the counter + would grow monotonically across turns and a stale echo for an + earlier turn's ``fm_N`` could match a mark name issued later. + """ + handler, sender = _make_handler(for_twilio=True) + bytes_ = b"\x00" * (CHUNK_BYTES * 2) + + # First send: two chunks ≤ window (3) so the loop yields after + # the first ``_wait_for_mark_window`` pre-check on chunk 3. + task1 = asyncio.create_task(handler._send_paced_first_message_bytes(bytes_)) + for _ in range(20): + await asyncio.sleep(0) + await handler.on_mark("fm_1") + await handler.on_mark("fm_2") + await task1 + assert handler._first_message_mark_counter == 2 + assert handler._pending_marks == [] + assert sender.marks == ["fm_1", "fm_2"] + + # Second send: counter must reset to 0 before iterating so the + # new sequence is fm_1, fm_2 — NOT fm_3, fm_4. + task2 = asyncio.create_task(handler._send_paced_first_message_bytes(bytes_)) + for _ in range(20): + await asyncio.sleep(0) + # New marks recorded by the sender are appended after the prior + # turn's two marks. + new_marks = sender.marks[2:] + assert new_marks == ["fm_1", "fm_2"] + assert handler._first_message_mark_counter == 2 + + await handler.on_mark("fm_1") + await handler.on_mark("fm_2") + await task2 + + async def test_cleanup_resets_counter(self) -> None: + """Cleanup must reset ``_first_message_mark_counter`` to 0 so a + re-used handler starts fresh on the next call. Defensive: the + per-send reset is the canonical path, but cleanup belt-and-braces + the cross-call boundary. + """ + handler, _sender = _make_handler(for_twilio=True) + handler._barge_in_pending_task = None + handler._barge_in_pending_since = None + handler._stt_task = None + handler._stt = None + handler._tts = None + handler._remote_handler = None + handler._resampler_8k_to_16k = None + + # Pretend a prior call left the counter at 7. + handler._first_message_mark_counter = 7 + + await handler.cleanup() + + assert handler._first_message_mark_counter == 0 diff --git a/libraries/python/tests/unit/test_metrics_store_hydrate.py b/libraries/python/tests/unit/test_metrics_store_hydrate.py index f54c0cba..447a7eeb 100644 --- a/libraries/python/tests/unit/test_metrics_store_hydrate.py +++ b/libraries/python/tests/unit/test_metrics_store_hydrate.py @@ -7,7 +7,7 @@ from __future__ import annotations import json -from datetime import datetime, timedelta, timezone +from datetime import datetime, timedelta from pathlib import Path import pytest @@ -67,9 +67,7 @@ def test_rebuilds_call_list_from_disk(tmp_path: Path) -> None: def test_idempotent_on_re_hydrate(tmp_path: Path) -> None: - _build_fixture( - tmp_path, [{"id": "CA-1", "iso": "2026-04-26T15:00:00.000Z"}] - ) + _build_fixture(tmp_path, [{"id": "CA-1", "iso": "2026-04-26T15:00:00.000Z"}]) store = MetricsStore() assert store.hydrate(str(tmp_path)) == 1 assert store.hydrate(str(tmp_path)) == 0 @@ -77,9 +75,7 @@ def test_idempotent_on_re_hydrate(tmp_path: Path) -> None: def test_tolerates_corrupt_metadata(tmp_path: Path) -> None: - _build_fixture( - tmp_path, [{"id": "CA-good", "iso": "2026-04-26T15:00:00.000Z"}] - ) + _build_fixture(tmp_path, [{"id": "CA-good", "iso": "2026-04-26T15:00:00.000Z"}]) bad_dir = tmp_path / "calls" / "2026" / "04" / "26" / "CA-bad" bad_dir.mkdir(parents=True, exist_ok=True) (bad_dir / "metadata.json").write_text("{ not valid json", encoding="utf-8") @@ -92,10 +88,7 @@ def test_tolerates_corrupt_metadata(tmp_path: Path) -> None: def test_respects_max_calls(tmp_path: Path) -> None: _build_fixture( tmp_path, - [ - {"id": f"CA-{i}", "iso": f"2026-04-26T15:0{i}:00.000Z"} - for i in range(7) - ], + [{"id": f"CA-{i}", "iso": f"2026-04-26T15:0{i}:00.000Z"} for i in range(7)], ) store = MetricsStore(max_calls=3) assert store.hydrate(str(tmp_path)) == 7 @@ -108,9 +101,7 @@ def test_respects_max_calls(tmp_path: Path) -> None: @pytest.mark.parametrize("invalid_name", ["not_numeric", ".DS_Store"]) def test_skips_non_numeric_directory_layers(tmp_path: Path, invalid_name: str) -> None: """Stray non-numeric YYYY/MM/DD entries must not break the walk.""" - _build_fixture( - tmp_path, [{"id": "CA-only", "iso": "2026-04-26T15:00:00.000Z"}] - ) + _build_fixture(tmp_path, [{"id": "CA-only", "iso": "2026-04-26T15:00:00.000Z"}]) (tmp_path / "calls" / invalid_name).mkdir(parents=True, exist_ok=True) store = MetricsStore() assert store.hydrate(str(tmp_path)) == 1 @@ -119,9 +110,7 @@ def test_skips_non_numeric_directory_layers(tmp_path: Path, invalid_name: str) - def test_skips_records_with_unparseable_started_at(tmp_path: Path) -> None: """A malformed ``started_at`` must NOT land in the store as epoch 0, which would corrupt every sort/range query that depends on it.""" - _build_fixture( - tmp_path, [{"id": "CA-good", "iso": "2026-04-26T15:00:00.000Z"}] - ) + _build_fixture(tmp_path, [{"id": "CA-good", "iso": "2026-04-26T15:00:00.000Z"}]) bad_dir = tmp_path / "calls" / "2026" / "04" / "26" / "CA-bad" bad_dir.mkdir(parents=True, exist_ok=True) (bad_dir / "metadata.json").write_text( @@ -168,3 +157,77 @@ def test_accepts_numeric_unix_seconds_timestamps(tmp_path: Path) -> None: listed = store.get_calls() assert listed[0]["started_at"] == 1745683200.0 assert listed[0]["ended_at"] == 1745683230.0 + + +def test_hydrate_lifts_top_level_cost_and_latency_into_metrics(tmp_path: Path) -> None: + """``CallLogger.log_call_end`` writes ``cost`` / ``latency`` / ``duration_ms`` + at the top of metadata.json (no ``metrics`` key). The hydrate path must + promote those into ``metrics`` so the dashboard renders cost and latency + instead of ``$0.00`` / ``—`` for hydrated calls. + """ + call_dir = tmp_path / "calls" / "2026" / "05" / "08" / "CA-real-shape" + call_dir.mkdir(parents=True, exist_ok=True) + (call_dir / "metadata.json").write_text( + json.dumps( + { + "schema_version": "1.0", + "call_id": "CA-real-shape", + "started_at": "2026-05-08T23:33:00.000Z", + "ended_at": "2026-05-08T23:33:57.000Z", + "duration_ms": 57400, + "status": "completed", + "caller": "", + "callee": "", + "telephony_provider": "twilio", + "provider_mode": "pipeline", + "agent": {"provider": "pipeline", "language": "en"}, + "turns": 9, + "cost": { + "stt": 0.001526, + "tts": 0.02988, + "llm": 0.000406, + "telephony": 0.0085, + "total": 0.040312, + }, + "latency": {"p50_ms": 2127.7, "p95_ms": 3461.7, "p99_ms": 3640.1}, + "error": None, + } + ), + encoding="utf-8", + ) + + store = MetricsStore() + assert store.hydrate(str(tmp_path)) == 1 + rec = store.get_calls()[0] + metrics = rec["metrics"] + assert metrics is not None + assert metrics["cost"]["total"] == pytest.approx(0.040312) + assert metrics["latency"]["p95_ms"] == pytest.approx(3461.7) + assert metrics["latency_avg"]["total_ms"] == pytest.approx(3461.7) + assert metrics["duration_seconds"] == pytest.approx(57.4) + assert metrics["telephony_provider"] == "twilio" + + +def test_hydrate_preserves_explicit_metrics_when_present(tmp_path: Path) -> None: + """If a metadata.json already has ``metrics`` (legacy or future shape) we + must NOT overwrite it with the top-level fallback. + """ + call_dir = tmp_path / "calls" / "2026" / "05" / "08" / "CA-explicit" + call_dir.mkdir(parents=True, exist_ok=True) + (call_dir / "metadata.json").write_text( + json.dumps( + { + "call_id": "CA-explicit", + "started_at": "2026-05-08T10:00:00Z", + "metrics": {"cost": {"total": 0.999}, "marker": "kept"}, + "cost": {"total": 0.001}, + "latency": {"p95_ms": 9999}, + } + ), + encoding="utf-8", + ) + store = MetricsStore() + assert store.hydrate(str(tmp_path)) == 1 + metrics = store.get_calls()[0]["metrics"] + assert metrics["marker"] == "kept" + assert metrics["cost"]["total"] == pytest.approx(0.999) diff --git a/libraries/python/tests/unit/test_metrics_unit.py b/libraries/python/tests/unit/test_metrics_unit.py index 30745d9e..7f9d0fd5 100644 --- a/libraries/python/tests/unit/test_metrics_unit.py +++ b/libraries/python/tests/unit/test_metrics_unit.py @@ -2,8 +2,6 @@ from __future__ import annotations -import time -from unittest.mock import patch import pytest @@ -196,6 +194,34 @@ def test_stt_audio_seconds_from_bytes(self) -> None: # Should have computed ~1 second of audio assert acc._total_stt_audio_seconds == pytest.approx(1.0, abs=0.01) + def test_twilio_stt_format_no_4x_overbill_regression(self) -> None: + """Regression guard: Twilio adapter must configure metrics with the + post-decode PCM16@16kHz format, not raw mulaw 8 kHz. + + Stream handler decodes inbound mulaw to PCM16@16 kHz before calling + ``add_stt_audio_bytes``. If metrics were left at ``(8000, 1)`` (mulaw), + a 60 s call would report 240 s of audio and bill Deepgram 4× the real + cost ($0.0192 vs $0.0048 at Nova-3's $0.0048/min). With the correct + ``(16000, 2)`` config — what the post-fix Twilio adapter installs — + the 1.92 MB byte count maps back to ~60 s and ~$0.0048. + """ + acc = _make_accumulator( + provider_mode="pipeline", + stt_provider="deepgram", + stt_model="nova-3", + ) + # Post-fix Twilio adapter installs this format. + acc.configure_stt_format(sample_rate=16000, bytes_per_sample=2) + + # 60 s of PCM16 @ 16 kHz = 60 * 16000 * 2 = 1_920_000 bytes. + acc.add_stt_audio_bytes(1_920_000) + metrics = acc.end_call() + + # NOT 240 s — that was the pre-fix 4× over-bill failure mode. + assert acc._total_stt_audio_seconds == pytest.approx(60.0, abs=0.01) + # Deepgram nova-3 default = $0.0048/min × 1 min = $0.0048. + assert metrics.cost.stt == pytest.approx(0.0048, abs=1e-6) + # --------------------------------------------------------------------------- # Cost calculation @@ -209,7 +235,7 @@ class TestCostCalculation: def test_pipeline_mode_cost(self) -> None: acc = _make_accumulator(provider_mode="pipeline") acc._total_stt_audio_seconds = 60.0 # 1 minute - acc._total_tts_characters = 1000 # 1k chars + acc._total_tts_characters = 1000 # 1k chars cost = acc._compute_cost(duration_seconds=60.0) assert cost.stt > 0 assert cost.tts > 0 @@ -235,7 +261,9 @@ def test_actual_telephony_cost_overrides_estimate(self) -> None: def test_actual_stt_cost_overrides_estimate(self) -> None: acc = _make_accumulator(provider_mode="pipeline") acc.set_actual_stt_cost(0.042) - acc._total_stt_audio_seconds = 600.0 # large value that would give different estimate + acc._total_stt_audio_seconds = ( + 600.0 # large value that would give different estimate + ) cost = acc._compute_cost(duration_seconds=60.0) assert cost.stt == pytest.approx(0.042, abs=1e-6) @@ -273,14 +301,18 @@ def test_end_call_no_turns(self) -> None: def test_realtime_usage_accumulation(self) -> None: acc = _make_accumulator(provider_mode="openai_realtime") - acc.record_realtime_usage({ - "input_token_details": {"audio_tokens": 100, "text_tokens": 50}, - "output_token_details": {"audio_tokens": 200, "text_tokens": 100}, - }) - acc.record_realtime_usage({ - "input_token_details": {"audio_tokens": 100, "text_tokens": 50}, - "output_token_details": {"audio_tokens": 200, "text_tokens": 100}, - }) + acc.record_realtime_usage( + { + "input_token_details": {"audio_tokens": 100, "text_tokens": 50}, + "output_token_details": {"audio_tokens": 200, "text_tokens": 100}, + } + ) + acc.record_realtime_usage( + { + "input_token_details": {"audio_tokens": 100, "text_tokens": 50}, + "output_token_details": {"audio_tokens": 200, "text_tokens": 100}, + } + ) assert acc._total_realtime_cost > 0 metrics = acc.end_call() assert metrics.cost.llm > 0 diff --git a/libraries/python/tests/unit/test_observability_attributes_unit.py b/libraries/python/tests/unit/test_observability_attributes_unit.py new file mode 100644 index 00000000..344f2ffc --- /dev/null +++ b/libraries/python/tests/unit/test_observability_attributes_unit.py @@ -0,0 +1,92 @@ +"""Unit tests for getpatter.observability.attributes — patter.* span helpers. + +Covers the public surface of the new attributes module: + +- ``patter_call_scope`` ContextVar lifecycle +- ``record_patter_attrs`` no-op when no scope is active +- ``record_patter_attrs`` stamps on the active span when a scope is active +- ``attach_span_exporter`` is idempotent on the same exporter object +- The Patter._attach_span_exporter public hook routes through correctly + +These tests avoid hitting any external service and use the in-memory +OTel test exporter so they remain fast and deterministic. +""" + +from __future__ import annotations + +import pytest + + +@pytest.mark.unit +class TestPatterCallScope: + """patter_call_scope binds call_id and side to the asyncio task tree.""" + + def test_scope_requires_non_empty_call_id(self) -> None: + from getpatter.observability.attributes import patter_call_scope + + with pytest.raises(ValueError): + with patter_call_scope(call_id=""): + pass + + def test_scope_round_trip_resets_contextvars(self) -> None: + from getpatter.observability.attributes import ( + _patter_call_id, + _patter_side, + patter_call_scope, + ) + + # Outside the scope the ContextVar is at its default (None / "uut"). + assert _patter_call_id.get() is None + assert _patter_side.get() == "uut" + + with patter_call_scope(call_id="CA000", side="driver"): + assert _patter_call_id.get() == "CA000" + assert _patter_side.get() == "driver" + + # After exit the ContextVar is reset. + assert _patter_call_id.get() is None + assert _patter_side.get() == "uut" + + +@pytest.mark.unit +class TestRecordPatterAttrs: + """record_patter_attrs is a safe no-op outside a scope.""" + + def test_no_scope_active_is_noop(self) -> None: + """Outside ``patter_call_scope`` the helper returns silently.""" + from getpatter.observability.attributes import record_patter_attrs + + # Must not raise even with no OTel scope; payload is dropped. + record_patter_attrs({"patter.cost.tts_chars": 42}) + + def test_inside_scope_no_otel_is_safe(self) -> None: + """Inside a scope, when OTel is not configured, the helper is still safe.""" + from getpatter.observability.attributes import ( + patter_call_scope, + record_patter_attrs, + ) + + with patter_call_scope(call_id="CA111"): + # Should not raise — the helper is defensive. + record_patter_attrs({"patter.cost.stt_seconds": 1.5}) + + +@pytest.mark.unit +class TestAttachSpanExporterPublic: + """Patter._attach_span_exporter is the public hook used by patter-agent-runner.""" + + def test_attach_span_exporter_stores_side_on_instance(self) -> None: + """Even without OTel SDK, the helper stamps ``_patter_side`` on the + Patter instance so downstream code (StreamHandler) can inherit it.""" + + # Build a minimal stand-in for Patter that exposes ``_patter_side``. + class _Stub: + _patter_side: str = "uut" + + from getpatter.observability.attributes import attach_span_exporter + + stub = _Stub() + # Pass an opaque exporter — when OTel SDK is missing the helper logs + # and returns; either way the side= arg must be stored on the stub. + attach_span_exporter(stub, exporter=object(), side="driver") + assert stub._patter_side == "driver" diff --git a/libraries/python/tests/unit/test_provider_warmup.py b/libraries/python/tests/unit/test_provider_warmup.py new file mode 100644 index 00000000..2778d002 --- /dev/null +++ b/libraries/python/tests/unit/test_provider_warmup.py @@ -0,0 +1,773 @@ +"""Unit tests for the concrete provider WebSocket / HTTP warmup overrides. + +Covers the per-provider ``warmup()`` overrides shipped on top of the no-op +default declared on :class:`STTProvider` / :class:`TTSProvider`. Each test +checks two invariants: + +* ``warmup()`` completes without raising (best-effort contract). +* When a provider opens a connection, it does NOT request any synthesis or + send any audio frames — billing-during-warmup must remain zero per the + per-provider docstrings. + +Tests use authentic real code paths — only the network boundary +(``websockets.connect`` / ``aiohttp.ws_connect`` / ``httpx`` / OpenAI +``response.create``) is mocked. See ``.claude/rules/authentic-tests.md``. +""" + +from __future__ import annotations + +import json +from unittest.mock import patch + +import pytest + +# --------------------------------------------------------------------------- +# Deepgram STT WS warmup +# --------------------------------------------------------------------------- + + +class _FakeAsyncWS: + """Minimal stand-in for a websockets ClientConnection used in warmup tests. + + Tracks ``send`` / ``recv`` / ``close`` calls so assertions can verify + no audio was sent and the socket was closed cleanly. + """ + + def __init__(self, *, recv_responses: list[str] | None = None) -> None: + self.send_calls: list[bytes | str] = [] + self.close_calls = 0 + self._recv_responses = list(recv_responses or []) + + async def send(self, payload: bytes | str) -> None: + self.send_calls.append(payload) + + async def recv(self) -> str: + if not self._recv_responses: + raise asyncio.TimeoutError + return self._recv_responses.pop(0) + + async def close(self) -> None: + self.close_calls += 1 + + +import asyncio # noqa: E402 — late import so _FakeAsyncWS can reference it + + +@pytest.mark.mocked +async def test_deepgram_stt_warmup_opens_and_closes_ws_without_audio() -> None: + """Deepgram warmup opens the WS, sleeps briefly, and closes — no audio.""" + from getpatter.providers.deepgram_stt import DeepgramSTT + + fake_ws = _FakeAsyncWS() + + async def fake_connect(*_args, **_kwargs): + return fake_ws + + stt = DeepgramSTT(api_key="test-key") + + with patch( + "getpatter.providers.deepgram_stt.websockets.connect", + side_effect=fake_connect, + ): + await stt.warmup() + + # WS opened and closed — no audio frames sent during warmup. + assert fake_ws.close_calls >= 1 + assert all(isinstance(call, str) for call in fake_ws.send_calls), ( + "warmup must never send binary audio frames (would consume billable seconds)" + ) + # Specifically: no binary chunks, and no protocol-level synthesis. + assert not any(isinstance(call, (bytes, bytearray)) for call in fake_ws.send_calls) + + +@pytest.mark.mocked +async def test_deepgram_stt_warmup_swallows_connect_errors() -> None: + """A network failure during warmup must not raise — best-effort contract.""" + from getpatter.providers.deepgram_stt import DeepgramSTT + + async def fake_connect(*_args, **_kwargs): + raise OSError("DNS down") + + stt = DeepgramSTT(api_key="test-key") + with patch( + "getpatter.providers.deepgram_stt.websockets.connect", + side_effect=fake_connect, + ): + # Must not raise. + await stt.warmup() + + +# --------------------------------------------------------------------------- +# Cartesia STT WS warmup +# --------------------------------------------------------------------------- + + +class _FakeAiohttpWS: + """Minimal stand-in for ``aiohttp.ClientWebSocketResponse``.""" + + def __init__(self) -> None: + self.send_bytes_calls: list[bytes] = [] + self.send_str_calls: list[str] = [] + self.closed = False + + async def send_bytes(self, data: bytes) -> None: + self.send_bytes_calls.append(data) + + async def send_str(self, payload: str) -> None: + self.send_str_calls.append(payload) + + async def close(self) -> None: + self.closed = True + + +class _FakeAiohttpSession: + def __init__(self, ws: _FakeAiohttpWS) -> None: + self._ws = ws + self.closed = False + + async def ws_connect(self, *_args, **_kwargs) -> _FakeAiohttpWS: + return self._ws + + async def close(self) -> None: + self.closed = True + + +@pytest.mark.mocked +async def test_cartesia_stt_warmup_opens_and_closes_ws_without_audio() -> None: + """Cartesia STT warmup opens WS, idles, closes — no audio bytes.""" + from getpatter.providers import cartesia_stt as mod + from getpatter.providers.cartesia_stt import CartesiaSTT + + fake_ws = _FakeAiohttpWS() + fake_session = _FakeAiohttpSession(fake_ws) + + stt = CartesiaSTT(api_key="test-key") + with patch.object(mod.aiohttp, "ClientSession", return_value=fake_session): + await stt.warmup() + + assert fake_ws.closed + assert fake_session.closed + # No audio frames sent during warmup — billing protection. + assert fake_ws.send_bytes_calls == [] + + +@pytest.mark.mocked +async def test_cartesia_stt_warmup_swallows_connect_errors() -> None: + from getpatter.providers import cartesia_stt as mod + from getpatter.providers.cartesia_stt import CartesiaSTT + + class _BoomSession: + async def ws_connect(self, *_a, **_k): + raise ConnectionError("network down") + + async def close(self) -> None: + return None + + stt = CartesiaSTT(api_key="test-key") + with patch.object(mod.aiohttp, "ClientSession", return_value=_BoomSession()): + await stt.warmup() # must not raise + + +@pytest.mark.mocked +async def test_cartesia_stt_warmup_handshake_error_does_not_leak_api_key( + caplog, +) -> None: + """Regression: a 401/403 from Cartesia must NOT log the request URL. + + Cartesia auth uses ``?api_key=...`` in the URL. The default + ``aiohttp.WSServerHandshakeError.__str__`` includes the URL, which + means a generic ``logger.debug("warmup failed: %s", exc)`` would + write the API key straight into application logs. + """ + import logging + + from getpatter.providers import cartesia_stt as mod + from getpatter.providers.cartesia_stt import CartesiaSTT + + secret_key = "ck_secret_THIS_MUST_NEVER_LEAK" + + class _HandshakeFailSession: + async def ws_connect(self, *_a, **_k): + # Build a real WSServerHandshakeError so the regression test + # exercises the same exception class as production. + from aiohttp import WSServerHandshakeError + from aiohttp.client_reqrep import RequestInfo + from yarl import URL + + url = URL(f"wss://api.cartesia.ai/stt/websocket?api_key={secret_key}") + req_info = RequestInfo( + url=url, + method="GET", + headers={}, # type: ignore[arg-type] + real_url=url, + ) + raise WSServerHandshakeError( + request_info=req_info, + history=(), + status=401, + message="Unauthorized", + headers=None, + ) + + async def close(self) -> None: + return None + + stt = CartesiaSTT(api_key=secret_key) + caplog.set_level(logging.DEBUG, logger="getpatter") + with patch.object( + mod.aiohttp, "ClientSession", return_value=_HandshakeFailSession() + ): + await stt.warmup() # must not raise + + # The API key must not appear in any captured log message. + for rec in caplog.records: + assert secret_key not in rec.getMessage(), ( + f"API key leaked in log: {rec.getMessage()!r}" + ) + assert "api_key=" not in rec.getMessage(), ( + f"URL with api_key= leaked in log: {rec.getMessage()!r}" + ) + # And we should still log SOMETHING — namely the HTTP status — so + # operators know the warmup failed and why. + assert any( + "HTTP 401" in rec.getMessage() or "401" in rec.getMessage() + for rec in caplog.records + ), ( + "expected status code in log; got: " + f"{[rec.getMessage() for rec in caplog.records]}" + ) + + +# --------------------------------------------------------------------------- +# AssemblyAI STT WS warmup +# --------------------------------------------------------------------------- + + +@pytest.mark.mocked +async def test_assemblyai_stt_warmup_opens_and_closes_ws_without_audio() -> None: + """AssemblyAI warmup opens WS, sends Terminate (no audio), closes.""" + from getpatter.providers import assemblyai_stt as mod + from getpatter.providers.assemblyai_stt import AssemblyAISTT + + fake_ws = _FakeAiohttpWS() + fake_session = _FakeAiohttpSession(fake_ws) + + stt = AssemblyAISTT(api_key="test-key") + with patch.object(mod.aiohttp, "ClientSession", return_value=fake_session): + await stt.warmup() + + assert fake_ws.closed + assert fake_session.closed + # No audio frames sent during warmup. + assert fake_ws.send_bytes_calls == [] + # Terminate frame is fine — it's a control message, not audio. + if fake_ws.send_str_calls: + for payload in fake_ws.send_str_calls: + parsed = json.loads(payload) + assert parsed.get("type") == "Terminate" + + +@pytest.mark.mocked +async def test_assemblyai_stt_warmup_swallows_connect_errors() -> None: + from getpatter.providers import assemblyai_stt as mod + from getpatter.providers.assemblyai_stt import AssemblyAISTT + + class _BoomSession: + async def ws_connect(self, *_a, **_k): + raise ConnectionError("network down") + + async def close(self) -> None: + return None + + stt = AssemblyAISTT(api_key="test-key") + with patch.object(mod.aiohttp, "ClientSession", return_value=_BoomSession()): + await stt.warmup() # must not raise + + +@pytest.mark.mocked +async def test_assemblyai_stt_warmup_handshake_error_does_not_leak_api_key( + caplog, +) -> None: + """Regression: a 401/403 from AssemblyAI must NOT log the request URL. + + AssemblyAI auth supports ``?token=...`` in the URL when + ``use_query_token=True``. The default + ``aiohttp.WSServerHandshakeError.__str__`` includes the URL, which + means a generic ``logger.debug("warmup failed: %s", exc)`` would + write the API key straight into application logs. + """ + import logging + + from getpatter.providers import assemblyai_stt as mod + from getpatter.providers.assemblyai_stt import AssemblyAISTT + + secret_key = "aai_secret_THIS_MUST_NEVER_LEAK" + + class _HandshakeFailSession: + async def ws_connect(self, *_a, **_k): + from aiohttp import WSServerHandshakeError + from aiohttp.client_reqrep import RequestInfo + from yarl import URL + + url = URL(f"wss://streaming.assemblyai.com/v3/ws?token={secret_key}") + req_info = RequestInfo( + url=url, + method="GET", + headers={}, # type: ignore[arg-type] + real_url=url, + ) + raise WSServerHandshakeError( + request_info=req_info, + history=(), + status=401, + message="Unauthorized", + headers=None, + ) + + async def close(self) -> None: + return None + + stt = AssemblyAISTT(api_key=secret_key, use_query_token=True) + caplog.set_level(logging.DEBUG, logger="getpatter") + with patch.object( + mod.aiohttp, "ClientSession", return_value=_HandshakeFailSession() + ): + await stt.warmup() # must not raise + + for rec in caplog.records: + assert secret_key not in rec.getMessage(), ( + f"API key leaked in log: {rec.getMessage()!r}" + ) + assert "token=" not in rec.getMessage(), ( + f"URL with token= leaked in log: {rec.getMessage()!r}" + ) + assert any( + "HTTP 401" in rec.getMessage() or "401" in rec.getMessage() + for rec in caplog.records + ), ( + "expected status code in log; got: " + f"{[rec.getMessage() for rec in caplog.records]}" + ) + + +# --------------------------------------------------------------------------- +# ElevenLabs WS TTS warmup +# --------------------------------------------------------------------------- + + +@pytest.mark.mocked +async def test_elevenlabs_ws_tts_warmup_opens_sends_keepalive_closes() -> None: + """ElevenLabs WS TTS warmup: opens WS, sends single-space keepalive, closes. + + Specifically MUST NOT send any text + flush:true (which would commit a + synthesis and consume billable characters). + """ + from getpatter.providers import elevenlabs_ws_tts as mod + from getpatter.providers.elevenlabs_ws_tts import ElevenLabsWebSocketTTS + + fake_ws = _FakeAsyncWS() + + async def fake_connect(*_args, **_kwargs): + return fake_ws + + tts = ElevenLabsWebSocketTTS(api_key="test-key") + with patch.object(mod.websockets, "connect", side_effect=fake_connect): + await tts.warmup() + + assert fake_ws.close_calls >= 1 + # Inspect every send during warmup: must be either the single-space + # keepalive `{"text": " "}` OR nothing else. Specifically NO `flush:true`. + for raw_payload in fake_ws.send_calls: + # Should be string / json + if isinstance(raw_payload, (bytes, bytearray)): + raise AssertionError("warmup sent binary frame — must be JSON only") + msg = json.loads(raw_payload) + # No `flush: true` (would commit synthesis and bill characters). + assert msg.get("flush") is not True, ( + f"warmup must not commit synthesis (flush:true). saw: {msg}" + ) + # Text must be empty/space — no real transcript. + text = msg.get("text", "") + assert text.strip() == "", f"warmup sent non-empty text: {text!r}" + + +@pytest.mark.mocked +async def test_elevenlabs_ws_tts_warmup_swallows_connect_errors() -> None: + from getpatter.providers import elevenlabs_ws_tts as mod + from getpatter.providers.elevenlabs_ws_tts import ElevenLabsWebSocketTTS + + async def fake_connect(*_args, **_kwargs): + raise OSError("DNS down") + + tts = ElevenLabsWebSocketTTS(api_key="test-key") + with patch.object(mod.websockets, "connect", side_effect=fake_connect): + await tts.warmup() # must not raise + + +@pytest.mark.mocked +async def test_elevenlabs_ws_warmup_bos_frame_matches_live_synthesize() -> None: + """Regression: warmup BOS bytes must equal synthesize() BOS bytes. + + If the warmup primer differs from the production BOS, ElevenLabs may + instantiate a different per-session worker for the warm path vs the + live path, defeating the warmup goal entirely. This test captures + both BOS frames and asserts they're byte-identical. + """ + from getpatter.providers import elevenlabs_ws_tts as mod + from getpatter.providers.elevenlabs_ws_tts import ElevenLabsWebSocketTTS + + # Configure with non-default voice_settings + auto_mode=False + + # chunk_length_schedule so the BOS frame carries every optional field. + tts = ElevenLabsWebSocketTTS( + api_key="test-key", + voice_settings={"stability": 0.7, "similarity_boost": 0.8}, + auto_mode=False, + chunk_length_schedule=[120, 160, 250, 290], + ) + + # --- Capture warmup BOS --- + warmup_ws = _FakeAsyncWS() + + async def fake_warmup_connect(*_args, **_kwargs): + return warmup_ws + + with patch.object(mod.websockets, "connect", side_effect=fake_warmup_connect): + await tts.warmup() + + warmup_bos_bytes: bytes | None = None + for payload in warmup_ws.send_calls: + # First send is the BOS frame. + if isinstance(payload, str): + warmup_bos_bytes = payload.encode("utf-8") + break + assert warmup_bos_bytes is not None, "warmup did not send any frame" + + # --- Capture synthesize BOS --- + class _SynthesizeFakeWS: + """Fake WS for synthesize() that yields one final-marker frame so + the generator ends quickly without any real audio.""" + + def __init__(self) -> None: + self.send_calls: list[str | bytes] = [] + self._recv_calls = 0 + + async def send(self, payload: str | bytes) -> None: + self.send_calls.append(payload) + + async def recv(self) -> str: + self._recv_calls += 1 + # First recv: send isFinal=True so generator returns immediately. + if self._recv_calls == 1: + return json.dumps({"isFinal": True}) + await asyncio.sleep(0) + raise asyncio.TimeoutError + + async def close(self) -> None: + pass + + synth_ws = _SynthesizeFakeWS() + + async def fake_synth_connect(*_args, **_kwargs): + return synth_ws + + with patch.object(mod.websockets, "connect", side_effect=fake_synth_connect): + gen = tts.synthesize("hello") + # Drain the generator — it should exit on the isFinal frame. + async for _chunk in gen: + pass + + synth_bos_bytes: bytes | None = None + for payload in synth_ws.send_calls: + if isinstance(payload, str): + synth_bos_bytes = payload.encode("utf-8") + break + assert synth_bos_bytes is not None, "synthesize did not send any frame" + + # The BOS bytes must be byte-identical so ElevenLabs picks the same + # per-session worker for warm and live. + assert warmup_bos_bytes == synth_bos_bytes, ( + f"BOS drift: warmup={warmup_bos_bytes!r}, synthesize={synth_bos_bytes!r}" + ) + # And specifically: must NOT include flush:true (would commit synthesis). + parsed = json.loads(warmup_bos_bytes.decode("utf-8")) + assert parsed.get("flush") is not True + assert parsed.get("text", "").strip() == "" + + +# --------------------------------------------------------------------------- +# Cartesia TTS HTTP warmup +# --------------------------------------------------------------------------- + + +@pytest.mark.mocked +async def test_cartesia_tts_warmup_issues_get_to_voices_endpoint() -> None: + """Cartesia TTS HTTP warmup issues GET /voices — no synthesis POST.""" + from getpatter.providers.cartesia_tts import CartesiaTTS + + captured: dict[str, object] = {} + + class _FakeResp: + async def __aenter__(self) -> "_FakeResp": + return self + + async def __aexit__(self, *_a) -> None: + return None + + async def read(self) -> bytes: + return b"" + + class _FakeSession: + def get(self, url: str, **kwargs: object) -> _FakeResp: + captured["url"] = url + captured["kwargs"] = kwargs + return _FakeResp() + + def post(self, *_a, **_k) -> _FakeResp: + captured["post_called"] = True + return _FakeResp() + + async def close(self) -> None: + return None + + tts = CartesiaTTS(api_key="test-key", session=_FakeSession()) # type: ignore[arg-type] + await tts.warmup() + + # GET landed on the /voices endpoint, not /tts/bytes (which would bill). + assert "voices" in str(captured.get("url", "")) + assert "post_called" not in captured, "warmup must not POST /tts/bytes" + + +@pytest.mark.mocked +async def test_cartesia_tts_warmup_swallows_errors() -> None: + from getpatter.providers.cartesia_tts import CartesiaTTS + + class _BoomSession: + def get(self, *_a, **_k): + raise RuntimeError("DNS down") + + async def close(self) -> None: + return None + + tts = CartesiaTTS(api_key="test-key", session=_BoomSession()) # type: ignore[arg-type] + await tts.warmup() # must not raise + + +# --------------------------------------------------------------------------- +# Inworld TTS HTTP warmup +# --------------------------------------------------------------------------- + + +@pytest.mark.mocked +async def test_inworld_tts_warmup_issues_get_voices_request() -> None: + """Inworld TTS warmup issues GET /tts/v1/voices — 2xx, never synthesises. + + Earlier revisions used HEAD against the POST-only streaming endpoint, + which returned 405. The new path uses the documented voices metadata + GET so the response is 2xx and no 405s are spammed into audit logs. + """ + from getpatter.providers.inworld_tts import InworldTTS + + captured: dict[str, object] = {} + + class _FakeResp: + status = 200 + + async def __aenter__(self) -> "_FakeResp": + return self + + async def __aexit__(self, *_a) -> None: + return None + + async def read(self) -> bytes: + return b"" + + class _FakeSession: + def get(self, url: str, **kwargs: object) -> _FakeResp: + captured["url"] = url + captured["method"] = "GET" + return _FakeResp() + + def post(self, *_a, **_k) -> _FakeResp: + captured["post_called"] = True + return _FakeResp() + + def head(self, *_a, **_k) -> _FakeResp: + captured["head_called"] = True + return _FakeResp() + + async def close(self) -> None: + return None + + tts = InworldTTS(auth_token="test-token", session=_FakeSession()) # type: ignore[arg-type] + await tts.warmup() + + assert captured.get("method") == "GET" + # URL must point at the voices metadata endpoint, not the + # POST-only streaming endpoint (which would have returned 405). + assert "/tts/v1/voices" in str(captured.get("url", "")) + assert "voice:stream" not in str(captured.get("url", "")), ( + "warmup must not target the POST-only streaming endpoint" + ) + assert "post_called" not in captured, "warmup must not POST the synth endpoint" + assert "head_called" not in captured, "warmup must not HEAD (returns 405)" + # Response status must be 2xx (verified by the fake responding with 200). + # The implementation does not surface the status, but the test + # confirms the call lands on a 2xx-returning route by asserting the URL. + + +@pytest.mark.mocked +async def test_inworld_tts_warmup_swallows_errors() -> None: + from getpatter.providers.inworld_tts import InworldTTS + + class _BoomSession: + def get(self, *_a, **_k): + raise RuntimeError("DNS down") + + async def close(self) -> None: + return None + + tts = InworldTTS(auth_token="test-token", session=_BoomSession()) # type: ignore[arg-type] + await tts.warmup() # must not raise + + +# --------------------------------------------------------------------------- +# OpenAI Realtime warmup (session.update — billing-safe, no response.create) +# --------------------------------------------------------------------------- + + +@pytest.mark.mocked +async def test_openai_realtime_warmup_sends_session_update_only() -> None: + """OpenAI Realtime warmup sends session.update + waits for session.updated. + + Critically: must NOT send ``response.create`` — that field is not in + the OpenAI Realtime schema and either (a) bills tokens for a real + response or (b) returns ``invalid_request_error``. Both are wrong. + + Must also NOT send ``input_audio_buffer.append`` (would consume + billable audio). + """ + from getpatter.providers import openai_realtime as mod + from getpatter.providers.openai_realtime import OpenAIRealtimeAdapter + + sent: list[str] = [] + + class _FakeWS: + def __init__(self) -> None: + # Pre-canned server frames: session.created → session.updated. + self._recv_queue = [ + json.dumps({"type": "session.created"}), + json.dumps({"type": "session.updated"}), + ] + self.closed = False + + async def send(self, payload: str) -> None: + sent.append(payload) + + async def recv(self) -> str: + if not self._recv_queue: + # Simulate a server idle — give the warmup time to time out. + await asyncio.sleep(0) + raise asyncio.TimeoutError + return self._recv_queue.pop(0) + + async def close(self) -> None: + self.closed = True + + fake_ws = _FakeWS() + + async def fake_connect(*_args, **_kwargs): + return fake_ws + + adapter = OpenAIRealtimeAdapter( + api_key="sk-test", + voice="alloy", + instructions="You are a test assistant.", + ) + with patch.object(mod.websockets, "connect", side_effect=fake_connect): + await adapter.warmup() + + # Must NOT send response.create — the field is not in the OpenAI + # Realtime schema and is billing-unsafe. + parsed = [json.loads(s) for s in sent] + for p in parsed: + assert p.get("type") != "response.create", ( + f"warmup must not invoke response.create — schema-invalid and " + f"billing-unsafe. saw: {p}" + ) + assert p.get("type") != "input_audio_buffer.append", ( + "warmup must not send audio — would consume billable seconds" + ) + # Must send exactly one session.update with the production fields. + updates = [p for p in parsed if p.get("type") == "session.update"] + assert len(updates) == 1, f"expected one session.update, got: {parsed}" + session = updates[0]["session"] + # Production fields must be primed identically to ``connect()`` so the + # upstream session state is warmed for the real call. + for required in ( + "input_audio_format", + "output_audio_format", + "voice", + "instructions", + "turn_detection", + "input_audio_transcription", + ): + assert required in session, f"session.update missing {required!r}: {session}" + assert session["voice"] == "alloy" + assert session["instructions"] == "You are a test assistant." + assert fake_ws.closed + + +@pytest.mark.mocked +async def test_openai_realtime_warmup_does_not_send_response_create() -> None: + """Regression: warmup never sends response.create — schema-invalid and unsafe.""" + from getpatter.providers import openai_realtime as mod + from getpatter.providers.openai_realtime import OpenAIRealtimeAdapter + + sent: list[str] = [] + + class _FakeWS: + def __init__(self) -> None: + self._recv_queue = [ + json.dumps({"type": "session.created"}), + json.dumps({"type": "session.updated"}), + ] + self.closed = False + + async def send(self, payload: str) -> None: + sent.append(payload) + + async def recv(self) -> str: + if not self._recv_queue: + await asyncio.sleep(0) + raise asyncio.TimeoutError + return self._recv_queue.pop(0) + + async def close(self) -> None: + self.closed = True + + fake_ws = _FakeWS() + + async def fake_connect(*_args, **_kwargs): + return fake_ws + + adapter = OpenAIRealtimeAdapter(api_key="sk-test") + with patch.object(mod.websockets, "connect", side_effect=fake_connect): + await adapter.warmup() + + for raw in sent: + assert "response.create" not in raw, ( + f"warmup must not send response.create — saw: {raw}" + ) + + +@pytest.mark.mocked +async def test_openai_realtime_warmup_swallows_connect_errors() -> None: + from getpatter.providers import openai_realtime as mod + from getpatter.providers.openai_realtime import OpenAIRealtimeAdapter + + async def fake_connect(*_args, **_kwargs): + raise OSError("DNS down") + + adapter = OpenAIRealtimeAdapter(api_key="sk-test") + with patch.object(mod.websockets, "connect", side_effect=fake_connect): + await adapter.warmup() # must not raise diff --git a/libraries/python/tests/unit/test_server_unit.py b/libraries/python/tests/unit/test_server_unit.py index eb8e943c..8b27a7d2 100644 --- a/libraries/python/tests/unit/test_server_unit.py +++ b/libraries/python/tests/unit/test_server_unit.py @@ -203,6 +203,57 @@ async def test_no_user_callback_still_calls_store(self) -> None: store.record_call_start.assert_called_once() + @pytest.mark.asyncio + async def test_call_log_start_pulls_caller_from_active_record( + self, tmp_path + ) -> None: + """Bug B regression: outbound calls have empty caller/callee in the + on_call_start data (the inline TwiML for outbound has no Stream + ```` entries, so the WS query string is empty). The + ``_on_call_start`` wrapper must pull the real numbers from the + in-memory store (populated at dial time by ``record_call_initiated``) + before persisting metadata.json. Otherwise every outbound call's + on-disk metadata has ``caller="" callee=""``. + """ + import json + + from getpatter.dashboard.store import MetricsStore + from getpatter.services.call_log import CallLogger + + srv = _make_server() + srv._metrics_store = MetricsStore() + # Pre-register the call as record_call_initiated would. + srv._metrics_store.record_call_initiated( + { + "call_id": "CA-outbound", + "caller": "+15551112222", + "callee": "+15553334444", + "direction": "outbound", + } + ) + # Real on-disk CallLogger so we can read back metadata.json. + srv._call_logger = CallLogger(tmp_path) + + on_start, _, _ = srv._wrap_callbacks() + # Simulate the bridge's on_call_start payload for an outbound call: + # the WS query string was empty so caller/callee are blank. + await on_start( + { + "call_id": "CA-outbound", + "caller": "", + "callee": "", + "direction": "outbound", + "telephony_provider": "twilio", + } + ) + + meta_paths = list(tmp_path.glob("calls/*/*/*/CA-outbound/metadata.json")) + assert len(meta_paths) == 1 + payload = json.loads(meta_paths[0].read_text("utf-8")) + # Phone redact mode default is "mask" — last-4 visible. + assert payload["caller"].endswith("2222") + assert payload["callee"].endswith("4444") + # --------------------------------------------------------------------------- # _create_app — route registration diff --git a/libraries/python/tests/unit/test_silero_vad.py b/libraries/python/tests/unit/test_silero_vad.py index 0a788255..31a81e35 100644 --- a/libraries/python/tests/unit/test_silero_vad.py +++ b/libraries/python/tests/unit/test_silero_vad.py @@ -215,6 +215,36 @@ async def test_process_frame_after_close_raises() -> None: await vad.process_frame(_silence_pcm(512), sample_rate=16000) +# --------------------------------------------------------------------------- +# reset() — one-shot barge-in fix +# --------------------------------------------------------------------------- + + +async def test_reset_returns_vad_to_silence() -> None: + """After reset() the VAD must emit a FRESH speech_start for the next + utterance — without it, the persisted ``_pub_speaking=True`` state would + suppress the transition and barge-in would feel one-shot.""" + vad, _ = _build_vad( + probs=[0.95, 0.95, 0.95, 0.95, 0.95, 0.95, 0.95, 0.95], + min_speech_duration=0.032, + ) + first = await vad.process_frame( + _sine_pcm(512, sample_rate=16000), sample_rate=16000 + ) + assert first is not None and first.type == "speech_start" + vad.reset() + second = await vad.process_frame( + _sine_pcm(512, sample_rate=16000), sample_rate=16000 + ) + assert second is not None and second.type == "speech_start" + + +async def test_reset_after_close_is_noop() -> None: + vad, _ = _build_vad(probs=[0.0]) + await vad.close() + vad.reset() # must not raise + + # --------------------------------------------------------------------------- # Integration (skipped by default — requires bundled model and onnxruntime) # --------------------------------------------------------------------------- diff --git a/libraries/python/tests/unit/test_stream_handler_unit.py b/libraries/python/tests/unit/test_stream_handler_unit.py index 45350558..92f9a533 100644 --- a/libraries/python/tests/unit/test_stream_handler_unit.py +++ b/libraries/python/tests/unit/test_stream_handler_unit.py @@ -470,9 +470,13 @@ async def test_barge_in_suppressed_during_aec_warmup(self) -> None: handler._aec = object() # Emulate ``_begin_speaking`` having just run — agent has been # speaking for less than the gate. - handler._speaking_started_at = time.time() - ( - MIN_AGENT_SPEAKING_S_BEFORE_BARGE_IN_AEC / 2 - ) + # First audio chunk reached the wire at ``half_gate`` ago — so the + # gate measured from ``_first_audio_sent_at`` is still inside the + # warmup window (post-fix anchor; was anchored on + # ``_speaking_started_at`` pre-0.6.2). + half_gate = MIN_AGENT_SPEAKING_S_BEFORE_BARGE_IN_AEC / 2 + handler._speaking_started_at = time.time() - half_gate + handler._first_audio_sent_at = time.time() - half_gate await handler._handle_barge_in( Transcript(text="hold on", is_final=True, speech_final=True) @@ -503,9 +507,9 @@ async def test_barge_in_fires_after_warmup_window(self) -> None: handler.audio_sender.send_clear = AsyncMock() handler._llm_cancel_event = asyncio.Event() handler._aec = object() - handler._speaking_started_at = time.time() - ( - MIN_AGENT_SPEAKING_S_BEFORE_BARGE_IN_AEC + 0.1 - ) + past_gate = MIN_AGENT_SPEAKING_S_BEFORE_BARGE_IN_AEC + 0.1 + handler._speaking_started_at = time.time() - past_gate + handler._first_audio_sent_at = time.time() - past_gate await handler._handle_barge_in( Transcript(text="hold on", is_final=True, speech_final=True) @@ -517,7 +521,7 @@ async def test_barge_in_fires_after_warmup_window(self) -> None: async def test_barge_in_fires_at_400ms_when_aec_off(self) -> None: """The bug fix: on PSTN deployments AEC is OFF and the gate - collapses to 0.25 s anti-flicker. A user saying "stop" 400 ms + collapses to 0.1 s anti-flicker. A user saying "stop" 400 ms into the agent's turn must cancel the agent — pre-fix this was silently suppressed by the hardcoded 1.0 s gate. """ @@ -535,20 +539,21 @@ async def test_barge_in_fires_at_400ms_when_aec_off(self) -> None: # AEC OFF (PSTN default) — gate is 0.25 s. handler._aec = None handler._speaking_started_at = time.time() - 0.4 + handler._first_audio_sent_at = time.time() - 0.4 await handler._handle_barge_in( Transcript(text="stop", is_final=True, speech_final=True) ) assert handler._llm_cancel_event.is_set(), ( - "barge-in must fire on PSTN at 400 ms — past the 0.25 s anti-flicker gate" + "barge-in must fire on PSTN at 400 ms — past the 0.1 s anti-flicker gate" ) async def test_barge_in_suppressed_within_anti_flicker_when_aec_off( self, ) -> None: - """Anti-flicker side: even with AEC off, sub-100 ms blips - (cough, click, line noise) are still suppressed — the 0.25 s + """Anti-flicker side: even with AEC off, sub-50 ms blips + (cough, click, line noise) are still suppressed — the 0.1 s gate stays in place.""" from getpatter.stream_handler import PipelineStreamHandler from getpatter.providers.base import Transcript @@ -562,14 +567,51 @@ async def test_barge_in_suppressed_within_anti_flicker_when_aec_off( handler.audio_sender.send_clear = AsyncMock() handler._llm_cancel_event = asyncio.Event() handler._aec = None - handler._speaking_started_at = time.time() - 0.1 + handler._speaking_started_at = time.time() - 0.05 + handler._first_audio_sent_at = time.time() - 0.05 # 50 ms — inside 0.1 s gate await handler._handle_barge_in( Transcript(text="stop", is_final=True, speech_final=True) ) assert not handler._llm_cancel_event.is_set(), ( - "barge-in must be suppressed within the 0.25 s anti-flicker window" + "barge-in must be suppressed within the 0.1 s anti-flicker window" + ) + assert handler._is_speaking is True + + async def test_barge_in_suppressed_before_first_audio_emitted(self) -> None: + """0.6.2 fix: ElevenLabs (and other cloud TTS) take 200-700 ms to + emit the first byte. While ``_begin_speaking`` has fired but the + first chunk has NOT yet hit the wire, VAD picking up background + noise ("hello?", breath, room ambience) must NOT trigger a + self-cancel. Pre-fix, a 250 ms anti-flicker gate measured from + ``_begin_speaking`` expired BEFORE TTS emitted any audio, + cancelling the agent's first turn before a single byte left. + """ + from getpatter.stream_handler import PipelineStreamHandler + from getpatter.providers.base import Transcript + import time + + handler = object.__new__(PipelineStreamHandler) + handler._is_speaking = True + handler.metrics = None + handler.call_id = "test-call" + handler.audio_sender = MagicMock() + handler.audio_sender.send_clear = AsyncMock() + handler._llm_cancel_event = asyncio.Event() + handler._aec = None # PSTN default, no AEC warmup gate + # ``_begin_speaking`` ran 500 ms ago — well past the 250 ms gate + # that pre-fix would let barge-in through. But TTS still hasn't + # emitted the first chunk (cloud provider first-byte latency). + handler._speaking_started_at = time.time() - 0.5 + handler._first_audio_sent_at = None + + await handler._handle_barge_in( + Transcript(text="hello?", is_final=True, speech_final=True) + ) + + assert not handler._llm_cancel_event.is_set(), ( + "barge-in must be suppressed until at least one TTS chunk has hit the wire" ) assert handler._is_speaking is True diff --git a/libraries/python/tests/unit/test_tts_facade_language.py b/libraries/python/tests/unit/test_tts_facade_language.py index d183da56..713956fa 100644 --- a/libraries/python/tests/unit/test_tts_facade_language.py +++ b/libraries/python/tests/unit/test_tts_facade_language.py @@ -39,15 +39,57 @@ def test_elevenlabs_facade_forwards_voice_settings() -> None: @pytest.mark.unit def test_elevenlabs_facade_defaults_keep_provider_defaults() -> None: - """Backward-compat: omitting the new kwargs leaves the provider defaults.""" + """Backward-compat: omitting the new kwargs leaves the provider defaults. + + Since 0.6.1 the ``elevenlabs.TTS`` facade defaults to the WebSocket + streaming transport (provider_key ``"elevenlabs_ws"``). The WS provider + does not chunk on the client side — chunking is driven by + ``chunk_length_schedule`` server-side — so ``chunk_size`` is no longer + a meaningful attribute. The kwarg remains accepted as a no-op for + backward compatibility with REST-era call sites; see + ``test_elevenlabs_rest_tts_preserves_chunk_size_default`` for the + explicit-REST equivalent of the legacy assertion. + """ from getpatter.tts import elevenlabs as eleven tts = eleven.TTS() assert tts.language_code is None assert tts.voice_settings is None + # WS transport is the new default — flip recorded by ``provider_key``. + assert tts.provider_key == "elevenlabs_ws" + # The historical REST ``chunk_size`` kwarg is still accepted (no-op) so + # existing user code does not break. + tts_with_chunk = eleven.TTS(chunk_size=4096) + assert tts_with_chunk.provider_key == "elevenlabs_ws" + + +@pytest.mark.unit +def test_elevenlabs_rest_tts_preserves_chunk_size_default() -> None: + """Explicit REST opt-out keeps the historical ``chunk_size=4096`` default. + + Users on the HTTP REST transport (``ElevenLabsRestTTS``) still drive + chunking client-side, so the attribute must remain available and the + default unchanged. + """ + from getpatter import ElevenLabsRestTTS + + tts = ElevenLabsRestTTS(api_key="test-key") assert tts.chunk_size == 4096 +@pytest.mark.unit +def test_elevenlabs_facade_returns_websocket_provider() -> None: + """The facade now defaults to the WebSocket adapter, not REST.""" + from getpatter import ElevenLabsRestTTS + from getpatter.providers.elevenlabs_ws_tts import ElevenLabsWebSocketTTS + from getpatter.tts import elevenlabs as eleven + + tts = eleven.TTS() + assert isinstance(tts, ElevenLabsWebSocketTTS) + # And conversely, ``ElevenLabsRestTTS`` is not aliased to the WS class. + assert ElevenLabsRestTTS is not ElevenLabsWebSocketTTS + + @pytest.mark.unit def test_elevenlabs_facade_for_twilio_keeps_optional_kwargs_default() -> None: """The carrier factories were not touched by the language fix — still diff --git a/libraries/python/tests/unit/test_twilio_bridge_unit.py b/libraries/python/tests/unit/test_twilio_bridge_unit.py index a4b0a91d..7bd61877 100644 --- a/libraries/python/tests/unit/test_twilio_bridge_unit.py +++ b/libraries/python/tests/unit/test_twilio_bridge_unit.py @@ -8,7 +8,6 @@ import base64 import json -from collections import deque from unittest.mock import AsyncMock, MagicMock, patch import pytest @@ -306,7 +305,10 @@ async def test_oversized_message_dropped( mock_create_metrics, mock_handler_cls, ) -> None: - from getpatter.telephony.twilio import twilio_stream_bridge, _MAX_WS_MESSAGE_BYTES + from getpatter.telephony.twilio import ( + twilio_stream_bridge, + _MAX_WS_MESSAGE_BYTES, + ) # Send an oversized message then stop huge_msg = "x" * (_MAX_WS_MESSAGE_BYTES + 1) @@ -399,8 +401,12 @@ async def test_metrics_finalized_on_end( openai_key="sk-test", ) + # PCM16@16kHz: the stream handler decodes inbound mulaw to PCM16 + # before passing bytes to add_stt_audio_bytes, so metrics must be + # configured for the post-decode format (not raw mulaw 8 kHz) — + # regression guard for the 4× STT cost over-bill. mock_metrics.configure_stt_format.assert_called_once_with( - sample_rate=8000, bytes_per_sample=1 + sample_rate=16000, bytes_per_sample=2 ) mock_metrics.end_call.assert_called_once() diff --git a/libraries/typescript/package-lock.json b/libraries/typescript/package-lock.json index 10b4935e..7052d901 100644 --- a/libraries/typescript/package-lock.json +++ b/libraries/typescript/package-lock.json @@ -1,12 +1,12 @@ { "name": "getpatter", - "version": "0.6.0", + "version": "0.6.1", "lockfileVersion": 3, "requires": true, "packages": { "": { "name": "getpatter", - "version": "0.6.0", + "version": "0.6.1", "license": "MIT", "dependencies": { "express": "^5.2.1", diff --git a/libraries/typescript/package.json b/libraries/typescript/package.json index 7aa9fe32..047b8aad 100644 --- a/libraries/typescript/package.json +++ b/libraries/typescript/package.json @@ -1,6 +1,6 @@ { "name": "getpatter", - "version": "0.6.0", + "version": "0.6.1", "description": "Open-source voice AI SDK — connect any AI agent to real phone calls in 4 lines of code", "license": "MIT", "author": { diff --git a/libraries/typescript/src/_speech-events.ts b/libraries/typescript/src/_speech-events.ts index 4bbf2252..24a19d31 100644 --- a/libraries/typescript/src/_speech-events.ts +++ b/libraries/typescript/src/_speech-events.ts @@ -4,28 +4,26 @@ * * Defines `SpeechEvents`, the per-call dispatcher that fires user-facing async * callbacks and (when available) records OpenTelemetry span events on the - * current call span. The 7 events mirror the public APIs of LiveKit Agents, - * Pipecat and OpenAI Realtime so downstream metrics map onto the canonical - * Hamming AI / Coval / Cekura voice-agent metric set without translation. + * current call span. The 7 events expose the canonical voice-agent metric + * set (user/agent state transitions, turn boundaries, TTFT, first-audio) so + * downstream metrics work without translation, and align with OpenAI + * Realtime where applicable. * * This module is private (leading underscore in the file name). The public * surface is the 7 ``on*`` getters/setters plus `conversationState` exposed * on the `Patter` instance, and `SpeechEvents` is re-exported at the package * root for advanced users (custom adapters, test harnesses). * - * Industry alignment: + * Event mapping (with the OpenAI Realtime equivalent where one exists): * - * User VAD start : LiveKit user_state_changed -> speaking / - * Pipecat VADUserStartedSpeakingFrame / - * OpenAI Realtime input_audio_buffer.speech_started - * User VAD end : ..._stopped (raw VAD edge — *not* end-of-utterance) - * User EOU : LiveKit user_turn_completed / Pipecat - * UserStoppedSpeakingFrame / OpenAI Realtime - * input_audio_buffer.committed - * Agent first wire: Pipecat BotStartedSpeakingFrame - * Agent done : Pipecat BotStoppedSpeakingFrame - * LLM first token : Pipecat LLMFullResponseStartFrame (per-turn TTFT) - * TTS first audio : Pipecat OutputAudioRawFrame (first per turn) + * User VAD start : OpenAI Realtime input_audio_buffer.speech_started + * User VAD end : OpenAI Realtime input_audio_buffer.speech_stopped + * (raw VAD edge — *not* end-of-utterance) + * User EOU : OpenAI Realtime input_audio_buffer.committed + * Agent first wire: first audio chunk of the agent turn that crosses the wire + * Agent done : last audio chunk of the agent turn (or barge-in) + * LLM first token : per-turn TTFT marker + * TTS first audio : first TTS audio chunk per turn */ import { getLogger } from "./logger"; diff --git a/libraries/typescript/src/audio/transcoding.ts b/libraries/typescript/src/audio/transcoding.ts index 9db945b8..d1d839da 100644 --- a/libraries/typescript/src/audio/transcoding.ts +++ b/libraries/typescript/src/audio/transcoding.ts @@ -333,10 +333,15 @@ export class StatefulResampler { if (totalInput === 0) return Buffer.alloc(0); - // Seed FIR history on first call. + // Seed FIR history with silence on the first call. The correct + // initial condition is zeros (there was no audio before this call). + // Seeding with input[0] created a startup transient when ElevenLabs + // audio began at non-zero amplitude — the all-input[0] history + // amplified the first output sample relative to steady-state, which + // sounded as crackling on the very first TTS chunk of a call. if (!this.firHistoryValid) { - this.firHistory[0] = input[0]; - this.firHistory[1] = input[0]; + this.firHistory[0] = 0; + this.firHistory[1] = 0; this.firHistoryValid = true; } diff --git a/libraries/typescript/src/client.ts b/libraries/typescript/src/client.ts index c1494535..2138df15 100644 --- a/libraries/typescript/src/client.ts +++ b/libraries/typescript/src/client.ts @@ -39,6 +39,7 @@ import type { MetricsStore } from "./dashboard/store"; import { Carrier as TwilioCarrier } from "./telephony/twilio"; import { Carrier as TelnyxCarrier } from "./telephony/telnyx"; import { Realtime as OpenAIRealtime } from "./engines/openai"; +import { Realtime2 as OpenAIRealtime2 } from "./engines/openai-2"; import { ConvAI as ElevenLabsConvAI } from "./engines/elevenlabs"; import { CloudflareTunnel, Static as StaticTunnel } from "./tunnels"; import { resolveLogRoot } from "./services/call-log"; @@ -51,6 +52,47 @@ import type { SpeechEventCallback, } from "./_speech-events"; +/** + * Maximum concurrent entries in the prewarm-first-message cache. Bounds + * memory consumption when an outbound flood (or attacker-controlled + * ``Patter.call`` invocations) would otherwise pile up tens of MB of + * orphan TTS bytes that never evict because the carrier never fires + * ``start``. When the cap is reached, new prewarm spawns are refused + * (logged at warn, call still proceeds with live TTS). See FIX #96 in + * the parity audit. Mirrors ``_PREWARM_CACHE_MAX`` in the Python client. + */ +export const PREWARM_CACHE_MAX = 200; + +/** + * Extra grace window beyond ``ringTimeout`` after which a prewarmed + * entry that was never consumed is forcibly evicted. The TTS bill was + * paid; without TTL eviction a carrier that never fires ``start`` (e.g. + * on a never-completed dial that bypassed the status callback) would + * leak the bytes for the lifetime of the Patter instance. + */ +export const PREWARM_TTL_GRACE_MS = 5_000; + +/** + * Safety TTL (ms) after which a parked provider WebSocket whose + * carrier never fired ``start`` is force-closed. 30 s is a comfortable + * superset of typical ring + AMD windows (Twilio ~25 s, Telnyx ~25 s). + */ +const PARKED_CONN_TTL_MS = 30_000; + +/** Parked provider WebSockets ready for adoption by a per-call StreamHandler. */ +export interface ParkedProviderConnections { + /** Pre-opened STT WS (Cartesia today; other adapters may add support later). */ + stt?: import('ws').WebSocket; + /** + * Pre-opened TTS WS handle (ElevenLabs WS today). The `bosSent` flag + * lets the live `synthesizeStream` skip its own BOS send when the + * prewarm pipeline already wrote it. + */ + tts?: import('./providers/elevenlabs-ws-tts').ElevenLabsParkedWS; + /** Pre-opened OpenAI Realtime WS (already through `session.updated`). */ + openaiRealtime?: import('ws').WebSocket; +} + /** Internal local-mode state — holds carrier + resolved runtime settings. */ export interface ResolvedLocalConfig { carrier: TwilioCarrier | TelnyxCarrier; @@ -85,6 +127,13 @@ function resolvePersistRoot(persist: boolean | string | undefined): string | nul return resolveLogRoot(); } +/** Close every parked socket inside a ``ParkedProviderConnections`` slot. */ +function closeParkedConnections(slot: ParkedProviderConnections): void { + if (slot.stt) { try { slot.stt.close(); } catch { /* ignore */ } } + if (slot.tts) { try { slot.tts.ws.close(); } catch { /* ignore */ } } + if (slot.openaiRealtime) { try { slot.openaiRealtime.close(); } catch { /* ignore */ } } +} + /** Top-level SDK entry point — wraps a carrier + embedded server + agent loop. */ export class Patter { private localConfig: ResolvedLocalConfig; @@ -107,6 +156,66 @@ export class Patter { */ private tunnelOwnsWebhookUrl = false; + /** + * Pre-rendered first-message TTS audio per outbound call_id. Populated + * by :meth:`call` when ``agent.prewarmFirstMessage`` is true; consumed + * by the StreamHandler firstMessage emit so the greeting streams + * instantly on ``start`` instead of paying the 200-700 ms TTS first-byte + * latency. See ``AgentOptions.prewarmFirstMessage``. + * + * Stores raw bytes in the TTS provider's native sample rate; the + * carrier-side audio sender resamples on emit. + */ + private prewarmAudio: Map = new Map(); + /** + * Call IDs whose prewarm cache slot has already been consumed — + * either by ``popPrewarmAudio`` (cache hit OR miss on the firstMessage + * emit path) or by ``recordPrewarmWaste`` (call ended before pickup). + * The prewarm task checks this set BEFORE writing bytes so a slow + * synth that finishes after the consumer already polled doesn't + * orphan bytes in ``prewarmAudio``. See FIX #92 in the parity audit. + */ + private prewarmConsumed: Set = new Set(); + /** + * Background tasks tracked so :meth:`disconnect` can wait on / drop any + * still-running prewarm-first-message synth before tearing down. + */ + private prewarmTasks: Set> = new Set(); + /** + * TTL eviction timers keyed by call_id so :meth:`disconnect` (and + * normal consumption / waste-record paths) can cancel any pending + * timer when the slot drains naturally. Without this, the timer + * would WARN spuriously after the cache was already emptied. + */ + private prewarmTtlTimers: Map = new Map(); + /** + * Pre-opened, fully-handshaked provider WebSockets keyed by + * carrier-issued call_id. Populated by ``parkProviderConnections`` + * during the carrier ringing window; consumed by the per-call + * StreamHandler at ``start`` via ``adoptWebSocket(...)`` so STT / TTS + * / Realtime audio can flow on the first turn without paying the + * 150-900 ms TLS + WS-upgrade + protocol-handshake round-trip again. + * + * Distinct from ``prewarmAudio`` (which holds pre-rendered TTS bytes + * for the first message); the two features are complementary and + * orthogonal — both can be active for the same call. + * + * Each slot may hold up to three parked connections (STT, TTS, + * Realtime). Drained by: + * - {@link popPrewarmedConnections} on the carrier ``start`` event + * (consumed normally — the handles transfer to the StreamHandler) + * - {@link recordPrewarmWaste} on call-termination paths (no-answer, + * busy, failed, canceled, AMD voicemail). Closes parked sockets. + * - {@link disconnect} on Patter teardown. Closes all parked sockets. + */ + private prewarmedConnections: Map = new Map(); + /** + * TTL eviction handles keyed by call_id for connections that are never + * adopted (e.g. a carrier that swallows ``start``). Closes the parked + * sockets so they don't leak past the safety window. + */ + private prewarmedConnTimers: Map = new Map(); + /** * Speech-edge events for turn-taking instrumentation. Public surface: the * seven `on*` proxy accessors below plus the `conversationState` snapshot. @@ -114,14 +223,16 @@ export class Patter { * the previous behaviour. * * See `src/_speech-events.ts` for the full event taxonomy and the - * industry-alignment table (LiveKit / Pipecat / OpenAI Realtime). + * OpenAI Realtime alignment table. */ public readonly speechEvents: SpeechEvents = new SpeechEvents(); // ---- Speech-edge event callback proxies ------------------------------ - // The seven `on*` properties below mirror the public APIs of LiveKit - // Agents, Pipecat and OpenAI Realtime. They proxy to `speechEvents` so - // the dispatcher remains the single source of truth (state + OTel). + // The seven `on*` properties below follow the canonical voice-agent + // metric set (user/agent state transitions, turn boundaries, TTFT, audio + // first-byte) and align with OpenAI Realtime where applicable. They + // proxy to `speechEvents` so the dispatcher remains the single source of + // truth (state + OTel). get onUserSpeechStarted(): SpeechEventCallback | null { return this.speechEvents.onUserSpeechStarted; @@ -174,8 +285,8 @@ export class Patter { /** * Snapshot of the current per-side state of the call. - * Mirrors LiveKit's `user_state_changed` / `agent_state_changed` - * payloads. Read-only and safe to call at any time. + * Returns the user_state / agent_state payload shape — read-only and + * safe to call at any time. */ get conversationState(): ConversationStateSnapshot { return this.speechEvents.conversationState; @@ -320,7 +431,7 @@ export class Patter { ); } const engine = opts.engine; - if (engine instanceof OpenAIRealtime) { + if (engine instanceof OpenAIRealtime || engine instanceof OpenAIRealtime2) { working = { ...working, provider: 'openai_realtime', @@ -340,7 +451,7 @@ export class Patter { }; } else { throw new Error( - "Unknown engine. Expected OpenAIRealtime or ElevenLabsConvAI instance.", + "Unknown engine. Expected OpenAIRealtime, OpenAIRealtime2, or ElevenLabsConvAI instance.", ); } } else if ( @@ -437,6 +548,21 @@ export class Patter { throw new Error('agent.systemPrompt is required'); } + // Pre-import AEC at serve startup so the first call doesn't pay the + // 150-400 ms ESM dynamic-import compile / link cost on the hot path. + // ``echoCancellation`` is opt-in and rarely set on PSTN, but when it + // is the lazy ``await import('./audio/aec')`` inside StreamHandler + // serialises with first-message TTS startup and eats first-turn + // latency. Eagerly importing here costs nothing for users who never + // enable AEC (the module is pure data — no side effects). + if (opts.agent.echoCancellation) { + try { + await import('./audio/aec'); + } catch (err) { + getLogger().debug(`AEC pre-import failed at serve(): ${String(err)}`); + } + } + // Validate port if (opts.port !== undefined) { if (typeof opts.port !== 'number' || opts.port < 1 || opts.port > 65535) { @@ -551,6 +677,20 @@ export class Patter { opts.dashboard ?? true, opts.dashboardToken ?? '', ); + // Forward the prewarm-audio accessor so the per-call StreamHandler can + // consume the pre-rendered first-message audio (if any) on ``start``. + this.embeddedServer.popPrewarmAudio = this.popPrewarmAudio; + // Forward the parked-connections accessor so the per-call + // StreamHandler can adopt pre-opened STT / TTS / Realtime WSs at + // ``start`` instead of paying the cold-handshake on first turn. + this.embeddedServer.popPrewarmedConnections = this.popPrewarmedConnections; + // Forward the waste-recorder so the carrier status / hangup webhook + // handlers can evict the cache when a call terminates before the + // media stream starts (no-answer, busy, failed, canceled, or AMD + // voicemail). Without this, ``recordPrewarmWaste`` is only invoked + // from ``endCall`` and the server-side teardown path leaks the + // bytes for the lifetime of the Patter instance. See FIX #91. + this.embeddedServer.recordPrewarmWaste = this.recordPrewarmWaste; try { await this.embeddedServer.start(port); // Server is now in `listen` state on 127.0.0.1:port — safe to place @@ -594,6 +734,374 @@ export class Patter { }); } + /** + * Pop and return the pre-synthesised first-message audio for ``callId``. + * + * Returns ``undefined`` when ``agent.prewarmFirstMessage`` was not set + * for the originating outbound call, or when the synth was still in + * flight at the moment the carrier emitted ``start`` (cache miss — the + * StreamHandler falls back to live TTS). + * + * Called by the per-call StreamHandler at the start of the firstMessage + * emit. Returning bytes here lets the handler skip the live TTS + * synthesis and stream the cached buffer directly. + * + * Marks ``callId`` as consumed regardless of cache hit/miss so a slow + * synth task that finishes after this call drops its bytes instead of + * orphaning them in ``prewarmAudio``. See FIX #92. + */ + popPrewarmAudio = (callId: string): Buffer | undefined => { + this.prewarmConsumed.add(callId); + const ttl = this.prewarmTtlTimers.get(callId); + if (ttl !== undefined) { + clearTimeout(ttl); + this.prewarmTtlTimers.delete(callId); + } + const buf = this.prewarmAudio.get(callId); + if (buf !== undefined) this.prewarmAudio.delete(callId); + return buf; + }; + + /** + * Log a warning if a prewarmed greeting was paid for but never used. + * The TTS bill for ``agent.firstMessage`` has already been incurred by + * the background synth task, so the user should know — opt-in feature + * with a known cost surface. + * + * Idempotent: the second call for the same ``callId`` is a no-op, so + * the status callback firing first and ``endCall`` running afterwards + * (or vice-versa) does not double-WARN. Public so the embedded + * server's webhook handlers can invoke it on no-answer / busy / + * failed / canceled / AMD-machine paths. See FIX #91. + */ + recordPrewarmWaste = (callId: string): void => { + // Always drain any parked provider WS — they're cheap to discard + // and we don't want to leak open sockets when the call dies. + this.closePrewarmedConnections(callId); + if (this.prewarmConsumed.has(callId)) { + this.prewarmAudio.delete(callId); + return; + } + this.prewarmConsumed.add(callId); + const ttl = this.prewarmTtlTimers.get(callId); + if (ttl !== undefined) { + clearTimeout(ttl); + this.prewarmTtlTimers.delete(callId); + } + const buf = this.prewarmAudio.get(callId); + if (buf !== undefined) { + this.prewarmAudio.delete(callId); + getLogger().warn( + `Prewarm wasted for call ${callId} — first-message TTS already paid ` + + `(~${buf.byteLength} bytes synthesised) but call ended before pickup.`, + ); + } + }; + + /** + * Pop and return the parked provider WebSockets for ``callId``, or + * ``undefined`` when no parked connections exist. + * + * Wired into ``EmbeddedServer.popPrewarmedConnections`` so the + * per-call ``StreamHandler`` can adopt the parked sockets at the + * carrier ``start`` event instead of opening fresh ones — saving + * ~150-900 ms of cold-start handshake on the first turn. + */ + popPrewarmedConnections = (callId: string): ParkedProviderConnections | undefined => { + const slot = this.prewarmedConnections.get(callId); + if (slot === undefined) return undefined; + this.prewarmedConnections.delete(callId); + const ttl = this.prewarmedConnTimers.get(callId); + if (ttl !== undefined) { + clearTimeout(ttl); + this.prewarmedConnTimers.delete(callId); + } + return slot; + }; + + /** + * Close any parked provider WebSockets for ``callId``. Wired into + * ``EmbeddedServer.closePrewarmedConnections`` so call-termination + * paths (no-answer, busy, failed, canceled, AMD voicemail) drop the + * sockets cleanly instead of leaving them to the upstream timeout. + */ + closePrewarmedConnections = (callId: string): void => { + const slot = this.prewarmedConnections.get(callId); + if (slot === undefined) return; + this.prewarmedConnections.delete(callId); + const ttl = this.prewarmedConnTimers.get(callId); + if (ttl !== undefined) { + clearTimeout(ttl); + this.prewarmedConnTimers.delete(callId); + } + closeParkedConnections(slot); + }; + + /** + * Open and park provider WebSockets in parallel with the carrier-side + * ``initiateCall``. Unlike :meth:`spawnProviderWarmup` (which closes + * the WS after a brief idle), the sockets opened here stay OPEN and + * are handed off to the per-call ``StreamHandler`` on ``start``. + * + * This is the structural fix for first-turn cold-start: on Node's + * ``ws`` package, opening + closing a WS does NOT warm TLS for the + * next open — every fresh ``new WebSocket()`` re-pays the full + * TCP + TLS + HTTP-101 round-trip. By keeping the WS open and + * adopting it directly, the live first turn skips the handshake + * entirely (saves ~150-900 ms depending on provider). + * + * Best-effort: each provider's parking task is wrapped in + * ``Promise.allSettled`` so a slow or failing endpoint cannot block + * the others. Providers without ``openParkedConnection`` contribute + * nothing — the call falls through to the cold ``connect()`` path + * for that provider. + */ + private parkProviderConnections(agent: AgentOptions, callId: string): void { + const stt = agent.stt as { openParkedConnection?: () => Promise } | undefined; + const tts = agent.tts as { openParkedConnection?: () => Promise } | undefined; + const sttOpen = typeof stt?.openParkedConnection === 'function' ? stt.openParkedConnection.bind(stt) : null; + const ttsOpen = typeof tts?.openParkedConnection === 'function' ? tts.openParkedConnection.bind(tts) : null; + if (!sttOpen && !ttsOpen) return; + + const slot: ParkedProviderConnections = {}; + this.prewarmedConnections.set(callId, slot); + + const startedAt = Date.now(); + const tasks: Array> = []; + if (sttOpen) { + tasks.push((async () => { + try { + const ws = await sttOpen(); + // Slot may have been drained while we were opening (call + // failed early, ``start`` already arrived and consumer + // already adopted nothing, etc.). Close cleanly in that case. + if (this.prewarmedConnections.get(callId) !== slot) { + try { ws.close(); } catch { /* ignore */ } + return; + } + slot.stt = ws; + getLogger().info( + `[PREWARM] callId=${callId} provider=stt ms=${Date.now() - startedAt}`, + ); + } catch (err) { + getLogger().debug(`Park STT failed for ${callId}: ${String(err)}`); + } + })()); + } + if (ttsOpen) { + tasks.push((async () => { + try { + const parked = await ttsOpen(); + if (this.prewarmedConnections.get(callId) !== slot) { + try { parked.ws.close(); } catch { /* ignore */ } + return; + } + slot.tts = parked; + getLogger().info( + `[PREWARM] callId=${callId} provider=tts ms=${Date.now() - startedAt}`, + ); + } catch (err) { + getLogger().debug(`Park TTS failed for ${callId}: ${String(err)}`); + } + })()); + } + + const task = (async () => { + await Promise.allSettled(tasks); + })(); + this.prewarmTasks.add(task); + void task.finally(() => { + this.prewarmTasks.delete(task); + // Schedule TTL cleanup so a never-adopted slot is force-closed. + if (!this.prewarmedConnections.has(callId)) return; + const handle = setTimeout(() => { + this.prewarmedConnTimers.delete(callId); + const orphan = this.prewarmedConnections.get(callId); + if (orphan === undefined) return; + this.prewarmedConnections.delete(callId); + closeParkedConnections(orphan); + getLogger().warn( + `[PREWARM] parked connections evicted by TTL for ${callId} — ` + + `call never reached start (~${(PARKED_CONN_TTL_MS / 1000).toFixed(0)}s).`, + ); + }, PARKED_CONN_TTL_MS); + handle.unref?.(); + this.prewarmedConnTimers.set(callId, handle); + }); + } + + /** + * Spawn a fire-and-forget task that warms up STT / TTS / LLM in + * parallel with the carrier-side ``initiateCall``. + * + * Best-effort: each provider's optional ``warmup()`` is wrapped in + * ``Promise.allSettled`` so a slow or failing endpoint cannot block + * the others. Providers without ``warmup`` contribute nothing. + */ + private spawnProviderWarmup(agent: AgentOptions): void { + const targets: Array<{ name: string; fn: () => Promise }> = []; + const collect = (provider: unknown, label: string): void => { + if (!provider || typeof provider !== 'object') return; + const fn = (provider as { warmup?: () => Promise }).warmup; + if (typeof fn !== 'function') return; + targets.push({ + name: label, + fn: fn.bind(provider) as () => Promise, + }); + }; + collect(agent.stt, 'stt'); + collect(agent.tts, 'tts'); + collect(agent.llm, 'llm'); + if (targets.length === 0) return; + + const task = (async () => { + const results = await Promise.allSettled(targets.map((t) => t.fn())); + results.forEach((r, i) => { + if (r.status === 'rejected') { + getLogger().debug( + `Provider warmup failed (${targets[i].name}): ${String(r.reason)}`, + ); + } + }); + })(); + this.prewarmTasks.add(task); + void task.finally(() => this.prewarmTasks.delete(task)); + } + + /** + * Pre-render ``agent.firstMessage`` to TTS bytes during the ringing + * window and stash them in ``prewarmAudio.set(callId, buf)``. + * + * Skipped silently when ``agent.prewarmFirstMessage`` is false or + * when ``agent.tts`` / ``agent.firstMessage`` is missing. The synth + * is bounded by ``ringTimeout`` (default 25 s) so a never-answered + * call doesn't tie up the TTS connection. On timeout / error the + * cache is left empty and the StreamHandler falls back to live TTS. + * + * **Pipeline mode only.** Realtime / ConvAI provider modes never + * consume the prewarm cache (the StreamHandler for those modes runs + * its first-message emit through the provider's own audio path). + * Spawning the prewarm in those modes pays the TTS bill for nothing + * — refused with a warn. + * + * **Capped at ``PREWARM_CACHE_MAX`` concurrent entries.** Refused + * with a warn when the cap is reached (the call still proceeds — + * StreamHandler falls back to live TTS). + */ + private spawnPrewarmFirstMessage( + agent: AgentOptions, + callId: string, + ringTimeout: number | null | undefined, + ): void { + if (!agent.prewarmFirstMessage) return; + // FIX #94 — Realtime / ConvAI never consume the cache. Refuse early + // so the user notices the silent TTS waste instead of paying for a + // synth no caller will ever hear. + const providerMode = (agent.provider as string | undefined) ?? 'openai_realtime'; + if (providerMode !== 'pipeline') { + getLogger().warn( + `agent.prewarmFirstMessage=true is only supported in pipeline mode ` + + `(provider=${providerMode}); skipping pre-synth to avoid wasted TTS spend.`, + ); + return; + } + const firstMessage = agent.firstMessage ?? ''; + const tts = agent.tts; + if (!firstMessage || !tts) return; + if (typeof tts.synthesizeStream !== 'function') return; + + // FIX #96 — refuse to spawn when the cache (live entries + + // in-flight synth tasks) would exceed the cap. Counting both + // active entries AND pending tasks keeps the bound honest under + // outbound-flood conditions where carrier ``start`` events lag. + const inFlight = this.prewarmAudio.size + this.prewarmTasks.size; + if (inFlight >= PREWARM_CACHE_MAX) { + getLogger().warn( + `Prewarm cache full (${inFlight}/${PREWARM_CACHE_MAX} in-flight) — ` + + `skipping pre-synth for call ${callId}; falling back to live TTS at pickup.`, + ); + return; + } + + const timeoutMs = (typeof ringTimeout === 'number' ? ringTimeout : 25) * 1000; + + const task = (async () => { + try { + const accumulate = async (): Promise => { + const chunks: Buffer[] = []; + for await (const chunk of tts.synthesizeStream(firstMessage)) { + // ``synthesizeStream`` typed return is ``Buffer``, but real + // adapters may yield a ``Uint8Array`` (or anything Buffer-y). + // Guard at runtime so we never crash on a typed-but-untrue + // chunk. + const u = chunk as unknown; + if (Buffer.isBuffer(u)) chunks.push(u); + else if (ArrayBuffer.isView(u)) + chunks.push(Buffer.from((u as Uint8Array).buffer, (u as Uint8Array).byteOffset, (u as Uint8Array).byteLength)); + } + return Buffer.concat(chunks); + }; + const timer = new Promise((_resolve, reject) => + setTimeout( + () => reject(new Error('prewarm-first-message timeout')), + timeoutMs, + ).unref?.(), + ); + const buf = await Promise.race([accumulate(), timer]); + if (buf.byteLength > 0) { + // FIX #92 — race guard. If the consumer already polled (cache + // hit or miss) before the synth finished, the StreamHandler + // has already fallen back to live TTS; writing bytes here + // would orphan them in ``prewarmAudio`` until ``endCall`` ever + // runs. + if (this.prewarmConsumed.has(callId)) { + getLogger().warn( + `Prewarm orphaned for call ${callId} — synth completed ` + + `(~${buf.byteLength} bytes) AFTER consumer polled; bytes dropped, ` + + `TTS bill already paid.`, + ); + return; + } + this.prewarmAudio.set(callId, buf); + getLogger().debug( + `Prewarm first-message ready for call ${callId} (${buf.byteLength} bytes)`, + ); + } + } catch (err) { + getLogger().debug( + `Prewarm first-message failed for call ${callId}: ${String(err)}`, + ); + } + })(); + this.prewarmTasks.add(task); + void task.finally(() => { + this.prewarmTasks.delete(task); + // FIX #96 — schedule TTL eviction once the synth task has produced + // (or failed to produce) cache bytes. If the carrier never fires + // ``start`` AND the status / hangup callback never runs (e.g. + // cloud-side telephony quirk), the entry would otherwise leak. + // The timer is no-op when the slot has already been drained. + if (!this.prewarmAudio.has(callId)) return; + const ttlMs = timeoutMs + PREWARM_TTL_GRACE_MS; + const handle = setTimeout(() => { + this.prewarmTtlTimers.delete(callId); + const orphan = this.prewarmAudio.get(callId); + if (orphan === undefined) return; + this.prewarmAudio.delete(callId); + this.prewarmConsumed.add(callId); + getLogger().warn( + `Prewarm bytes evicted by TTL — call ${callId} never consumed them ` + + `(~${orphan.byteLength} bytes synthesised, ${(ttlMs / 1000).toFixed(1)}s ` + + `after ringTimeout).`, + ); + }, ttlMs); + // Don't keep the event loop alive on the eviction timer alone — + // matches the behaviour of the timeout race above. + handle.unref?.(); + this.prewarmTtlTimers.set(callId, handle); + }); + } + /** Place an outbound call via the configured carrier. */ async call(options: LocalCallOptions): Promise { if (!options.to) { @@ -624,6 +1132,15 @@ export class Patter { this.embeddedServer.onMachineDetection = options.onMachineDetection; } + // Pre-warm provider connections in parallel with the carrier-side + // ``initiateCall`` so DNS / TLS / HTTP/2 handshakes complete during + // the ringing window (3-15 s typically). Best-effort: warmup + // failures are logged at debug and never abort the call. Off when + // the user explicitly sets ``agent.prewarm: false``. + if (options.agent.prewarm !== false) { + this.spawnProviderWarmup(options.agent); + } + if (carrier.kind === 'telnyx') { // Telnyx outbound call via Call Control API. // Note: ``stream_url``/``stream_track`` are NOT accepted on @@ -661,20 +1178,29 @@ export class Patter { if (!response.ok) { throw new ProvisionError(`Failed to initiate Telnyx call: ${await response.text()}`); } - if (this.embeddedServer) { - try { - const body = (await response.clone().json()) as { data?: { call_control_id?: string } }; - const callId = body.data?.call_control_id; - if (callId) { - this.embeddedServer.metricsStore.recordCallInitiated({ - call_id: callId, - caller: phoneNumber, - callee: options.to, - direction: 'outbound', - }); - } - } catch { - /* non-fatal */ + let telnyxCallId: string | undefined; + try { + const body = (await response.clone().json()) as { data?: { call_control_id?: string } }; + telnyxCallId = body.data?.call_control_id; + } catch { + /* non-fatal */ + } + if (this.embeddedServer && telnyxCallId) { + this.embeddedServer.metricsStore.recordCallInitiated({ + call_id: telnyxCallId, + caller: phoneNumber, + callee: options.to, + direction: 'outbound', + }); + } + if (telnyxCallId) { + this.spawnPrewarmFirstMessage(options.agent, telnyxCallId, effectiveRingTimeout); + // Park provider WebSockets in parallel so the per-call + // StreamHandler can adopt them at ``start`` instead of paying + // the cold-handshake on first turn. Off when the user + // explicitly sets ``agent.prewarm: false``. + if (options.agent.prewarm !== false) { + this.parkProviderConnections(options.agent, telnyxCallId); } } return; @@ -742,31 +1268,41 @@ export class Patter { // Also log the Twilio notifications URL so users can self-diagnose // call-quality issues (warning 21626, fatal 11100, etc.) without // having to hunt them down via the Twilio Console. - if (this.embeddedServer) { - try { - const body = (await response.clone().json()) as { - sid?: string; - subresource_uris?: { notifications?: string }; - }; - const callSid = body.sid; - if (callSid) { - this.embeddedServer.metricsStore.recordCallInitiated({ - call_id: callSid, - caller: phoneNumber, - callee: options.to, - direction: 'outbound', - }); - const notificationsPath = body.subresource_uris?.notifications; - if (notificationsPath) { - getLogger().info( - `Outbound call ${callSid} placed. ` + - `Twilio notifications: https://api.twilio.com${notificationsPath} ` + - '(check here if the call drops with no audio).', - ); - } - } - } catch { - /* non-fatal — the statusCallback will register anyway */ + let twilioCallSid: string | undefined; + let twilioNotificationsPath: string | undefined; + try { + const body = (await response.clone().json()) as { + sid?: string; + subresource_uris?: { notifications?: string }; + }; + twilioCallSid = body.sid; + twilioNotificationsPath = body.subresource_uris?.notifications; + } catch { + /* non-fatal — the statusCallback will register anyway */ + } + if (this.embeddedServer && twilioCallSid) { + this.embeddedServer.metricsStore.recordCallInitiated({ + call_id: twilioCallSid, + caller: phoneNumber, + callee: options.to, + direction: 'outbound', + }); + if (twilioNotificationsPath) { + getLogger().info( + `Outbound call ${twilioCallSid} placed. ` + + `Twilio notifications: https://api.twilio.com${twilioNotificationsPath} ` + + '(check here if the call drops with no audio).', + ); + } + } + if (twilioCallSid) { + this.spawnPrewarmFirstMessage(options.agent, twilioCallSid, effectiveRingTimeout); + // Park provider WebSockets in parallel so the per-call + // StreamHandler can adopt them at ``start`` instead of paying + // the cold-handshake on first turn. Off when the user + // explicitly sets ``agent.prewarm: false``. + if (options.agent.prewarm !== false) { + this.parkProviderConnections(options.agent, twilioCallSid); } } } @@ -775,8 +1311,45 @@ export class Patter { * Stop the embedded server and any running tunnel. Safe to call multiple * times. Leaves the instance reusable: a subsequent ``serve()`` works as * if the previous lifecycle never happened. + * + * Also clears any pending TTL eviction timers, awaits in-flight + * prewarm-first-message synth tasks (best-effort, with a 1 s safety + * timeout), and clears the prewarm cache. Without this a still-running + * TTS WS keeps the user billed long after SDK teardown, and stale + * entries leak across ``serve`` / ``disconnect`` cycles. See FIX #93. */ async disconnect(): Promise { + // Clear pending TTL eviction timers and drain in-flight prewarm + // synth tasks BEFORE tearing the server down so the synth tasks + // observe a clean cancellation point and don't end up writing + // bytes to a cache we're about to drop. + for (const handle of this.prewarmTtlTimers.values()) { + clearTimeout(handle); + } + this.prewarmTtlTimers.clear(); + if (this.prewarmTasks.size > 0) { + // Promise.allSettled with a 1 s safety timeout — most synth tasks + // observe their wait_for-style timer and return promptly; a + // pathological hang must not block the disconnect path. + const drain = Promise.allSettled(Array.from(this.prewarmTasks)); + const timer = new Promise((resolve) => + setTimeout(resolve, 1_000).unref?.(), + ); + await Promise.race([drain, timer]); + } + this.prewarmTasks.clear(); + this.prewarmAudio.clear(); + this.prewarmConsumed.clear(); + // Close every parked provider WS so we don't leak sockets across + // ``serve`` / ``disconnect`` cycles (or process shutdown). + for (const handle of this.prewarmedConnTimers.values()) { + clearTimeout(handle); + } + this.prewarmedConnTimers.clear(); + for (const slot of this.prewarmedConnections.values()) { + closeParkedConnections(slot); + } + this.prewarmedConnections.clear(); if (this.tunnelHandle) { this.tunnelHandle.stop(); this.tunnelHandle = null; @@ -832,6 +1405,9 @@ export class Patter { if (!callSid) { throw new Error('callSid must be a non-empty string'); } + // If the call had a prewarmed first-message that was never consumed + // (call ended before pickup), surface the wasted spend. + this.recordPrewarmWaste(callSid); const carrier = this.localConfig.carrier; if (carrier.kind === 'twilio') { const auth = Buffer.from(`${carrier.accountSid}:${carrier.authToken}`).toString('base64'); diff --git a/libraries/typescript/src/dashboard/routes.ts b/libraries/typescript/src/dashboard/routes.ts index 3d4ab981..89e5e862 100644 --- a/libraries/typescript/src/dashboard/routes.ts +++ b/libraries/typescript/src/dashboard/routes.ts @@ -42,7 +42,12 @@ export function mountDashboard(app: Express, store: MetricsStore, token = ''): v }); app.get('/api/dashboard/calls/:callId', auth, (req, res) => { - const call = store.getCall(String(req.params.callId)); + // Fall back to the active record so the live-transcript polling path + // (``useTranscript`` in the dashboard SPA) sees turns as they accumulate + // during the call. Without this fallback the route 404s while the call + // is in flight and the live transcript pane stays empty. + const callId = String(req.params.callId); + const call = store.getCall(callId) ?? store.getActive(callId); if (!call) { res.status(404).json({ error: 'Not found' }); return; @@ -58,6 +63,37 @@ export function mountDashboard(app: Express, store: MetricsStore, token = ''): v res.json(store.getAggregates()); }); + // --- Soft delete --- + // + // ``DELETE /api/dashboard/calls/:callId`` removes a single call from the + // dashboard view + aggregates. ``POST /api/dashboard/calls/delete`` accepts + // a batch ``{ call_ids: [...] }``. Both are idempotent and never touch + // the on-disk artefacts — those serve as the durable backup. Active calls + // are silently skipped so a mid-call delete cannot orphan the live pane. + // Parity with Python. + + app.delete('/api/dashboard/calls/:callId', auth, (req, res) => { + const callId = String(req.params.callId); + const accepted = store.deleteCalls([callId]); + res.json({ deleted: accepted, count: accepted.length }); + }); + + app.post('/api/dashboard/calls/delete', auth, (req, res) => { + const body = (req.body ?? {}) as { call_ids?: unknown }; + const raw = body.call_ids; + if (!Array.isArray(raw)) { + res + .status(400) + .json({ error: "Expected JSON body { 'call_ids': [...] }" }); + return; + } + const ids = raw.filter( + (cid): cid is string => typeof cid === 'string' && cid.length > 0, + ); + const accepted = store.deleteCalls(ids); + res.json({ deleted: accepted, count: accepted.length }); + }); + // --- SSE endpoint --- app.get('/api/dashboard/events', auth, (req, res) => { @@ -147,7 +183,11 @@ export function mountApi(app: Express, store: MetricsStore, token = ''): void { }); app.get('/api/v1/calls/:callId', auth, (req, res) => { - const call = store.getCall(String(req.params.callId)); + // Same fall-through as ``/api/dashboard/calls/:callId`` — return the + // active record while the call is in flight so external integrations + // can poll a single endpoint regardless of call state. + const callId = String(req.params.callId); + const call = store.getCall(callId) ?? store.getActive(callId); if (!call) { res.status(404).json({ error: 'Call not found' }); return; diff --git a/libraries/typescript/src/dashboard/store.ts b/libraries/typescript/src/dashboard/store.ts index 8af06e74..f1be69a4 100644 --- a/libraries/typescript/src/dashboard/store.ts +++ b/libraries/typescript/src/dashboard/store.ts @@ -47,6 +47,19 @@ export class MetricsStore extends EventEmitter { private readonly maxCalls: number; private calls: CallRecord[] = []; private activeCalls: Map = new Map(); + /** + * User-driven soft delete: call_ids the operator removed from the + * dashboard view. The on-disk artefacts written by ``CallLogger`` + * (``metadata.json``, ``transcript.jsonl``) are intentionally NOT + * touched — they serve as the durable backup. All read paths + * (``getCalls`` / ``getCall`` / ``getAggregates`` / ``getCallsInRange`` + * / ``hydrate``) filter against this set so the call is invisible + * to the UI and excluded from rolling metrics. Populated from + * ``/.deleted_call_ids.json`` on hydrate so deletions + * survive a process restart. Parity with Python. + */ + private deletedCallIds: Set = new Set(); + private deletedIdsPath: string | null = null; /** * Accepts either a numeric ``maxCalls`` (legacy positional — matches the @@ -143,6 +156,18 @@ export class MetricsStore extends EventEmitter { active.status = status; Object.assign(active, extra); if (TERMINAL.has(status)) { + // Preserve the running transcript and per-turn metrics accumulated + // on the active record. Without this, a Twilio statusCallback that + // arrives before the WS ``stop`` frame (and before + // ``recordCallEnd``) would create a placeholder entry with no + // transcript and no turns — and any dashboard fetch in that race + // window would render the live-transcript pane blank. See BUG 2. + // + // TODO(0.6.2): updateCallStatus writes synthetic records with + // metrics:undefined when status callbacks arrive before + // recordCallEnd. The dashboard masks this via mergeCallPreserving + // (useDashboardData.ts) but the root cause is here — recordCallEnd + // should be the only writer to the completed buffer. const entry: CallRecord = { call_id: callId, caller: active.caller || '', @@ -152,6 +177,12 @@ export class MetricsStore extends EventEmitter { ended_at: Date.now() / 1000, status, metrics: null, + ...(active.turns && active.turns.length > 0 + ? { turns: active.turns } + : {}), + ...(active.transcript && active.transcript.length > 0 + ? { transcript: active.transcript } + : {}), ...extra, }; this.activeCalls.delete(callId); @@ -182,6 +213,42 @@ export class MetricsStore extends EventEmitter { if (active) { if (!active.turns) active.turns = []; active.turns.push(turn); + + // Mirror each completed round-trip into a flat ``transcript`` array on + // the active record so the live-transcript pane (``useTranscript`` in + // the SPA, primary mapper path) sees an accumulating ``user → assistant + // → user → assistant → …`` history without depending on the + // ``TurnMetrics`` shape. The previous implementation only populated + // ``active.turns`` (the metrics shape) and the SPA's fallback path + // re-derived a transcript from it — but any consumer that read + // ``record.transcript`` first (the canonical shape used by completed + // calls) saw an empty array, so the live pane could blank between + // round-trips. Mirroring keeps the two paths in sync. See dashboard + // BUG 1. + if (!active.transcript) active.transcript = []; + const turnRecord = turn as { + user_text?: unknown; + agent_text?: unknown; + timestamp?: unknown; + }; + const userText = + typeof turnRecord.user_text === 'string' ? turnRecord.user_text : ''; + const agentText = + typeof turnRecord.agent_text === 'string' ? turnRecord.agent_text : ''; + const ts = + typeof turnRecord.timestamp === 'number' + ? turnRecord.timestamp + : Date.now() / 1000; + if (userText.length > 0) { + active.transcript.push({ role: 'user', text: userText, timestamp: ts }); + } + if (agentText.length > 0 && agentText !== '[interrupted]') { + active.transcript.push({ + role: 'assistant', + text: agentText, + timestamp: ts, + }); + } } this.publish('turn_complete', { call_id: callId, turn: turn as Record }); @@ -195,27 +262,89 @@ export class MetricsStore extends EventEmitter { const active = this.activeCalls.get(callId); this.activeCalls.delete(callId); + // The Twilio ``statusCallback`` for ``CallStatus=completed`` arrives + // shortly before the WS ``stop`` frame and runs ``updateCallStatus``, + // which already moved the row from ``activeCalls`` into ``calls[]``. + // By the time ``recordCallEnd`` runs the active record is gone and the + // completed entry already exists. Without this lookup we'd push a + // second row with ``started_at=0`` (no active to copy from) and empty + // caller/callee — which is then ranked first by ``getCalls`` (newest + // wins) and the older, well-formed row gets shadowed. End result: the + // call disappears from the dashboard's 24 h window. See dashboard + // BUG C. + let existingIdx = -1; + if (active === undefined) { + for (let i = this.calls.length - 1; i >= 0; i--) { + if (this.calls[i].call_id === callId) { + existingIdx = i; + break; + } + } + } + const existing = existingIdx >= 0 ? this.calls[existingIdx] : undefined; + // Preserve explicit status set by a statusCallback during the call // (e.g. "no-answer" from Twilio) — fall back to "completed" when the // row was still showing the normal "in-progress" state at hang-up. - const activeStatus = active?.status; + const priorStatus = active?.status ?? existing?.status; const resolvedStatus = - activeStatus && activeStatus !== 'in-progress' ? activeStatus : 'completed'; + priorStatus && priorStatus !== 'in-progress' ? priorStatus : 'completed'; + // Resolve the final transcript and turns. ``data.transcript`` from the + // SDK is the authoritative ``history.entries`` snapshot at hang-up; when + // it's missing or empty (e.g. webhook-rejected inbound, or the active + // record was already moved to ``calls[]`` by an earlier statusCallback + // and the data payload doesn't carry one), fall back to the running + // transcript we accumulated on the active record via ``recordTurn``. + // This keeps the live-transcript pane stable across the call_status + // (``completed``) → call_end gap. See dashboard BUG 2. + const dataTranscript = data.transcript as CallRecord['transcript']; + const resolvedTranscript: CallRecord['transcript'] = + dataTranscript && dataTranscript.length > 0 + ? dataTranscript + : active?.transcript && active.transcript.length > 0 + ? active.transcript + : existing?.transcript && existing.transcript.length > 0 + ? existing.transcript + : []; + const resolvedTurns: unknown[] | undefined = + active?.turns && active.turns.length > 0 + ? active.turns + : existing?.turns && existing.turns.length > 0 + ? existing.turns + : undefined; const entry: CallRecord = { call_id: callId, - caller: (data.caller as string) || active?.caller || '', - callee: (data.callee as string) || active?.callee || '', - direction: active?.direction || (data.direction as string) || 'inbound', - started_at: active?.started_at || 0, + caller: + (data.caller as string) || + active?.caller || + existing?.caller || + '', + callee: + (data.callee as string) || + active?.callee || + existing?.callee || + '', + direction: + active?.direction || + existing?.direction || + (data.direction as string) || + 'inbound', + started_at: active?.started_at || existing?.started_at || 0, ended_at: Date.now() / 1000, - transcript: (data.transcript as CallRecord['transcript']) || [], + transcript: resolvedTranscript, + ...(resolvedTurns ? { turns: resolvedTurns } : {}), status: resolvedStatus, - metrics: metrics ?? null, + metrics: metrics ?? existing?.metrics ?? null, }; - this.calls.push(entry); - if (this.calls.length > this.maxCalls) { - this.calls = this.calls.slice(-this.maxCalls); + if (existingIdx >= 0) { + // Update in place so the buffer doesn't grow a duplicate row. + this.calls[existingIdx] = entry; + } else { + this.calls.push(entry); + if (this.calls.length > this.maxCalls) { + this.calls = this.calls.slice(-this.maxCalls); + } } this.publish('call_end', { @@ -224,20 +353,106 @@ export class MetricsStore extends EventEmitter { }); } - /** Return a window of completed calls in newest-first order. */ + /** + * Return a window of completed calls in newest-first order. + * + * Soft-deleted call_ids (see ``deleteCalls``) are filtered out so the + * dashboard never re-shows a row the user removed. The on-disk + * artefacts are intentionally preserved as a backup. + */ getCalls(limit = 50, offset = 0): CallRecord[] { - const ordered = [...this.calls].reverse(); + const visible = this.calls.filter((c) => !this.deletedCallIds.has(c.call_id)); + const ordered = visible.reverse(); return ordered.slice(offset, offset + limit); } - /** Look up a completed call by id (newest match wins). */ + /** + * Look up a completed call by id (newest match wins). + * + * Soft-deleted call_ids resolve to ``null`` so the SPA's detail pane + * cannot render a row the user removed. + */ getCall(callId: string): CallRecord | null { + if (this.deletedCallIds.has(callId)) return null; for (let i = this.calls.length - 1; i >= 0; i--) { if (this.calls[i].call_id === callId) return this.calls[i]; } return null; } + /** + * Soft-delete one or more calls from the dashboard view. + * + * Adds each ``call_id`` to an in-memory set. Subsequent reads via + * ``getCalls`` / ``getCall`` / ``getAggregates`` / ``getCallsInRange`` + * exclude the deleted ids, so rolling metrics (avg latency, total + * spend) are recomputed without them. The on-disk + * ``metadata.json`` / ``transcript.jsonl`` files written by + * ``CallLogger`` are NOT touched — they serve as a durable backup + * the operator can audit outside the dashboard. + * + * Active calls are never deletable. A call_id that is currently + * in ``activeCalls`` is silently skipped so a mid-call delete + * from the UI cannot orphan the live transcript pane. + * + * Persisted to ``/.deleted_call_ids.json`` (best-effort) + * when ``hydrate()`` has been called with a log root. Parity with + * Python ``delete_calls``. + * + * @returns The list of call_ids actually accepted as deleted. + */ + deleteCalls(callIds: readonly string[]): string[] { + const ids = new Set(); + for (const cid of callIds || []) { + if (typeof cid === 'string' && cid && !this.activeCalls.has(cid)) { + ids.add(cid); + } + } + if (ids.size === 0) return []; + const accepted: string[] = []; + for (const cid of ids) { + if (!this.deletedCallIds.has(cid)) { + this.deletedCallIds.add(cid); + accepted.push(cid); + } + } + if (accepted.length === 0) return []; + accepted.sort(); + this.persistDeletedIds(); + this.publish('calls_deleted', { call_ids: accepted }); + return accepted; + } + + /** Whether ``callId`` was soft-deleted from the dashboard. */ + isDeleted(callId: string): boolean { + return this.deletedCallIds.has(callId); + } + + /** Snapshot of soft-deleted call_ids (sorted). */ + getDeletedCallIds(): string[] { + return Array.from(this.deletedCallIds).sort(); + } + + /** Atomically persist the deleted-ids set to disk. Best-effort. */ + private persistDeletedIds(): void { + if (this.deletedIdsPath === null) return; + try { + const dir = path.dirname(this.deletedIdsPath); + fs.mkdirSync(dir, { recursive: true }); + const tmp = this.deletedIdsPath + '.tmp'; + const payload = { + version: 1, + deleted_call_ids: Array.from(this.deletedCallIds).sort(), + }; + fs.writeFileSync(tmp, JSON.stringify(payload, null, 2), 'utf8'); + fs.renameSync(tmp, this.deletedIdsPath); + } catch (err) { + getLogger().debug( + `MetricsStore.persistDeletedIds: ${String(err)}`, + ); + } + } + /** Look up an active call by id (returns undefined if not active or unknown). */ getActive(callId: string): CallRecord | undefined { return this.activeCalls.get(callId); @@ -248,9 +463,17 @@ export class MetricsStore extends EventEmitter { return Array.from(this.activeCalls.values()); } - /** Compute summary statistics across the buffered call history. */ + /** + * Compute summary statistics across the buffered call history. + * + * Soft-deleted calls are excluded so rolling metrics (avg latency, + * total spend) match exactly what the operator sees in the call list. + */ getAggregates(): Record { - const totalCalls = this.calls.length; + const visible = this.calls.filter( + (c) => !this.deletedCallIds.has(c.call_id), + ); + const totalCalls = visible.length; if (totalCalls === 0) { return { total_calls: 0, @@ -271,7 +494,7 @@ export class MetricsStore extends EventEmitter { let costLlm = 0; let costTel = 0; - for (const call of this.calls) { + for (const call of visible) { const m = call.metrics as Record | null; if (!m) continue; const cost = (m.cost as Record) || {}; @@ -282,7 +505,10 @@ export class MetricsStore extends EventEmitter { costTel += cost.telephony || 0; totalDuration += (m.duration_seconds as number) || 0; const avgLat = (m.latency_avg as Record) || {}; - const tMs = avgLat.total_ms || 0; + // Prefer the user-perceived wait time (agent_response_ms) — falls + // back to round-trip total_ms only when the SDK didn't record the + // breakdown (legacy hydrate path). + const tMs = avgLat.agent_response_ms || avgLat.total_ms || 0; if (tMs > 0) { totalLatency += tMs; latencyCount++; @@ -306,9 +532,13 @@ export class MetricsStore extends EventEmitter { }; } - /** Return calls whose `started_at` falls within `[fromTs, toTs]` (Unix seconds). */ + /** + * Return calls whose `started_at` falls within `[fromTs, toTs]` (Unix + * seconds). Soft-deleted calls are filtered out. + */ getCallsInRange(fromTs = 0, toTs = 0): CallRecord[] { return this.calls.filter((call) => { + if (this.deletedCallIds.has(call.call_id)) return false; const started = call.started_at || 0; if (fromTs && started < fromTs) return false; if (toTs && started > toTs) return false; @@ -316,9 +546,13 @@ export class MetricsStore extends EventEmitter { }); } - /** Number of completed calls currently in the ring buffer. */ + /** Number of completed (non-deleted) calls currently in the ring buffer. */ get callCount(): number { - return this.calls.length; + let n = 0; + for (const c of this.calls) { + if (!this.deletedCallIds.has(c.call_id)) n++; + } + return n; } /** @@ -333,6 +567,32 @@ export class MetricsStore extends EventEmitter { */ hydrate(logRoot: string | null | undefined): number { if (!logRoot) return 0; + + // Wire the deleted-ids persistence path FIRST so any subsequent + // ``deleteCalls`` call (even before history hydrates) lands in the + // right file. Restore the set from disk so deletions survive a + // process restart. + const deletedIdsPath = path.join(logRoot, '.deleted_call_ids.json'); + this.deletedIdsPath = deletedIdsPath; + if (fs.existsSync(deletedIdsPath)) { + try { + const raw = fs.readFileSync(deletedIdsPath, 'utf8'); + const payload = JSON.parse(raw) as { deleted_call_ids?: unknown }; + const arr = Array.isArray(payload.deleted_call_ids) + ? (payload.deleted_call_ids as unknown[]) + : []; + for (const cid of arr) { + if (typeof cid === 'string' && cid.length > 0) { + this.deletedCallIds.add(cid); + } + } + } catch (err) { + getLogger().debug( + `MetricsStore.hydrate: skipping ${deletedIdsPath}: ${String(err)}`, + ); + } + } + const callsRoot = path.join(logRoot, 'calls'); if (!fs.existsSync(callsRoot)) return 0; @@ -374,6 +634,19 @@ export class MetricsStore extends EventEmitter { ); continue; } + // CallLogger writes the transcript to a separate ``transcript.jsonl`` + // file (one turn per line) — ``metadata.json`` only carries a turn + // count. Without this fallback, hydrated past calls render with an + // empty transcript pane: the SPA polls /api/dashboard/calls/:id, + // the route serves the hydrated record verbatim, and ``transcript`` + // is ``[]``. Read the sibling file and synthesise the entry list + // the SPA expects so the pane populates on click. + if (!record.transcript || record.transcript.length === 0) { + const fromJsonl = loadTranscriptJsonl( + path.join(childPath, 'transcript.jsonl'), + ); + if (fromJsonl.length > 0) record.transcript = fromJsonl; + } collected.push(record); seen.add(callId); } catch (err) { @@ -404,6 +677,63 @@ export class MetricsStore extends EventEmitter { } } +/** + * Build a ``metrics`` object from top-level CallLogger fields. ``CallLogger`` + * writes ``cost`` / ``latency`` / ``duration_ms`` / ``telephony_provider`` at + * the top of ``metadata.json``, but the dashboard UI reads them from + * ``metrics``. Without this fallback every hydrated call shows ``$0.00`` and + * ``—`` for cost and latency. + */ +function metricsFromTopLevel( + meta: Record, +): Record | null { + const cost = + meta.cost && typeof meta.cost === 'object' + ? (meta.cost as Record) + : null; + const latency = + meta.latency && typeof meta.latency === 'object' + ? (meta.latency as Record) + : null; + const durationMs = meta.duration_ms; + const telephony = meta.telephony_provider; + if (cost === null && latency === null && durationMs == null && !telephony) { + return null; + } + const out: Record = {}; + if (cost !== null) out.cost = cost; + if (latency !== null) { + // Prefer the full LatencyBreakdown objects (avg/p50/p95/p99) when the + // server persisted them. Old metadata.json files only carry flat + // ``p50_ms/p95_ms/p99_ms`` totals — synthesize a minimal latency_avg + // from those so the table still shows a number, but no breakdown is + // available for those historical rows. + const fullAvg = latency.avg && typeof latency.avg === 'object' ? (latency.avg as Record) : null; + const fullP50 = latency.p50 && typeof latency.p50 === 'object' ? (latency.p50 as Record) : null; + const fullP95 = latency.p95 && typeof latency.p95 === 'object' ? (latency.p95 as Record) : null; + const fullP99 = latency.p99 && typeof latency.p99 === 'object' ? (latency.p99 as Record) : null; + if (fullAvg) out.latency_avg = fullAvg; + if (fullP50) out.latency_p50 = fullP50; + if (fullP95) out.latency_p95 = fullP95; + if (fullP99) out.latency_p99 = fullP99; + if (!fullAvg && !fullP50 && !fullP95) { + const totalMs = + (typeof latency.p95_ms === 'number' && latency.p95_ms) || + (typeof latency.p50_ms === 'number' && latency.p50_ms) || + 0; + out.latency_avg = { total_ms: totalMs }; + } + out.latency = latency; + } + if (typeof durationMs === 'number' && durationMs > 0) { + out.duration_seconds = durationMs / 1000; + } + if (typeof telephony === 'string' && telephony) { + out.telephony_provider = telephony; + } + return Object.keys(out).length > 0 ? out : null; +} + /** * Translate a CallLogger ``metadata.json`` payload into a ``CallRecord``. * Returns ``null`` when ``started_at`` is missing or unparseable — the record @@ -421,7 +751,7 @@ function metadataToCallRecord( const metrics = meta.metrics && typeof meta.metrics === 'object' ? (meta.metrics as Record) - : null; + : metricsFromTopLevel(meta); const transcript = Array.isArray(meta.transcript) ? (meta.transcript as CallRecord['transcript']) : []; @@ -438,6 +768,54 @@ function metadataToCallRecord( }; } +/** + * Reconstruct the dashboard ``transcript`` array from a CallLogger + * ``transcript.jsonl`` file. Each line is a turn record carrying + * ``user_text`` / ``agent_text`` / ``ts`` (ISO-8601). We expand each + * non-empty side into a separate ``{role, text, timestamp}`` entry so the + * SPA's ``toUiTranscript`` mapper can render them in order. Returns an + * empty array on any IO/parse failure — hydrate is best-effort and a + * malformed transcript file should not block the call row from showing. + */ +function loadTranscriptJsonl( + filePath: string, +): NonNullable { + try { + if (!fs.existsSync(filePath)) return []; + const raw = fs.readFileSync(filePath, 'utf8'); + const lines = raw.split('\n').filter((l) => l.trim().length > 0); + const out: NonNullable = []; + for (const line of lines) { + let row: Record; + try { + row = JSON.parse(line) as Record; + } catch { + continue; + } + const tsIso = typeof row.ts === 'string' ? Date.parse(row.ts) : NaN; + const tsNumeric = + typeof row.timestamp === 'number' ? row.timestamp * 1000 : NaN; + const timestamp = Number.isFinite(tsIso) + ? tsIso + : Number.isFinite(tsNumeric) + ? tsNumeric + : 0; + const userText = typeof row.user_text === 'string' ? row.user_text : ''; + const agentText = + typeof row.agent_text === 'string' ? row.agent_text : ''; + if (userText.length > 0) { + out.push({ role: 'user', text: userText, timestamp }); + } + if (agentText.length > 0 && agentText !== '[interrupted]') { + out.push({ role: 'assistant', text: agentText, timestamp }); + } + } + return out; + } catch { + return []; + } +} + /** * Parse a metadata timestamp into Unix seconds. 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diff --git a/libraries/typescript/src/engines/openai-2.ts b/libraries/typescript/src/engines/openai-2.ts new file mode 100644 index 00000000..e9eaf3d3 --- /dev/null +++ b/libraries/typescript/src/engines/openai-2.ts @@ -0,0 +1,70 @@ +/** + * OpenAI Realtime 2 engine — marker class for Patter client dispatch. + * + * Wraps `gpt-realtime-2` (GA Realtime API). Separate marker from + * {@link import('./openai').Realtime} because the GA endpoint speaks a + * different `session.update` wire shape; the client dispatches to + * `OpenAIRealtime2Adapter` when this marker is passed. + */ + +/** Constructor options for the OpenAI `Realtime2` engine marker. */ +export interface Realtime2Options { + /** API key. Falls back to OPENAI_API_KEY env var when omitted. */ + apiKey?: string; + /** GA Realtime model. Defaults to `gpt-realtime-2`. */ + model?: string; + /** Voice preset. Defaults to alloy. */ + voice?: string; + /** + * Reasoning-effort tier. When omitted the field is not sent and the + * server default applies. OpenAI recommends `"low"` for production + * voice flows — higher tiers add measurable per-turn latency. + */ + reasoningEffort?: 'minimal' | 'low' | 'medium' | 'high'; + /** + * Override for `audio.input.transcription.model`. Omit to keep the + * adapter default (`whisper-1`). Use `"gpt-realtime-whisper"` for + * low-latency transcript partials. + */ + inputAudioTranscriptionModel?: string; +} + +/** + * OpenAI Realtime 2 engine marker — selects `gpt-realtime-2` on the GA + * Realtime API. + * + * @example + * ```ts + * import { Patter, Twilio, OpenAIRealtime2 } from "getpatter"; + * + * const phone = new Patter({ carrier: new Twilio(), phoneNumber: "+1..." }); + * const agent = phone.agent({ + * engine: new OpenAIRealtime2({ reasoningEffort: "low" }), + * systemPrompt: "You are a friendly receptionist.", + * firstMessage: "Hello! How can I help?", + * }); + * ``` + */ +export class Realtime2 { + readonly kind = "openai_realtime_2" as const; + readonly apiKey: string; + readonly model: string; + readonly voice: string; + readonly reasoningEffort?: 'minimal' | 'low' | 'medium' | 'high'; + readonly inputAudioTranscriptionModel?: string; + + constructor(opts: Realtime2Options = {}) { + const key = opts.apiKey ?? process.env.OPENAI_API_KEY; + if (!key) { + throw new Error( + "OpenAI Realtime 2 requires an apiKey. Pass { apiKey: 'sk-...' } or " + + "set OPENAI_API_KEY in the environment.", + ); + } + this.apiKey = key; + this.model = opts.model ?? "gpt-realtime-2"; + this.voice = opts.voice ?? "alloy"; + this.reasoningEffort = opts.reasoningEffort; + this.inputAudioTranscriptionModel = opts.inputAudioTranscriptionModel; + } +} diff --git a/libraries/typescript/src/index.ts b/libraries/typescript/src/index.ts index 9eaacb1e..4069c133 100644 --- a/libraries/typescript/src/index.ts +++ b/libraries/typescript/src/index.ts @@ -67,6 +67,16 @@ export { callsToCsv, callsToJson } from "./dashboard/export"; export { mountDashboard, mountApi } from "./dashboard/routes"; export { notifyDashboard } from "./dashboard/persistence"; export { LLMLoop, OpenAILLMProvider, DefaultToolExecutor } from "./llm-loop"; +export { + MinWordsStrategy, + evaluateStrategies as evaluateBargeInStrategies, + resetStrategies as resetBargeInStrategies, +} from "./services/barge-in-strategies"; +export type { + BargeInStrategy, + EvaluateContext as BargeInEvaluateContext, + MinWordsStrategyOptions, +} from "./services/barge-in-strategies"; export type { LLMProvider, LLMChunk, @@ -122,10 +132,14 @@ export { } from "./providers/speechmatics-stt"; // New namespaced TTS classes. +// `ElevenLabsTTS` is the public facade — defaults to HTTP REST (pcm_16000). +// `ElevenLabsWebSocketTTS` is the WebSocket streaming variant. +// `ElevenLabsRestTTS` is a direct alias of the HTTP provider class. export { TTS as ElevenLabsTTS } from "./tts/elevenlabs"; export type { ElevenLabsTTSOptions } from "./tts/elevenlabs"; export { TTS as ElevenLabsWebSocketTTS } from "./tts/elevenlabs-ws"; export type { ElevenLabsWebSocketOptions } from "./tts/elevenlabs-ws"; +export { ElevenLabsTTS as ElevenLabsRestTTS } from "./providers/elevenlabs-tts"; export { TTS as OpenAITTS } from "./tts/openai"; export type { OpenAITTSOptions } from "./tts/openai"; export { TTS as CartesiaTTS } from "./tts/cartesia"; @@ -153,6 +167,19 @@ export type { GoogleLLMOptions } from "./llm/google"; export { SileroVAD } from "./providers/silero-vad"; export type { SileroVADOptions, SileroSampleRate } from "./providers/silero-vad"; +// Noise-suppression audio filters (opt-in, plug into ``agent.audioFilter``). +// DeepFilterNet — community ONNX, no license required. +export { DeepFilterNetFilter } from "./providers/deepfilternet-filter"; +export type { DeepFilterNetOptions } from "./providers/deepfilternet-filter"; +// Krisp VIVA — scaffold for parity with Python SDK. Throws at construction +// until Krisp publishes an official Node binding. See file header. +export { + KrispVivaFilter, + KrispSampleRate, + KrispFrameDuration, +} from "./providers/krisp-filter"; +export type { KrispVivaFilterOptions } from "./providers/krisp-filter"; + // Telephony carriers. export { Carrier as Twilio } from "./telephony/twilio"; export type { TwilioCarrierOptions } from "./telephony/twilio"; @@ -162,6 +189,9 @@ export type { TelnyxCarrierOptions } from "./telephony/telnyx"; // Realtime / ConvAI engines. export { Realtime as OpenAIRealtime } from "./engines/openai"; export type { RealtimeOptions as OpenAIRealtimeOptions } from "./engines/openai"; +export { Realtime2 as OpenAIRealtime2 } from "./engines/openai-2"; +export type { Realtime2Options as OpenAIRealtime2Options } from "./engines/openai-2"; +export { OpenAIRealtime2Adapter } from "./providers/openai-realtime-2"; export { ConvAI as ElevenLabsConvAI } from "./engines/elevenlabs"; export type { ConvAIOptions as ElevenLabsConvAIOptions } from "./engines/elevenlabs"; diff --git a/libraries/typescript/src/llm-loop.ts b/libraries/typescript/src/llm-loop.ts index d01ed207..8f30ad52 100644 --- a/libraries/typescript/src/llm-loop.ts +++ b/libraries/typescript/src/llm-loop.ts @@ -380,6 +380,22 @@ export interface LLMProvider { tools?: Array> | null, opts?: LLMStreamOptions, ): AsyncGenerator; + /** + * Optional best-effort pre-call DNS / TLS / HTTP-keepalive warmup. + * + * Called once per outbound call from ``Patter.call`` when the agent has + * ``prewarm: true`` (the default). Concrete providers (OpenAI, + * Anthropic, Google, Cerebras, Groq) override this to issue a + * lightweight HTTPS GET to their inference endpoint so by the time the + * first ``stream()`` call lands, the connection pool already has a + * warm socket. Failures are logged at debug level and never abort the + * call — pure latency optimisation. + * + * Optional on the interface (``warmup?: ...``) so providers without a + * warmup hook still satisfy the type. Detected via runtime + * ``typeof provider.warmup === 'function'`` in the client. + */ + warmup?(): Promise; } // --------------------------------------------------------------------------- @@ -412,6 +428,8 @@ export interface OpenAILLMSamplingOptions { /** LLM provider backed by OpenAI Chat Completions (streaming). */ export class OpenAILLMProvider implements LLMProvider { + /** Stable pricing/dashboard key — read by stream-handler/metrics. */ + static readonly providerKey = 'openai'; private readonly apiKey: string; readonly model: string; private readonly temperature?: number; @@ -440,6 +458,37 @@ export class OpenAILLMProvider implements LLMProvider { this.stop = sampling.stop; } + /** Subclasses (Cerebras, Groq) override this with their own host. */ + protected get baseUrl(): string { + return 'https://api.openai.com/v1'; + } + + /** + * Pre-call DNS / TLS / HTTP-keepalive warmup. + * + * Issues a lightweight ``GET ${baseUrl}/models`` so DNS, TLS and HTTP/2 + * are already up by the time the first ``chat.completions`` call lands. + * Best-effort: 5 s timeout, all exceptions swallowed at debug level. + * + * Note: an HTTPS GET warms DNS + TLS + connection pool but does NOT + * warm the inference path itself; for true inference warmup a real + * low-token request is needed, left as a follow-up. STT / TTS providers ship concrete + * WebSocket-based prewarms (Cartesia / Deepgram / AssemblyAI for STT; + * ElevenLabs WS for TTS) which save 200-500 ms each — those dominate + * the cold-start latency budget. + */ + async warmup(): Promise { + try { + await fetch(`${this.baseUrl}/models`, { + method: 'GET', + headers: { Authorization: `Bearer ${this.apiKey}` }, + signal: AbortSignal.timeout(5_000), + }); + } catch (err) { + getLogger().debug(`LLM warmup failed (best-effort): ${String(err)}`); + } + } + /** Stream OpenAI Chat Completions chunks for the given messages/tools. */ async *stream( messages: Array>, @@ -626,6 +675,11 @@ export class LLMLoop { // Fix 10: track provider/model so usage chunks can be attributed for billing. private readonly _providerName: string; private readonly _modelName: string; + // Diagnostics for the char/4 fallback billing path (see iterate loop). + // Counted per-LLMLoop instance (i.e. per call). Surfaced only via logs + // — keeps recordLlmUsage's public signature unchanged. Parity with Python. + private _usageMissingCount = 0; + private _loggedUsageFallback = false; // Optional async observer fired after a successful tool execution so // the host SDK (StreamHandler in pipeline mode) can surface tool calls // into the transcript timeline / `onTranscript` callback. Mirrors the @@ -772,6 +826,7 @@ export class LLMLoop { const toolCallsAccumulated = new Map(); const textParts: string[] = []; let hasToolCalls = false; + let usageChunkReceived = false; for await (const chunk of this.provider.stream(messages, this.openaiTools, opts)) { if (chunk.type === 'text' && chunk.content) { @@ -788,6 +843,7 @@ export class LLMLoop { } } else if (chunk.type === 'usage') { // Fix 10: forward token usage to the metrics accumulator for billing. + usageChunkReceived = true; metrics?.recordLlmUsage( this._providerName, this._modelName, @@ -815,6 +871,53 @@ export class LLMLoop { } } + // Fallback billing: some providers (Cerebras streaming has been + // observed to do this on certain chunk-shape variants) don't emit + // a ``usage`` chunk even with ``stream_options: { include_usage: true }``. + // Without this fallback the LLM cost silently shows ~0 for the + // whole call. char/4 is the canonical OpenAI-tokenizer rough estimate; + // conservative-upward is preferable to silent zero. Parity with Python. + if (!usageChunkReceived && metrics) { + let inputChars = 0; + for (const m of messages) { + const c = (m as { content?: unknown }).content; + if (typeof c === 'string') inputChars += c.length; + } + const outputChars = textParts.reduce((s, p) => s + p.length, 0); + const estimatedInput = Math.max(1, Math.floor(inputChars / 4)); + const estimatedOutput = Math.max(1, Math.floor(outputChars / 4)); + metrics.recordLlmUsage( + this._providerName, + this._modelName, + estimatedInput, + estimatedOutput, + 0, + 0, + ); + this._usageMissingCount += 1; + // First fallback in this call → INFO so the operator sees it once. + // Subsequent iterations only DEBUG to avoid spamming logs on long + // tool-loop turns where every iteration is char/4-billed. Parity Py. + if (!this._loggedUsageFallback) { + this._loggedUsageFallback = true; + getLogger().info( + `llm_usage_fallback provider=${this._providerName} ` + + `model=${this._modelName} input_chars=${inputChars} ` + + `output_chars=${outputChars} est_input_tokens=${estimatedInput} ` + + `est_output_tokens=${estimatedOutput}`, + ); + } else { + getLogger().debug( + `llm_usage_fallback provider=${this._providerName} ` + + `model=${this._modelName} iteration=${iter} ` + + `input_chars=${inputChars} output_chars=${outputChars} ` + + `est_input_tokens=${estimatedInput} ` + + `est_output_tokens=${estimatedOutput} ` + + `total_missing=${this._usageMissingCount}`, + ); + } + } + if (!hasToolCalls) { if (hasAfterLlmResponse && hookExecutor && hookCtx) { const finalText = allEmittedText.join(''); diff --git a/libraries/typescript/src/metrics.ts b/libraries/typescript/src/metrics.ts index 2f31cbc3..96c3dce9 100644 --- a/libraries/typescript/src/metrics.ts +++ b/libraries/typescript/src/metrics.ts @@ -26,7 +26,22 @@ import type { /** Per-turn latency breakdown across the STT/LLM/TTS pipeline. */ export interface LatencyBreakdown { + /** + * STT finalization time: end-of-speech (VAD stop or STT speech_final) → + * final transcript delivery. This is the engineering metric — pure STT + * processing latency, independent of how long the user spoke. Industry + * benchmarks (Picovoice, Deepgram, Gladia, Speechmatics) all report this + * number as "STT latency". Falls back to turn_start when the endpoint + * signal is unavailable (degraded provider, batch STT, etc.). + */ stt_ms: number; + /** + * Duration of the user's utterance (turn_start → end-of-speech). Useful + * to distinguish "user spoke for 4s" from "STT took 4s to finalize" — + * they used to be conflated in stt_ms before 0.6.1. Optional — undefined + * when the endpoint signal is unavailable. + */ + user_speech_duration_ms?: number; /** * Backwards-compatible LLM bucket. With the split below, this now reflects * the user-perceived first-token latency (TTFT) when streaming is available @@ -120,6 +135,12 @@ export interface CallMetrics { tts_provider: string; llm_provider: string; telephony_provider: string; + /** Model identifiers per provider (e.g. "ink-whisper", "eleven_flash_v2_5", + * "gpt-oss-120b"). Surface on the dashboard cost breakdown so operators + * can attribute per-call spend to a specific model. */ + stt_model?: string; + tts_model?: string; + llm_model?: string; } // ---- CallControl interface ---- @@ -237,6 +258,10 @@ export class CallMetricsAccumulator { private _actualSttCost: number | null = null; // Fix 10: accumulated LLM token cost for non-Realtime pipeline mode. private _totalLlmCost = 0; + // Last LLM model identifier from a recordLlmUsage call — emitted on + // CallMetrics.llm_model so the dashboard cost panel can display + // "Cerebras gpt-oss-120b" instead of just "Cerebras". + private _llmModel = ''; // ---- EventBus integration (item 3) ---- private _eventBus: EventBus | undefined; @@ -260,6 +285,23 @@ export class CallMetricsAccumulator { private _reportOnlyInitialTtfb: boolean; private _initialTtfbEmitted = false; + // ---- Barge-in anchor hygiene ---- + /** + * Last barge-in detection timestamp (hrTimeMs). Used by + * ``_computeTurnLatency`` to gate endpoint_ms / stt_ms emission on turns + * that started immediately after a barge-in — those turns have unreliable + * VAD/STT anchors and would otherwise pollute the p95 distribution with + * synthetic 6+ second spikes. + */ + private _lastBargeinAt: number | null = null; + /** + * Count of turns where ``recordSttComplete`` fired but no legitimate VAD + * ``speech_end`` had stamped ``_endpointSignalAt``. Exposed via metrics so + * we can spot environments where PSTN packet loss is dropping VAD stops + * (the common cause of missing endpoint signals). + */ + private _endpointSignalMissingCount = 0; + constructor(opts: { callId: string; providerMode: string; @@ -348,13 +390,51 @@ export class CallMetricsAccumulator { } } + /** + * Anchor the current turn at a legitimate VAD ``speech_start`` event. + * + * Industry-standard pattern: every VAD ``speech_start`` that fires while the agent + * is NOT in the suppressed warmup window re-anchors the turn timer to + * the wall-clock moment the user actually started speaking. Re-anchors: + * + * * ``_turnStart`` — fixes the case where a phantom ``speech_start`` + * during agent TTS or a partial transcript from the previous user + * attempt already stamped the field. Without this, the legitimate + * user-speech ``speech_start`` no-op'd and ``user_speech_duration_ms`` + * inflated from ~1 s to 5-7 s (the original "I waited 7 seconds" + * dashboard symptom). + * * ``_endpointSignalAt``, ``_vadStoppedAt``, ``_sttFinalAt`` — any + * stale anchor from a rejected barge-in / dropped final transcript + * on the same uncommitted turn is cleared, so the next + * ``recordVadStop`` / ``recordSttFinalTimestamp`` stamps fresh. + * * ``_sttComplete``, ``_llmFirstToken``, ``_initialTtfbEmitted`` — same + * rationale for the downstream pipeline timestamps. + * + * No-op once the turn is committed (``_turnCommittedMono`` set): a + * VAD ``speech_start`` after commit belongs to the NEXT turn's + * barge-in path, handled by ``recordTurnInterrupted`` instead. + */ + anchorUserSpeechStart(): void { + if (this._turnCommittedMono !== null) return; + this._turnStart = hrTimeMs(); + this._endpointSignalAt = null; + this._vadStoppedAt = null; + this._sttFinalAt = null; + this._sttComplete = null; + this._llmFirstToken = null; + this._initialTtfbEmitted = false; + } + /** Stamp end-of-STT, capture the user's transcript, and accrue billed STT seconds. */ recordSttComplete(text: string, audioSeconds = 0): void { this._sttComplete = hrTimeMs(); this._sttFinalAt = this._sttComplete; - // STT-final is the fallback endpoint signal when no VAD-stop fired earlier. + // Don't fake _endpointSignalAt from _sttComplete — that creates dishonest + // endpoint_ms == stt_ms outliers. Honest "undefined" is better than a + // 6818ms percentile spike. The counter lets us know if this happens often + // (VAD speech_end being dropped on PSTN packets is the common cause). if (this._endpointSignalAt === null) { - this._endpointSignalAt = this._sttComplete; + this._endpointSignalMissingCount++; } this._turnUserText = text; this._turnSttAudioSeconds = audioSeconds; @@ -472,7 +552,12 @@ export class CallMetricsAccumulator { * ``recordTtsStopped`` to compute ``bargein_ms``. */ recordBargeinDetected(ts?: number): void { - this._bargeinDetectedAt = ts ?? hrTimeMs(); + const t = ts ?? hrTimeMs(); + this._bargeinDetectedAt = t; + // Stamp _lastBargeinAt on the same monotonic clock as _turnStart so the + // post-barge-in anchor-gating in _computeTurnLatency stays valid (see + // the comment there for rationale). + this._lastBargeinAt = t; } /** @@ -517,7 +602,17 @@ export class CallMetricsAccumulator { timestamp: Date.now() / 1000, }; this._turns.push(turn); + // Emit the turn record BEFORE reset so subscribers see the interrupted + // turn with its anchors still intact. Parity with recordTurnComplete(). + this._eventBus?.emit('turn_ended', { callId: this.callId, turn }); + this._eventBus?.emit('metrics_collected', { callId: this.callId, turn }); this._resetTurnState(); + // Extra paranoia: explicitly null out anchors that have caused leaks + // into subsequent turns when a barge-in is in flight. _resetTurnState + // already clears them, but keep this belt-and-braces line so future + // refactors that touch _resetTurnState don't silently regress us. + this._turnCommittedMono = null; + this._endpointSignalAt = null; return turn; } @@ -706,6 +801,7 @@ export class CallMetricsAccumulator { cacheReadTokens = 0, cacheWriteTokens = 0, ): void { + this._llmModel = model; this._totalLlmCost += calculateLlmCost( provider, model, inputTokens, outputTokens, @@ -753,6 +849,9 @@ export class CallMetricsAccumulator { tts_provider: this.ttsProvider, llm_provider: this.llmProvider, telephony_provider: this.telephonyProvider, + stt_model: this.sttModel, + tts_model: this.ttsModel, + llm_model: this._llmModel, }; this._eventBus?.emit('call_ended', { callId: this.callId, metrics }); @@ -765,6 +864,16 @@ export class CallMetricsAccumulator { return this._computeCost(duration); } + /** + * Number of turns where recordSttComplete fired without a prior legitimate + * VAD speech_end. Surfaced for diagnostics — a non-zero value points at + * dropped VAD stops (commonly PSTN packet loss), which is why we stopped + * faking _endpointSignalAt from _sttComplete in 0.6.x. + */ + get endpointSignalMissingCount(): number { + return this._endpointSignalMissingCount; + } + // ---- Internal ---- private _resetTurnState(): void { @@ -781,6 +890,10 @@ export class CallMetricsAccumulator { this._bargeinStoppedAt = null; this._turnUserText = ''; this._turnSttAudioSeconds = 0; + // Reset initial-TTFB latch so EventBus TTFB emission re-fires on the new + // turn. Without this, with reportOnlyInitialTtfb=true we lose the TTFB + // metric on the first turn after a barge-in / new turn. + this._initialTtfbEmitted = false; } private _computeTurnLatency(): LatencyBreakdown { @@ -793,22 +906,35 @@ export class CallMetricsAccumulator { let endpoint_ms: number | undefined; let bargein_ms: number | undefined; let tts_total_ms: number | undefined; - - // ``stt_ms`` is the wall-clock window from the first audio byte with - // detected speech to the final transcript. It includes the user's speech - // duration AND the provider's endpointing wait — both contribute to the - // time the agent is blocked waiting on STT, so this is what matters for - // UX. To isolate provider-only processing latency you'd need an external - // VAD signalling end-of-speech *before* the STT provider's own decision, - // which streaming providers like Deepgram do not expose separately - // (they emit speech_final and is_final in the same chunk). - if (this._turnStart !== null && this._sttComplete !== null) { - stt_ms = this._sttComplete - this._turnStart; + let user_speech_duration_ms: number | undefined; + + // Post-barge-in turns have unreliable anchors. Drop endpoint_ms / stt_ms + // to avoid polluting the p95 distribution with synthetic spikes. The + // honest "undefined" makes the metric usable for SLO/alerting; without + // this gate, a single barge-in produces 6+ second p95 outliers. + const postBargein = + this._lastBargeinAt !== null && + this._turnStart !== null && + Math.abs(this._turnStart - this._lastBargeinAt) <= 100; + + // ``stt_ms`` measures pure STT finalization: end-of-speech (VAD stop or + // STT speech_final) → final transcript delivery. This is the + // engineering metric reported as "STT latency" by the industry. When + // the endpoint signal is unavailable (degraded provider, batch STT) + // fall back to the legacy turn_start anchor so the field is never + // spuriously zero. + if (this._sttComplete !== null) { + const anchor = this._endpointSignalAt ?? this._turnStart; + if (anchor !== null) { + stt_ms = Math.max(0, this._sttComplete - anchor); + } + } + if (this._turnStart !== null && this._endpointSignalAt !== null) { + user_speech_duration_ms = Math.max( + 0, + this._endpointSignalAt - this._turnStart, + ); } - // Note: an ``stt_endpointing_ms`` (post-speech wait) metric would be - // useful but Deepgram emits speech_final and final-transcript in the same - // chunk, so the gap collapses to ~0. To get a meaningful value we'd need - // an external VAD (Silero) signalling end-of-speech earlier. Deferred. // ``llm_ms`` is the user-facing latency that maps to UX: time-to-first-token // from end-of-STT. ``llm_total_ms`` captures the full generation duration // (stt_complete → llm_complete) so it can be tracked separately for @@ -870,6 +996,15 @@ export class CallMetricsAccumulator { agent_response_ms = round(endpoint_ms + llm_ttft_ms + tts_ms, 1); } + // Post-barge-in anchor hygiene: when the current turn began within 100 ms + // of the last detected barge-in, the VAD/STT anchors are unreliable. Drop + // the polluted endpoint_ms and stt_ms so percentile aggregations ignore + // them (stt_ms = 0 is excluded by nonZero() in _computePercentileLatency). + if (postBargein) { + stt_ms = 0; + endpoint_ms = undefined; + } + // Note: in Realtime mode OpenAI handles STT+LLM+TTS as a single opaque // pipeline, so stt_ms / llm_ms / tts_ms stay 0 and only total_ms is // meaningful. Dashboards should prefer total_ms as the end-to-end proxy @@ -878,6 +1013,9 @@ export class CallMetricsAccumulator { return { stt_ms: round(stt_ms, 1), llm_ms: round(llm_ms, 1), + ...(user_speech_duration_ms !== undefined + ? { user_speech_duration_ms: round(user_speech_duration_ms, 1) } + : {}), ...(llm_ttft_ms !== undefined ? { llm_ttft_ms: round(llm_ttft_ms, 1) } : {}), ...(llm_total_ms !== undefined ? { llm_total_ms: round(llm_total_ms, 1) } : {}), tts_ms: round(tts_ms, 1), @@ -978,6 +1116,8 @@ export class CallMetricsAccumulator { const endpointAvg = optAvg('endpoint_ms'); const bargeinAvg = optAvg('bargein_ms'); const ttsTotalAvg = optAvg('tts_total_ms'); + const userSpeechAvg = optAvg('user_speech_duration_ms'); + const agentResponseAvg = optAvg('agent_response_ms'); return { stt_ms: round(turns.reduce((s, t) => s + t.latency.stt_ms, 0) / n, 1), llm_ms: round(turns.reduce((s, t) => s + t.latency.llm_ms, 0) / n, 1), @@ -988,6 +1128,8 @@ export class CallMetricsAccumulator { ...(endpointAvg !== undefined ? { endpoint_ms: endpointAvg } : {}), ...(bargeinAvg !== undefined ? { bargein_ms: bargeinAvg } : {}), ...(ttsTotalAvg !== undefined ? { tts_total_ms: ttsTotalAvg } : {}), + ...(userSpeechAvg !== undefined ? { user_speech_duration_ms: userSpeechAvg } : {}), + ...(agentResponseAvg !== undefined ? { agent_response_ms: agentResponseAvg } : {}), }; } @@ -1016,6 +1158,8 @@ export class CallMetricsAccumulator { const endpointP = optPct('endpoint_ms'); const bargeinP = optPct('bargein_ms'); const ttsTotalP = optPct('tts_total_ms'); + const userSpeechP = optPct('user_speech_duration_ms'); + const agentResponseP = optPct('agent_response_ms'); return { stt_ms: round(percentile(nonZero(turns.map((t) => t.latency.stt_ms)), p), 1), llm_ms: round(percentile(nonZero(turns.map((t) => t.latency.llm_ms)), p), 1), @@ -1026,6 +1170,8 @@ export class CallMetricsAccumulator { ...(endpointP !== undefined ? { endpoint_ms: endpointP } : {}), ...(bargeinP !== undefined ? { bargein_ms: bargeinP } : {}), ...(ttsTotalP !== undefined ? { tts_total_ms: ttsTotalP } : {}), + ...(userSpeechP !== undefined ? { user_speech_duration_ms: userSpeechP } : {}), + ...(agentResponseP !== undefined ? { agent_response_ms: agentResponseP } : {}), }; } } diff --git a/libraries/typescript/src/observability/attributes.ts b/libraries/typescript/src/observability/attributes.ts new file mode 100644 index 00000000..c46862ab --- /dev/null +++ b/libraries/typescript/src/observability/attributes.ts @@ -0,0 +1,309 @@ +/** + * ``patter.*`` span attribute helpers — TypeScript mirror of + * ``libraries/python/getpatter/observability/attributes.py``. + * + * No-op when OpenTelemetry isn't wired up. The helpers exist primarily so + * provider adapters in both SDKs can share the same call sites without + * each one re-implementing the "is OTel available?" gating. When tracing + * is disabled (``PATTER_OTEL_ENABLED`` unset, ``@opentelemetry/api`` not + * installed, or no active call scope), every helper is a fast no-op. + * + * Parity contract with the Python helpers: + * - ``record_patter_attrs`` ↔ ``recordPatterAttrs`` + * - ``patter_call_scope`` ↔ ``patterCallScope`` + * - ``attach_span_exporter`` ↔ ``attachSpanExporter`` + * + * The semantics differ in one structural way: Python uses ``ContextVar`` + * to propagate the active call ID through asyncio task trees. JS has no + * equivalent built-in, so this module uses a module-level "current call" + * cell and a stack to support nested ``patterCallScope`` invocations on + * the same loop. That's fine because the TS SDK has at most one active + * call scope per Node process / per request handler in practice. + */ +import { getLogger } from '../logger'; +import { ENV_FLAG, isTracingEnabled } from './tracing'; + +/** + * "uut" = unit-under-test, the default value stamped on + * ``patter.side`` when no driver/UUT split is configured. + */ +export const DEFAULT_SIDE = 'uut'; + +interface CallScopeFrame { + readonly callId: string; + readonly side: string; +} + +const _scopeStack: CallScopeFrame[] = []; + +function _currentScope(): CallScopeFrame | null { + return _scopeStack.length > 0 ? _scopeStack[_scopeStack.length - 1] : null; +} + +interface OtelApiShape { + trace: { + getTracer(name: string): { + startSpan( + name: string, + options?: { attributes?: Record }, + ): { + setAttribute(key: string, value: unknown): unknown; + end(): unknown; + isRecording?(): boolean; + }; + }; + getActiveSpan?(): { + setAttribute(key: string, value: unknown): unknown; + isRecording?(): boolean; + } | null; + }; +} + +function _tryLoadOtelApi(): OtelApiShape | null { + try { + // eslint-disable-next-line @typescript-eslint/no-require-imports + return require('@opentelemetry/api') as OtelApiShape; + } catch { + return null; + } +} + +/** + * Stamp ``patter.*`` attributes on the current span, augmenting them with + * the ambient ``patter.call_id`` / ``patter.side`` from the active + * ``patterCallScope``. No-op when tracing is disabled or no scope is + * active. + * + * Behaviour mirrors the Python helper: + * - If an active recording span exists, attributes are stamped on it. + * - Otherwise a transient zero-duration ``patter.billable`` span is + * opened to carry the attributes. Some collectors filter + * zero-duration spans; callers that need guaranteed attribution + * should wrap their billable work in their own span. + * + * Caller-provided ``patter.call_id`` / ``patter.side`` keys win over the + * scope's values. + */ +export function recordPatterAttrs(attrs: Readonly>): void { + if (!isTracingEnabled()) return; + const scope = _currentScope(); + if (scope === null) return; + + const api = _tryLoadOtelApi(); + if (!api) return; + + const full: Record = { ...attrs }; + if (full['patter.call_id'] === undefined) full['patter.call_id'] = scope.callId; + if (full['patter.side'] === undefined) full['patter.side'] = scope.side; + + try { + const active = api.trace.getActiveSpan?.() ?? null; + if (active && (active.isRecording === undefined || active.isRecording())) { + for (const [k, v] of Object.entries(full)) { + try { + active.setAttribute(k, v); + } catch { + // Swallow — OTel must never crash the call path. + } + } + return; + } + } catch { + // fall through to billable-span fallback + } + + try { + const tracer = api.trace.getTracer('getpatter.observability'); + const span = tracer.startSpan('patter.billable', { attributes: full }); + try { + span.end(); + } catch { + // Swallow. + } + } catch { + // Swallow. + } +} + +/** + * Bind ``callId`` and ``side`` to the active span scope for the duration + * of ``fn``. Mirrors the Python ``patter_call_scope`` context manager: + * any ``recordPatterAttrs`` call made inside ``fn`` (or anything ``fn`` + * awaits) sees the bound values. + * + * Note: JavaScript has no ContextVar equivalent, so this uses a + * module-level stack. Concurrent overlapping scopes on the same event + * loop will see the innermost scope's values — fine for the SDK's + * one-call-per-handler model. If callers need true async-context + * isolation, install ``AsyncLocalStorage``-backed propagation via the + * OTel SDK's context manager. + */ +export async function patterCallScope( + options: { readonly callId: string; readonly side?: string }, + fn: () => Promise, +): Promise { + if (!options.callId) { + throw new Error('patterCallScope requires non-empty callId'); + } + const frame: CallScopeFrame = { + callId: options.callId, + side: options.side ?? DEFAULT_SIDE, + }; + _scopeStack.push(frame); + try { + return await fn(); + } finally { + // Defensive pop: locate this exact frame in case nested scopes + // raised and left the stack in an unexpected state. + const idx = _scopeStack.lastIndexOf(frame); + if (idx >= 0) _scopeStack.splice(idx, 1); + } +} + +/** + * Wire an OTel ``SpanExporter`` into the SDK's tracer provider and + * remember the configured ``side`` on the Patter instance so the + * per-call handler reads it when entering ``patterCallScope``. + * + * Mirrors the Python ``attach_span_exporter`` contract: + * - Stores ``side`` on ``patterInstance._patterSide`` unconditionally + * (works even when ``@opentelemetry/*`` peer deps are missing). + * - Idempotent on the *same exporter object reference*. Two distinct + * exporter instances pointing at the same backend will both be + * attached and spans will be exported twice — hold a single + * exporter object across calls to avoid duplicates. + * + * When tracing isn't enabled (env flag off / SDK peer deps absent), the + * call is a no-op aside from storing ``_patterSide``. + */ +export function attachSpanExporter( + patterInstance: { _patterSide?: string } & Record, + exporter: unknown, + options: { readonly side?: string } = {}, +): void { + const side = options.side ?? DEFAULT_SIDE; + patterInstance._patterSide = side; + + if (!isTracingEnabled()) { + getLogger().debug( + `attachSpanExporter: ${ENV_FLAG} not enabled or tracer unavailable; only side= stored`, + ); + return; + } + + // SDK wire-up packages are optional. When absent, fall through silently + // — Python does the same. + let sdkTraceBase: + | { + BasicTracerProvider: new (opts?: unknown) => { + addSpanProcessor?(p: unknown): void; + _patterAttachedExporters?: Set; + }; + SimpleSpanProcessor: new (exporter: unknown) => unknown; + } + | null = null; + let sdkTraceNode: + | { + NodeTracerProvider: new (opts?: unknown) => { + addSpanProcessor?(p: unknown): void; + _patterAttachedExporters?: Set; + }; + } + | null = null; + try { + // eslint-disable-next-line @typescript-eslint/no-require-imports + sdkTraceBase = require('@opentelemetry/sdk-trace-base'); + } catch { + sdkTraceBase = null; + } + try { + // eslint-disable-next-line @typescript-eslint/no-require-imports + sdkTraceNode = require('@opentelemetry/sdk-trace-node'); + } catch { + sdkTraceNode = null; + } + if (!sdkTraceBase) { + getLogger().warn( + 'attachSpanExporter: @opentelemetry/sdk-trace-base is not installed; ' + + 'spans will not be exported. Install ' + + '@opentelemetry/sdk-trace-base + @opentelemetry/sdk-trace-node.', + ); + return; + } + + const api = _tryLoadOtelApi(); + if (!api) return; + + let provider: + | ({ + addSpanProcessor?(p: unknown): void; + _patterAttachedExporters?: Set; + } & Record) + | null = null; + + try { + // Prefer the existing global provider — never replace a host-app + // TracerProvider silently (parity with Python's behaviour). + const tracerApi = api.trace as unknown as { + getTracerProvider?(): unknown; + }; + const existing = tracerApi.getTracerProvider?.() ?? null; + if ( + existing && + typeof (existing as { addSpanProcessor?: unknown }).addSpanProcessor === 'function' + ) { + provider = existing as unknown as typeof provider; + } + } catch { + provider = null; + } + + if (!provider) { + if (!sdkTraceNode) { + getLogger().warn( + 'attachSpanExporter: no SDK TracerProvider registered and ' + + '@opentelemetry/sdk-trace-node is not installed; cannot wire exporter.', + ); + return; + } + try { + provider = new sdkTraceNode.NodeTracerProvider(); + const trace = api.trace as unknown as { + setGlobalTracerProvider?(p: unknown): void; + }; + trace.setGlobalTracerProvider?.(provider); + } catch (e) { + getLogger().debug( + `attachSpanExporter: failed to construct NodeTracerProvider: ${String( + (e as Error)?.message ?? e, + )}`, + ); + return; + } + } + + // Idempotency: track attached exporters by reference on the provider. + let seen = provider._patterAttachedExporters; + if (!seen) { + seen = new Set(); + provider._patterAttachedExporters = seen; + } + if (seen.has(exporter)) return; + + try { + const processor = new sdkTraceBase.SimpleSpanProcessor(exporter); + provider.addSpanProcessor?.(processor); + seen.add(exporter); + } catch (e) { + getLogger().debug( + `attachSpanExporter: failed to register exporter: ${String( + (e as Error)?.message ?? e, + )}`, + ); + } +} + +/** Internal: reset module state (primarily for tests; not part of the public API). */ +export function _resetPatterAttrsForTesting(): void { + _scopeStack.length = 0; +} diff --git a/libraries/typescript/src/observability/index.ts b/libraries/typescript/src/observability/index.ts index 3a2ae5db..ae94bcea 100644 --- a/libraries/typescript/src/observability/index.ts +++ b/libraries/typescript/src/observability/index.ts @@ -22,6 +22,14 @@ export { } from './tracing'; export type { Span, InitTracingOptions } from './tracing'; +// ---- patter.* span attribute helpers (parity with Python) ---- +export { + recordPatterAttrs, + patterCallScope, + attachSpanExporter, + DEFAULT_SIDE, +} from './attributes'; + /** * Call lifecycle event — TS mirror of ``getpatter.models.CallEvent``. * diff --git a/libraries/typescript/src/pricing.ts b/libraries/typescript/src/pricing.ts index d8e5d762..f194d191 100644 --- a/libraries/typescript/src/pricing.ts +++ b/libraries/typescript/src/pricing.ts @@ -107,14 +107,26 @@ export const DEFAULT_PRICING: Record = { // STT — per minute of audio processed. deepgram: { unit: PricingUnit.MINUTE, - // Default = Nova-3 streaming monolingual ($0.0077/min). Previous $0.0043 - // was the batch rate; streaming is ~80% more expensive. - price: 0.0077, + // Default = Nova-3 streaming monolingual ($0.0048/min, current Pay- + // As-You-Go promotional rate). Source: https://deepgram.com/pricing + // (verified 2026-05-11). The promo replaces the standard $0.0077/min + // quoted at Nova-3 launch and is the rate customers actually pay + // today; revisit when Deepgram removes the "Limited-time promotional + // rates on streaming" banner. + price: 0.0048, models: { - 'nova-3': { price: 0.0077 }, - 'nova-3-multilingual': { price: 0.0092 }, + // Nova-3 family — current flagship. + 'nova-3': { price: 0.0048 }, + 'nova-3-multilingual': { price: 0.0058 }, + // Flux family — new event-driven turn-taking STT (2026 launch). + flux: { price: 0.0065 }, + 'flux-english': { price: 0.0065 }, + 'flux-multilingual': { price: 0.0078 }, + // Legacy Nova-2 / Nova-1 — still supported but no longer featured on + // the public pricing page; rates kept as last verified. 'nova-2': { price: 0.0058 }, nova: { price: 0.0043 }, + // Whisper Cloud via Deepgram — separate tier. 'whisper-large': { price: 0.0048 }, 'whisper-medium': { price: 0.0048 }, }, @@ -153,27 +165,30 @@ export const DEFAULT_PRICING: Record = { // retired; users were being over-billed ~4.3x. speechmatics: { unit: PricingUnit.MINUTE, price: 0.004 }, // TTS — per 1,000 characters synthesized. + // Source: https://elevenlabs.io/pricing/api (verified 2026-05-11). The + // per-1K-character API/overage rate is flat across all plan tiers (Free + // through Business); only the included character bundle varies by plan. elevenlabs: { unit: PricingUnit.THOUSAND_CHARS, - // Default = eleven_flash_v2_5 (Patter's default model) at $0.06/1k. - price: 0.06, + // Default = eleven_flash_v2_5 (Patter's default model) at $0.05/1k. + price: 0.05, models: { - eleven_flash_v2_5: { price: 0.06 }, + eleven_flash_v2_5: { price: 0.05 }, eleven_turbo_v2_5: { price: 0.05 }, - eleven_multilingual_v2: { price: 0.18 }, - eleven_monolingual_v1: { price: 0.18 }, - eleven_v3: { price: 0.30 }, + eleven_multilingual_v2: { price: 0.10 }, + eleven_monolingual_v1: { price: 0.10 }, + eleven_v3: { price: 0.10 }, }, }, // ElevenLabs WebSocket streaming TTS shares pricing with REST. elevenlabs_ws: { unit: PricingUnit.THOUSAND_CHARS, - price: 0.06, + price: 0.05, models: { - eleven_flash_v2_5: { price: 0.06 }, + eleven_flash_v2_5: { price: 0.05 }, eleven_turbo_v2_5: { price: 0.05 }, - eleven_multilingual_v2: { price: 0.18 }, - eleven_v3: { price: 0.30 }, + eleven_multilingual_v2: { price: 0.10 }, + eleven_v3: { price: 0.10 }, }, }, openai_tts: { @@ -303,7 +318,24 @@ export const DEFAULT_PRICING: Record = { // calls on a local number). For US toll-free inbound ($0.022/min) or US // outbound local ($0.0140/min), override via Patter({ pricing: { twilio: {...} } }). twilio: { unit: PricingUnit.MINUTE, price: 0.0085 }, + // Telnyx — direction-aware rates as of 2026-05-11. + // Sources: + // https://telnyx.com/pricing/elastic-sip + // https://telnyx.com/pricing/voice-api + // US inbound (DID / local termination, Pay-As-You-Go): $0.0035/min + // US outbound (Pay-As-You-Go, mid-range of $0.005-$0.009): $0.007/min + // Billing granularity is per-MINUTE (Telnyx rounds partial minutes up + // on the invoice; prior internal docs incorrectly claimed per-second). + // The legacy ``telnyx`` key is preserved at the outbound rate as a + // safe fallback for users who override ``pricing: { telnyx: {...} }`` + // without knowing the direction; the metrics layer currently uses + // this flat key (direction is not threaded through to + // ``calculateTelephonyCost``). Direction-aware billing can be enabled + // by override-only: ``new Patter({ pricing: { telnyx: { unit: 'minute', + // price: 0.0035 } } })`` to bill all inbound at the lower rate. telnyx: { unit: PricingUnit.MINUTE, price: 0.007 }, + telnyx_inbound: { unit: PricingUnit.MINUTE, price: 0.0035 }, + telnyx_outbound: { unit: PricingUnit.MINUTE, price: 0.007 }, }; function cloneProviderEntry(entry: ProviderPricing): ProviderPricing { @@ -566,16 +598,18 @@ export const llmPricing: Record> = { 'gemma2-9b-it': { input: 0.20, output: 0.20 }, }, cerebras: { - // Rates as of 2026-05-08; verify against cerebras.net/inference. - // ``gpt-oss-120b`` is the Patter default for Cerebras (set in 0.5.4). - // On WSE-3 hardware every model size saturates the downstream TTS - // consumption rate (~150-300 tok/sec), so the 120B price stays in line - // with the 70B tier rather than scaling with weight count. - 'gpt-oss-120b': { input: 0.85, output: 1.20 }, - 'llama3.1-8b': { input: 0.10, output: 0.20 }, + // Rates as of 2026-05-11 verified against the canonical per-model docs + // pages at ``https://inference-docs.cerebras.ai/models/``. The + // previous 2026-05-08 update overcharged across the board (gpt-oss-120b + // 2.4x input, qwen-3-235b 1.67x input) because it conflated the launch + // blog quotes with the "Exploration pricing" banner now shown on each + // model page. Parity with libraries/python/getpatter/pricing.py. + 'gpt-oss-120b': { input: 0.35, output: 0.75 }, + 'llama3.1-8b': { input: 0.10, output: 0.10 }, 'llama-3.3-70b': { input: 0.85, output: 1.20 }, 'qwen-3-32b': { input: 0.40, output: 0.80 }, - 'qwen-3-235b-a22b-instruct-2507': { input: 1.00, output: 1.50 }, + 'qwen-3-235b-a22b-instruct-2507': { input: 0.60, output: 1.20 }, + 'qwen-3-coder-480b': { input: 2.00, output: 2.00 }, 'zai-glm-4.7': { input: 0.85, output: 1.20 }, }, // OpenAI Chat Completions (non-Realtime) — mirrors the Python SDK pricing table. diff --git a/libraries/typescript/src/provider-factory.ts b/libraries/typescript/src/provider-factory.ts index a3221de3..c5f0cc06 100644 --- a/libraries/typescript/src/provider-factory.ts +++ b/libraries/typescript/src/provider-factory.ts @@ -61,11 +61,23 @@ export interface STTAdapter { * entirely; the stream handler does an optional-chained call. */ finalize?(): void | Promise; + /** + * Optional best-effort pre-call DNS / TLS / HTTP-keepalive warmup. + * Default behaviour is a no-op — providers that benefit (e.g. + * provider WebSockets with a slow handshake) can override. Failures + * must never abort the call. + */ + warmup?(): Promise; } /** Shape shared by every TTS adapter in the SDK. */ export interface TTSAdapter { synthesizeStream(text: string): AsyncIterable; + /** + * Optional best-effort pre-call DNS / TLS / HTTP-keepalive warmup. + * Default behaviour is a no-op. Failures must never abort the call. + */ + warmup?(): Promise; } /** diff --git a/libraries/typescript/src/providers/anthropic-llm.ts b/libraries/typescript/src/providers/anthropic-llm.ts index b051402f..c062ac0f 100644 --- a/libraries/typescript/src/providers/anthropic-llm.ts +++ b/libraries/typescript/src/providers/anthropic-llm.ts @@ -105,6 +105,8 @@ interface AnthropicSystemBlock { /** LLM provider backed by Anthropic's Messages API (streaming). */ export class AnthropicLLMProvider implements LLMProvider { + /** Stable pricing/dashboard key — read by stream-handler/metrics. */ + static readonly providerKey = 'anthropic'; private readonly apiKey: string; private readonly model: string; private readonly maxTokens: number; @@ -129,6 +131,29 @@ export class AnthropicLLMProvider implements LLMProvider { this.promptCaching = options.promptCaching ?? true; } + /** + * Pre-call DNS / TLS warmup for the Anthropic Messages API. + * Issues a lightweight ``GET https://api.anthropic.com/v1/models`` so + * DNS, TLS and HTTP/2 are already up by the time the first ``messages`` + * call lands. Best-effort: 5 s timeout, exceptions swallowed at debug. + */ + async warmup(): Promise { + try { + // ``url`` points at .../messages — derive the .../models sibling. + const modelsUrl = this.url.replace(/\/messages\/?$/, '/models'); + await fetch(modelsUrl, { + method: 'GET', + headers: { + 'x-api-key': this.apiKey, + 'anthropic-version': this.anthropicVersion, + }, + signal: AbortSignal.timeout(5_000), + }); + } catch (err) { + getLogger().debug(`Anthropic LLM warmup failed (best-effort): ${String(err)}`); + } + } + /** Stream Patter-format LLM chunks for the given OpenAI-style chat history. */ async *stream( messages: Array>, diff --git a/libraries/typescript/src/providers/assemblyai-stt.ts b/libraries/typescript/src/providers/assemblyai-stt.ts index 7cd377ca..4b22483d 100644 --- a/libraries/typescript/src/providers/assemblyai-stt.ts +++ b/libraries/typescript/src/providers/assemblyai-stt.ts @@ -143,6 +143,8 @@ export class AssemblyAISTTNotConnectedError extends Error { /** Streaming STT adapter for AssemblyAI's v3 Universal Streaming API. */ export class AssemblyAISTT { + /** Stable pricing/dashboard key — read by stream-handler/metrics. */ + static readonly providerKey = 'assemblyai'; private ws: WebSocket | null = null; private readonly callbacks: Set = new Set(); private closing = false; @@ -271,6 +273,72 @@ export class AssemblyAISTT { return headers; } + /** + * Pre-call WebSocket warmup for the AssemblyAI v3 `/v3/ws` endpoint. + * + * Opens the WS (DNS + TLS + auth handshake), idles ~250 ms so the + * AssemblyAI edge keeps the session state warm, then sends Terminate + * and closes. By the time `connect()` is invoked at call-pickup the + * resolver and TLS session are hot — net wire time saving of + * 200-500 ms. + * + * Billing safety: AssemblyAI Universal Streaming bills on streamed + * audio seconds (per https://www.assemblyai.com/pricing). Opening + + * closing the WebSocket without forwarding any audio frames does + * not consume billable seconds. Best-effort: failures logged at + * debug level. + */ + async warmup(): Promise { + const url = this.buildUrl(); + const headers = this.buildHeaders(); + let ws: WebSocket | null = null; + try { + ws = await new Promise((resolve, reject) => { + const sock = new WebSocket(url, { headers }); + const timer = setTimeout(() => { + try { + sock.close(); + } catch { + // ignore + } + reject(new Error('AssemblyAI STT warmup connect timeout')); + }, 5000); + sock.once('open', () => { + clearTimeout(timer); + resolve(sock); + }); + sock.once('error', (err: Error) => { + clearTimeout(timer); + reject(err); + }); + }); + // Idle briefly so the provider edge keeps session state warm. + await new Promise((r) => setTimeout(r, 250)); + try { + ws.send(JSON.stringify({ type: AssemblyAIClientFrame.TERMINATE })); + } catch { + // ignore + } + } catch (err) { + // IMPORTANT: ``String(err)`` for a `ws` handshake failure can + // include the request URL, which carries the API key as a + // ``?token=...`` query parameter when ``useQueryToken`` is set. + // Log only the HTTP status (when present) or the error class + // name — never the full URL or message. + getLogger().debug( + `AssemblyAI STT warmup failed (best-effort): ${describeWarmupError(err)}`, + ); + } finally { + if (ws) { + try { + ws.close(); + } catch { + // ignore + } + } + } + } + /** Open the streaming WebSocket and arm message handlers. */ async connect(): Promise { this.closing = false; @@ -562,3 +630,24 @@ function averageConfidence(words: readonly AssemblyAIWord[]): number { } return total / words.length; } + +/** + * Render a warmup error for logging without leaking the request URL. + * + * `String(err)` on a `ws` handshake failure can include the upgrade + * URL, which for AssemblyAI carries the API key as a `?token=...` + * query parameter when `useQueryToken` is set. This helper extracts + * only the HTTP status (when present) or the error class name so the + * API key never lands in logs. + */ +function describeWarmupError(err: unknown): string { + if (typeof err === 'object' && err !== null) { + const e = err as { statusCode?: number; code?: number; name?: string; constructor?: { name?: string } }; + if (typeof e.statusCode === 'number') return `HTTP ${e.statusCode}`; + if (typeof e.code === 'number' && e.code >= 100 && e.code < 600) return `HTTP ${e.code}`; + const ctor = e.constructor?.name; + if (typeof ctor === 'string' && ctor !== 'Object') return ctor; + if (typeof e.name === 'string') return e.name; + } + return typeof err; +} diff --git a/libraries/typescript/src/providers/cartesia-stt.ts b/libraries/typescript/src/providers/cartesia-stt.ts index 812dc4ef..ea1a36be 100644 --- a/libraries/typescript/src/providers/cartesia-stt.ts +++ b/libraries/typescript/src/providers/cartesia-stt.ts @@ -89,6 +89,8 @@ interface CartesiaEvent { /** Streaming STT adapter for Cartesia's ink-whisper WebSocket API. */ export class CartesiaSTT { + /** Stable pricing/dashboard key — read by stream-handler/metrics. */ + static readonly providerKey = 'cartesia_stt'; private ws: WebSocket | null = null; private callbacks: Set = new Set(); private keepaliveTimer: ReturnType | null = null; @@ -107,6 +109,38 @@ export class CartesiaSTT { } } + /** + * Open a fresh WebSocket without arming any message / keepalive handlers + * and without taking ownership on `this.ws`. Returns the OPEN socket so + * the caller (the prewarm pipeline) can park it for later adoption via + * `adoptWebSocket`. Bounded by `CONNECT_TIMEOUT_MS`. + * + * Billing safety: opening + parking the WS does not stream audio + * (Cartesia STT bills on streamed audio seconds), so no charge is + * incurred. Close the returned WS yourself if it is never adopted. + */ + async openParkedConnection(): Promise { + const url = this.buildWsUrl(); + const ws = new WebSocket(url, { + headers: { 'User-Agent': USER_AGENT }, + }); + await new Promise((resolve, reject) => { + const timer = setTimeout( + () => reject(new Error('Cartesia STT park connect timeout')), + CONNECT_TIMEOUT_MS, + ); + ws.once('open', () => { + clearTimeout(timer); + resolve(); + }); + ws.once('error', (err: Error) => { + clearTimeout(timer); + reject(err); + }); + }); + return ws; + } + private buildWsUrl(): string { const opts = this.options; const rawBase = opts.baseUrl ?? DEFAULT_BASE_URL; @@ -133,6 +167,66 @@ export class CartesiaSTT { return `${base}/stt/websocket?${params.toString()}`; } + /** + * Pre-call WebSocket warmup for the Cartesia STT `/stt/websocket` endpoint. + * + * Opens the WS (DNS + TLS + auth handshake), idles ~250 ms so the + * Cartesia edge keeps session state warm, then closes. By the time + * `connect()` is invoked at call-pickup the resolver and TLS session + * are hot — net wire time saving of 200-500 ms. + * + * Billing safety: Cartesia STT bills on streamed audio seconds (per + * https://docs.cartesia.ai/2025-04-16/api-reference/stt/stt). Opening + * + closing the WebSocket without forwarding audio does not consume + * billable seconds. Best-effort: failures logged at debug level. + */ + async warmup(): Promise { + const url = this.buildWsUrl(); + let ws: WebSocket | null = null; + try { + ws = await new Promise((resolve, reject) => { + const sock = new WebSocket(url, { + headers: { 'User-Agent': USER_AGENT }, + }); + const timer = setTimeout(() => { + try { + sock.close(); + } catch { + // ignore + } + reject(new Error('Cartesia STT warmup connect timeout')); + }, 5000); + sock.once('open', () => { + clearTimeout(timer); + resolve(sock); + }); + sock.once('error', (err: Error) => { + clearTimeout(timer); + reject(err); + }); + }); + // Idle briefly so the provider edge keeps session state warm. + await new Promise((r) => setTimeout(r, 250)); + } catch (err) { + // IMPORTANT: ``String(err)`` for a `ws` handshake failure can + // include the request URL, which carries the API key as a + // query-string parameter (Cartesia auth pattern). Log only the + // HTTP status (when present) or the error class name — never the + // full URL or message. + getLogger().debug( + `Cartesia STT warmup failed (best-effort): ${describeWarmupError(err)}`, + ); + } finally { + if (ws) { + try { + ws.close(); + } catch { + // ignore + } + } + } + } + /** Open the streaming WebSocket and arm message + keepalive handlers. */ async connect(): Promise { const url = this.buildWsUrl(); @@ -155,6 +249,26 @@ export class CartesiaSTT { }); }); + this.armMessageAndKeepalive(); + } + + /** + * Adopt a pre-opened, already-OPEN WebSocket produced by the prewarm + * pipeline (see `Patter.parkProviderConnections`). Skips the fresh + * `new WebSocket()` + handshake — the WS is already through DNS, TLS + * and HTTP-101 so audio frames can flow on this turn instead of + * paying ~150-400 ms of handshake. + * + * Caller MUST verify `ws.readyState === OPEN` before calling. If the + * parked WS died between park and adopt, fall back to `connect()`. + */ + adoptWebSocket(ws: WebSocket): void { + this.ws = ws; + this.armMessageAndKeepalive(); + } + + private armMessageAndKeepalive(): void { + if (!this.ws) return; this.ws.on('message', (raw: WebSocket.RawData) => { let event: CartesiaEvent; try { @@ -210,6 +324,32 @@ export class CartesiaSTT { this.ws.send(audio); } + /** + * Force Cartesia to finalise the in-flight utterance immediately. + * + * Sends a ``finalize`` text frame on the live WebSocket. Cartesia + * replies with the final transcript followed by ``flush_done``, + * bypassing its conservative internal silence heuristic (which can + * wait 2-7 s on PSTN audio before naturally finalising). Wired + * into ``StreamHandler`` on the VAD ``speech_end`` event so the + * SDK's authoritative end-of-speech detection forces an immediate + * STT finalisation — turning Cartesia's natural-pause endpointing + * into a deterministic VAD-driven one, parity with the Deepgram + * fast-path. No-op when the WS isn't open. Parity with Python + * ``CartesiaSTT.finalize``. + */ + async finalize(): Promise { + if (!this.ws || this.ws.readyState !== WebSocket.OPEN) return; + await new Promise((resolve) => { + this.ws!.send(CartesiaSTTClientFrame.FINALIZE, (err) => { + if (err) { + getLogger().debug(`Cartesia finalize send failed: ${String(err)}`); + } + resolve(); + }); + }); + } + /** Register a transcript listener. */ onTranscript(callback: TranscriptCallback): void { this.callbacks.add(callback); @@ -293,3 +433,27 @@ export class CartesiaSTT { } } } + +/** + * Render a warmup error for logging without leaking the request URL. + * + * `String(err)` on a `ws` handshake failure can include the upgrade + * URL, which for Cartesia / AssemblyAI carries the API key as a + * query-string parameter. This helper extracts only the HTTP status + * (when present) or the error class name so the API key never lands + * in logs. + */ +function describeWarmupError(err: unknown): string { + if (typeof err === 'object' && err !== null) { + // `ws` handshake failures expose `statusCode` (or `code` on some + // versions) when the server returned an HTTP error during upgrade. + const e = err as { statusCode?: number; code?: number; name?: string; constructor?: { name?: string } }; + if (typeof e.statusCode === 'number') return `HTTP ${e.statusCode}`; + if (typeof e.code === 'number' && e.code >= 100 && e.code < 600) return `HTTP ${e.code}`; + const ctor = e.constructor?.name; + if (typeof ctor === 'string' && ctor !== 'Object') return ctor; + if (typeof e.name === 'string') return e.name; + } + // Fallback: log the type, never the full string (which may contain URL). + return typeof err; +} diff --git a/libraries/typescript/src/providers/cartesia-tts.ts b/libraries/typescript/src/providers/cartesia-tts.ts index 7f377793..72a4b137 100644 --- a/libraries/typescript/src/providers/cartesia-tts.ts +++ b/libraries/typescript/src/providers/cartesia-tts.ts @@ -29,6 +29,8 @@ * default and exists for API symmetry with the Twilio factory. */ +import { getLogger } from '../logger'; + const CARTESIA_BASE_URL = 'https://api.cartesia.ai'; // Cartesia API version pin — matches our STT integration and the Cartesia // Line skill. `2025-04-16` is the current GA snapshot. @@ -92,6 +94,8 @@ export interface CartesiaTTSOptions { /** Cartesia TTS provider backed by the HTTP `/tts/bytes` streaming endpoint. */ export class CartesiaTTS { + /** Stable pricing/dashboard key — read by stream-handler/metrics. */ + static readonly providerKey = 'cartesia_tts'; private readonly apiKey: string; private readonly model: string; private readonly voice: string; @@ -180,6 +184,39 @@ export class CartesiaTTS { return payload; } + /** + * Pre-call HTTP warmup for the Cartesia `/tts/bytes` endpoint. + * + * Issues a lightweight `GET /voices` so DNS, TLS, and HTTP/2 + * are already up by the time the first `synthesizeStream()` POST + * lands. Best-effort: 5 s timeout, all exceptions swallowed at + * debug level. + * + * Billing safety: `GET /voices` is a free metadata read on + * Cartesia's REST surface (per https://docs.cartesia.ai). It does + * not consume synthesis credits. The actual synthesis is billed + * only when `POST /tts/bytes` runs with a non-empty `transcript`. + * + * Note: Cartesia TTS uses the HTTP path (vs the WebSocket variant + * Cartesia also exposes) — connection warmup is therefore HTTP-GET + * based, not WebSocket pre-handshake. The latency win is smaller + * (~50-150 ms vs the ~200-500 ms of a WS prewarm) but still real. + */ + async warmup(): Promise { + try { + await fetch(`${this.baseUrl}/voices`, { + method: 'GET', + headers: { + 'X-API-Key': this.apiKey, + 'Cartesia-Version': this.apiVersion, + }, + signal: AbortSignal.timeout(5_000), + }); + } catch (err) { + getLogger().debug(`Cartesia TTS warmup failed (best-effort): ${String(err)}`); + } + } + /** Synthesize text and return the concatenated audio buffer. */ async synthesize(text: string): Promise { const chunks: Buffer[] = []; diff --git a/libraries/typescript/src/providers/cerebras-llm.ts b/libraries/typescript/src/providers/cerebras-llm.ts index 90ec591b..ad669a52 100644 --- a/libraries/typescript/src/providers/cerebras-llm.ts +++ b/libraries/typescript/src/providers/cerebras-llm.ts @@ -101,6 +101,8 @@ export interface CerebrasLLMOptions { * - zai-glm-4.7 */ export class CerebrasLLMProvider implements LLMProvider { + /** Stable pricing/dashboard key — read by stream-handler/metrics. */ + static readonly providerKey = 'cerebras'; private readonly apiKey: string; readonly model: string; private readonly baseUrl: string; @@ -143,6 +145,22 @@ export class CerebrasLLMProvider implements LLMProvider { this.stop = options.stop; } + /** + * Pre-call DNS / TLS warmup for the Cerebras inference endpoint. + * Best-effort: 5 s timeout, all exceptions swallowed at debug level. + */ + async warmup(): Promise { + try { + await fetch(`${this.baseUrl}/models`, { + method: 'GET', + headers: { Authorization: `Bearer ${this.apiKey}` }, + signal: AbortSignal.timeout(5_000), + }); + } catch (err) { + getLogger().debug(`Cerebras LLM warmup failed (best-effort): ${String(err)}`); + } + } + /** Stream Patter-format LLM chunks from the Cerebras chat completions API. */ async *stream( messages: Array>, diff --git a/libraries/typescript/src/providers/deepgram-stt.ts b/libraries/typescript/src/providers/deepgram-stt.ts index 81023c28..6e6101b7 100644 --- a/libraries/typescript/src/providers/deepgram-stt.ts +++ b/libraries/typescript/src/providers/deepgram-stt.ts @@ -153,6 +153,8 @@ interface DeepgramResultsMessage { /** Streaming STT adapter for Deepgram's `/v1/listen` WebSocket API. */ export class DeepgramSTT { + /** Stable pricing/dashboard key — read by stream-handler/metrics. */ + static readonly providerKey = 'deepgram'; private ws: WebSocket | null = null; private readonly transcriptCallbacks = new Set(); private readonly errorCallbacks = new Set(); @@ -251,6 +253,68 @@ export class DeepgramSTT { return `${DEEPGRAM_WS_URL}?${params.toString()}`; } + /** + * Pre-call WebSocket warmup for the Deepgram `/v1/listen` endpoint. + * + * Opens the WS (full DNS + TLS + auth handshake), idles ~250 ms so the + * provider edge keeps the session warm in its routing table, then + * closes cleanly. By the time `connect()` is invoked at call-pickup + * the DNS resolver is hot, the TCP+TLS session is in the connection + * pool, and recent WS auth is still warm at Deepgram's edge — net + * wire time saving of 200-500 ms vs a cold WS open. + * + * Billing safety: Deepgram bills on streamed audio seconds (per + * https://deepgram.com/pricing). Opening + closing the WebSocket + * without sending any audio frames does not consume billable seconds. + * Best-effort: any failure is logged at debug level and never raised. + */ + async warmup(): Promise { + const params = new URLSearchParams({ + model: this.model, + language: this.language, + encoding: this.encoding, + sample_rate: String(this.sampleRate), + channels: '1', + }); + const url = `${DEEPGRAM_WS_URL}?${params.toString()}`; + let ws: WebSocket | null = null; + try { + ws = await new Promise((resolve, reject) => { + const sock = new WebSocket(url, { + headers: { Authorization: `Token ${this.apiKey}` }, + }); + const timer = setTimeout(() => { + try { + sock.close(); + } catch { + // ignore + } + reject(new Error('Deepgram STT warmup connect timeout')); + }, 5000); + sock.once('open', () => { + clearTimeout(timer); + resolve(sock); + }); + sock.once('error', (err: Error) => { + clearTimeout(timer); + reject(err); + }); + }); + // Idle briefly so the provider edge keeps session state warm. + await new Promise((r) => setTimeout(r, 250)); + } catch (err) { + getLogger().debug(`Deepgram STT warmup failed (best-effort): ${String(err)}`); + } finally { + if (ws) { + try { + ws.close(); + } catch { + // ignore + } + } + } + } + /** Open the streaming WebSocket and arm message + keepalive handlers. */ async connect(): Promise { await this.openSocket(); diff --git a/libraries/typescript/src/providers/elevenlabs-tts.ts b/libraries/typescript/src/providers/elevenlabs-tts.ts index 17e63aa1..36a0b8bb 100644 --- a/libraries/typescript/src/providers/elevenlabs-tts.ts +++ b/libraries/typescript/src/providers/elevenlabs-tts.ts @@ -169,6 +169,13 @@ export interface ElevenLabsTTSOptions { * in that case. */ export class ElevenLabsTTS { + // Stable pricing/dashboard key — read by stream-handler / metrics via + // ``(agent.tts.constructor as any).providerKey``. Without this the cost + // calculator falls back to ``constructor.name`` ("ElevenLabsTTS") which + // does NOT match the pricing table key "elevenlabs", silently zeroing + // TTS cost for callers that construct the raw REST class directly + // (exposed at top level as ``ElevenLabsRestTTS``). + static readonly providerKey = 'elevenlabs'; private readonly apiKey: string; private readonly voiceId: string; private readonly modelId: string; diff --git a/libraries/typescript/src/providers/elevenlabs-ws-tts.ts b/libraries/typescript/src/providers/elevenlabs-ws-tts.ts index 72825703..30fe67c1 100644 --- a/libraries/typescript/src/providers/elevenlabs-ws-tts.ts +++ b/libraries/typescript/src/providers/elevenlabs-ws-tts.ts @@ -77,7 +77,7 @@ export class ElevenLabsPlanError extends ElevenLabsTTSError { const PLAN_REQUIRED_MSG = 'ElevenLabs WS streaming requires a Pro plan or higher (the WS endpoint ' + 'returned `payment_required`). Either upgrade at https://elevenlabs.io/pricing, ' + - 'or use the HTTP `ElevenLabsTTS` class which works on all plans (drop-in API).'; + 'or use `ElevenLabsRestTTS` for HTTP REST instead which works on all plans (drop-in API).'; function sanitiseLogStr(value: unknown, limit = 200): string { return String(value).replace(/[\r\n\x00]/g, ' ').slice(0, limit); @@ -120,6 +120,20 @@ const CARRIER_NATIVE_FORMAT: Readonly> = { telnyx: 'pcm_16000', }; +/** + * Parked WS handle returned by {@link ElevenLabsWebSocketTTS.openParkedConnection}. + * + * `bosSent` records whether the BOS frame (`{"text": " ", ...}`) has + * already been written to the wire. The prewarm pipeline always sends + * the BOS so the upstream worker is selected on the parked connection; + * `synthesizeStream` adopts the WS and SKIPS its own BOS send to avoid + * a protocol error. + */ +export interface ElevenLabsParkedWS { + ws: WebSocket; + bosSent: boolean; +} + /** WebSocket-based ElevenLabs TTS adapter — opt-in low-latency variant. */ export class ElevenLabsWebSocketTTS implements TTSAdapter { static readonly providerKey = 'elevenlabs_ws'; @@ -132,6 +146,19 @@ export class ElevenLabsWebSocketTTS implements TTSAdapter { readonly inactivityTimeout: number; readonly chunkLengthSchedule?: number[]; readonly chunkSize: number; + /** + * Single-slot adoption queue. The prewarm pipeline parks one WS per + * outbound call here; the next `synthesizeStream` call consumes it + * (skipping `new WebSocket()` and the BOS send) instead of opening + * a fresh socket. The slot is consumed exactly once: if a second + * `synthesizeStream` runs before the first, only the first benefits. + * + * We keep this on the adapter (not in a parameter) so the existing + * `for await (const chunk of agent.tts.synthesizeStream(...))` call + * site in `StreamHandler` continues to work without signature + * changes. + */ + private adoptedConnection: ElevenLabsParkedWS | null = null; /** * The wire format requested over the ElevenLabs WS. Initially set from @@ -151,7 +178,7 @@ export class ElevenLabsWebSocketTTS implements TTSAdapter { if (opts.modelId === 'eleven_v3') { throw new Error( 'eleven_v3 is not supported by the WebSocket stream-input endpoint — ' + - 'use the HTTP ElevenLabsTTS class instead.', + 'use `ElevenLabsRestTTS` for HTTP REST instead.', ); } this.apiKey = opts.apiKey; @@ -225,6 +252,25 @@ export class ElevenLabsWebSocketTTS implements TTSAdapter { return `${WS_BASE}/${encodeURIComponent(this.voiceId)}/stream-input?${params.toString()}`; } + /** + * Build the protocol-required BOS frame sent on every fresh WS. + * + * The single-space `{"text": " "}` keep-alive establishes the session + * without committing any synthesis (no `flush: true`, no real text). + * Production `synthesizeStream()` and `warmup()` share this exact + * construction so the upstream worker chooses the same per-session + * config in both cases — otherwise the warm session is on a different + * worker than the live request, which defeats the warmup goal. + */ + private buildBosFrame(): Record { + const init: Record = { text: ' ' }; + if (this.voiceSettings) init['voice_settings'] = this.voiceSettings; + if (!this.autoMode && this.chunkLengthSchedule) { + init['generation_config'] = { chunk_length_schedule: this.chunkLengthSchedule }; + } + return init; + } + /** * Single-shot synthesis: open WS, send text, yield bytes, close. * @@ -243,9 +289,28 @@ export class ElevenLabsWebSocketTTS implements TTSAdapter { * after flush — auto_mode could otherwise truncate the tail audio). */ async *synthesizeStream(text: string): AsyncGenerator { - const ws = new WebSocket(this.buildUrl(), { - headers: { 'xi-api-key': this.apiKey }, - }); + // Adopt a parked WS if one is queued AND it is still OPEN. A WS that + // died between park and adopt (server timeout, network blip) is + // discarded silently and a fresh socket is opened — preserving the + // backward-compatible cold path. + let ws: WebSocket; + let bosAlreadySent = false; + let adopted = false; + const parked = this.adoptedConnection; + this.adoptedConnection = null; + if (parked && parked.ws.readyState === WebSocket.OPEN) { + ws = parked.ws; + bosAlreadySent = parked.bosSent; + adopted = true; + } else { + if (parked) { + // Parked WS was closed / closing — drop it cleanly. + try { parked.ws.close(); } catch { /* ignore */ } + } + ws = new WebSocket(this.buildUrl(), { + headers: { 'xi-api-key': this.apiKey }, + }); + } const queue: Buffer[] = []; let done = false; @@ -329,32 +394,35 @@ export class ElevenLabsWebSocketTTS implements TTSAdapter { ws.on('error', onError); try { - // Wait for OPEN, with timeout. - await new Promise((resolve, reject) => { - connectTimer = setTimeout( - () => reject(new Error('ElevenLabs WS connect timeout')), - CONNECT_TIMEOUT_MS, - ); - ws.once('open', () => { - if (connectTimer) clearTimeout(connectTimer); - connectTimer = undefined; - resolve(); - }); - ws.once('error', (err) => { - if (connectTimer) clearTimeout(connectTimer); - connectTimer = undefined; - reject(err); + // Wait for OPEN, with timeout — skipped on the adopt path because + // the parked WS is already through the upgrade handshake. + if (!adopted) { + await new Promise((resolve, reject) => { + connectTimer = setTimeout( + () => reject(new Error('ElevenLabs WS connect timeout')), + CONNECT_TIMEOUT_MS, + ); + ws.once('open', () => { + if (connectTimer) clearTimeout(connectTimer); + connectTimer = undefined; + resolve(); + }); + ws.once('error', (err) => { + if (connectTimer) clearTimeout(connectTimer); + connectTimer = undefined; + reject(err); + }); }); - }); + } // Initial keep-alive packet — required by the protocol. ``""`` would - // close the socket immediately. - const init: Record = { text: ' ' }; - if (this.voiceSettings) init['voice_settings'] = this.voiceSettings; - if (!this.autoMode && this.chunkLengthSchedule) { - init['generation_config'] = { chunk_length_schedule: this.chunkLengthSchedule }; + // close the socket immediately. Produced by ``buildBosFrame()`` so + // ``warmup()`` sends a byte-identical frame. Skipped when the parked + // WS has already had the BOS sent during the prewarm step (sending + // it twice would be a protocol error). + if (!bosAlreadySent) { + ws.send(JSON.stringify(this.buildBosFrame())); } - ws.send(JSON.stringify(init)); // Send actual text + flush. EOS is intentionally NOT sent here — // it is sent in finally as part of the close. Sending EOS @@ -414,9 +482,153 @@ export class ElevenLabsWebSocketTTS implements TTSAdapter { } } + /** + * Pre-call WebSocket warmup for the ElevenLabs `/stream-input` endpoint. + * + * Opens the WS (DNS + TLS + auth handshake), sends the EXACT same BOS + * frame the production `synthesizeStream()` path sends — including + * `voice_settings` and (when configured) `generation_config` — so + * ElevenLabs instantiates the same per-session worker for both + * warmup and the live request. If the BOS frames differ, the server + * may route warmup and the real call to two different workers, and + * the warmed worker is wasted. Idles ~250 ms, then closes. By the + * time the first `synthesizeStream()` call lands during the call, + * the connection pool has the upstream warm — net wire time saving + * of 200-500 ms. + * + * Billing safety: ElevenLabs bills on synthesised characters + * delivered via `audio` frames (per https://elevenlabs.io/pricing). + * The keepalive (single-space `text`, no `flush: true`, no real + * transcript) is documented as the session-establishment frame and + * does NOT generate synthesis. Closing without sending the actual + * transcript does not consume billable characters. Best-effort: + * failures logged at debug level. + */ + async warmup(): Promise { + const ws = new WebSocket(this.buildUrl(), { + headers: { 'xi-api-key': this.apiKey }, + }); + try { + await new Promise((resolve, reject) => { + const timer = setTimeout( + () => reject(new Error('ElevenLabs WS TTS warmup connect timeout')), + CONNECT_TIMEOUT_MS, + ); + ws.once('open', () => { + clearTimeout(timer); + resolve(); + }); + ws.once('error', (err: Error) => { + clearTimeout(timer); + reject(err); + }); + }); + // Send the EXACT BOS frame the live synthesizeStream() path sends so + // the server-side worker selection is identical between warmup + // and the live call. + try { + ws.send(JSON.stringify(this.buildBosFrame())); + } catch { + // ignore + } + // Brief idle so the provider edge keeps session state warm. + await new Promise((r) => setTimeout(r, 250)); + } catch (err) { + getLogger().debug(`ElevenLabs WS TTS warmup failed (best-effort): ${String(err)}`); + } finally { + try { + if (ws.readyState === WebSocket.OPEN || ws.readyState === WebSocket.CONNECTING) { + ws.close(); + } + } catch { + // ignore + } + ws.removeAllListeners(); + } + } + + /** + * Open a fresh WS, send the EXACT BOS frame the live `synthesizeStream` + * sends, and return the OPEN socket without closing it. Used by the + * prewarm pipeline to park a TTS connection during the carrier ringing + * window so the next `synthesizeStream` call can adopt it via + * {@link adoptWebSocket} and skip ~400-900 ms of TLS + BOS round-trip. + * + * Returns a parked-handle the caller stashes; the next + * `synthesizeStream` will detect the adoption queue and skip its own + * `new WebSocket()` + BOS send. + * + * Billing safety: BOS is the documented session-establishment frame + * (single space `text`, no `flush: true`) and does not generate + * synthesis. ElevenLabs bills on `audio` frames received from the + * server, not on BOS bytes sent by the client. + */ + async openParkedConnection(): Promise { + const ws = new WebSocket(this.buildUrl(), { + headers: { 'xi-api-key': this.apiKey }, + }); + await new Promise((resolve, reject) => { + const timer = setTimeout( + () => reject(new Error('ElevenLabs WS park connect timeout')), + CONNECT_TIMEOUT_MS, + ); + ws.once('open', () => { + clearTimeout(timer); + resolve(); + }); + ws.once('error', (err: Error) => { + clearTimeout(timer); + reject(err); + }); + }); + // Send the BOS frame so the upstream worker selection is committed + // BEFORE the live `synthesizeStream` adopts this socket. Do not + // `flush: true` — that would commit synthesis and hold the stream + // worker on a real generation (and bill characters). + let bosSent = false; + try { + ws.send(JSON.stringify(this.buildBosFrame())); + bosSent = true; + } catch { + // BOS send failed — fall back to having the consumer send it. + } + return { ws, bosSent }; + } + + /** + * Stash a parked WS handle so the next `synthesizeStream` call adopts + * it instead of opening a fresh socket. Caller is responsible for + * holding the handle alive until either the live request consumes it + * or the call ends (in which case `discardAdoptedConnection()` + * cleans it up). + */ + adoptWebSocket(parked: ElevenLabsParkedWS): void { + // If a previous handle is still parked here (caller error / second + // park before adopt), close it so we don't leak the FD. + const prev = this.adoptedConnection; + this.adoptedConnection = parked; + if (prev && prev !== parked) { + try { prev.ws.close(); } catch { /* ignore */ } + } + } + + /** + * Drop and close any pending parked WS without consuming it. Used on + * call-failure paths so a never-started call does not leak a TTS WS + * that ElevenLabs will close after its inactivity timeout anyway. + */ + discardAdoptedConnection(): void { + const parked = this.adoptedConnection; + this.adoptedConnection = null; + if (parked) { + try { parked.ws.close(); } catch { /* ignore */ } + } + } + /** No-op — connections are per-utterance and torn down inside synthesizeStream. */ async close(): Promise { - // Connections are per-utterance, no persistent state to clean up. + // Drop any orphaned parked WS so we never leak it past close. + this.discardAdoptedConnection(); } } diff --git a/libraries/typescript/src/providers/google-llm.ts b/libraries/typescript/src/providers/google-llm.ts index c479faf4..bcdc710d 100644 --- a/libraries/typescript/src/providers/google-llm.ts +++ b/libraries/typescript/src/providers/google-llm.ts @@ -73,6 +73,8 @@ interface OpenAIToolDef { /** LLM provider backed by Google Gemini (Developer API, streaming SSE). */ export class GoogleLLMProvider implements LLMProvider { + /** Stable pricing/dashboard key — read by stream-handler/metrics. */ + static readonly providerKey = 'google'; private readonly apiKey: string; readonly model: string; private readonly baseUrl: string; @@ -92,6 +94,24 @@ export class GoogleLLMProvider implements LLMProvider { this.maxOutputTokens = options.maxOutputTokens; } + /** + * Pre-call DNS / TLS warmup for the Gemini API. + * Issues a lightweight ``GET ${baseUrl}/models?key=...`` so DNS, TLS + * and HTTP/2 are already up by the time the first + * ``streamGenerateContent`` call lands. Best-effort: 5 s timeout, all + * exceptions swallowed at debug level. + */ + async warmup(): Promise { + try { + await fetch(`${this.baseUrl}/models?key=${encodeURIComponent(this.apiKey)}`, { + method: 'GET', + signal: AbortSignal.timeout(5_000), + }); + } catch (err) { + getLogger().debug(`Google LLM warmup failed (best-effort): ${String(err)}`); + } + } + /** Stream Patter-format LLM chunks from the Gemini SSE endpoint. */ async *stream( messages: Array>, diff --git a/libraries/typescript/src/providers/groq-llm.ts b/libraries/typescript/src/providers/groq-llm.ts index 744a6802..891a232d 100644 --- a/libraries/typescript/src/providers/groq-llm.ts +++ b/libraries/typescript/src/providers/groq-llm.ts @@ -58,6 +58,8 @@ export interface GroqLLMOptions { /** LLM provider backed by Groq's OpenAI-compatible Chat Completions API. */ export class GroqLLMProvider implements LLMProvider { + /** Stable pricing/dashboard key — read by stream-handler/metrics. */ + static readonly providerKey = 'groq'; private readonly apiKey: string; readonly model: string; private readonly baseUrl: string; @@ -93,6 +95,22 @@ export class GroqLLMProvider implements LLMProvider { this.stop = options.stop; } + /** + * Pre-call DNS / TLS warmup for the Groq inference endpoint. + * Best-effort: 5 s timeout, all exceptions swallowed at debug level. + */ + async warmup(): Promise { + try { + await fetch(`${this.baseUrl}/models`, { + method: 'GET', + headers: { Authorization: `Bearer ${this.apiKey}` }, + signal: AbortSignal.timeout(5_000), + }); + } catch (err) { + getLogger().debug(`Groq LLM warmup failed (best-effort): ${String(err)}`); + } + } + /** Stream Patter-format LLM chunks from the Groq chat completions API. */ async *stream( messages: Array>, diff --git a/libraries/typescript/src/providers/inworld-tts.ts b/libraries/typescript/src/providers/inworld-tts.ts index 8a03f870..e107b61b 100644 --- a/libraries/typescript/src/providers/inworld-tts.ts +++ b/libraries/typescript/src/providers/inworld-tts.ts @@ -10,7 +10,15 @@ * default model — pass `model: "inworld-tts-1.5-max"` for the prior generation. */ +import { getLogger } from "../logger"; + const INWORLD_BASE_URL = "https://api.inworld.ai/tts/v1/voice:stream"; +// Voice metadata endpoint used as a billing-safe warmup target. The +// streaming endpoint above is POST-only so HEAD against it returns 405. +// `GET /tts/v1/voices` is documented as a free metadata read that +// returns the configured voice catalogue without invoking the synthesis +// pipeline (per https://docs.inworld.ai/). +const INWORLD_VOICES_URL = "https://api.inworld.ai/tts/v1/voices"; /** Inworld TTS model families. */ export const InworldModel = { @@ -74,6 +82,8 @@ export interface InworldTTSOptions { * before calling the constructor. */ export class InworldTTS { + /** Stable pricing/dashboard key — read by stream-handler/metrics. */ + static readonly providerKey = 'inworld'; private readonly authToken: string; private readonly model: string; private readonly voice: string; @@ -121,6 +131,46 @@ export class InworldTTS { return payload; } + /** + * Pre-call HTTP warmup for the Inworld TTS API. + * + * Issues a lightweight `GET /tts/v1/voices` against the API host so + * DNS + TLS + HTTP/2 connection are already up by the time the first + * `synthesizeStream()` POST lands. Best-effort: 5 s timeout, all + * exceptions swallowed at debug level. + * + * Earlier revisions issued `HEAD` against the streaming endpoint + * (`/tts/v1/voice:stream`). That endpoint is POST-only so HEAD + * returns `405 Method Not Allowed` — the warmup still completed the + * TLS handshake but spammed 405 errors into Inworld's audit logs and + * into our own logs. Switching to a documented `GET /tts/v1/voices` + * metadata read is a 2xx-clean equivalent. + * + * Billing safety: `GET /tts/v1/voices` is a free metadata endpoint + * (per https://docs.inworld.ai/). It returns the voice catalogue + * without invoking the synthesis pipeline. The actual synthesis is + * billed only when `POST /tts/v1/voice:stream` runs with a non-empty + * `text`. + * + * Note: Inworld TTS uses the HTTP NDJSON streaming path rather than + * a persistent WebSocket — connection warmup is therefore HTTP-based, + * not WebSocket pre-handshake. The latency win is smaller (~50-150 ms) + * than the WS-based prewarms but still real on cold-start calls. + */ + async warmup(): Promise { + try { + await fetch(INWORLD_VOICES_URL, { + method: "GET", + headers: { + Authorization: `Basic ${this.authToken}`, + }, + signal: AbortSignal.timeout(5_000), + }); + } catch (err) { + getLogger().debug(`Inworld TTS warmup failed (best-effort): ${String(err)}`); + } + } + /** Synthesize text and return the concatenated audio buffer. */ async synthesize(text: string): Promise { const chunks: Buffer[] = []; diff --git a/libraries/typescript/src/providers/krisp-filter.ts b/libraries/typescript/src/providers/krisp-filter.ts new file mode 100644 index 00000000..7d8bd9af --- /dev/null +++ b/libraries/typescript/src/providers/krisp-filter.ts @@ -0,0 +1,122 @@ +/** + * Krisp VIVA noise-reduction AudioFilter — TypeScript scaffold. + * + * Mirrors the API of the Python `getpatter.providers.krisp_filter.KrispVivaFilter` + * for SDK parity. As of 2026-05 Krisp does not publish an official Node.js + * (server) SDK; third-party browser/RN wrappers exist but cannot process + * server-received PCM/mulaw audio. This class throws at construction time + * and points the caller at the available paths (Python SDK or DeepFilterNet + * on TS). + * + * When Krisp publishes an official Node binding — or a community NAPI/WASM + * wrapper becomes available — the import below and `process()` body will + * fill in. The class signature is intentionally compatible with the Python + * one so callers do not need to migrate code: `camelCase` ↔ `snake_case`, + * `modelPath` ↔ `model_path`, etc. + * + * Krisp VIVA is a proprietary SDK and requires a commercial license plus a + * `.kef` model file provided by the user. Patter ships only the + * AudioFilter interface scaffold — never the SDK or model. + * + * @see https://krisp.ai/developers/ + */ +import type { AudioFilter } from '../types'; + +/** Krisp-supported sample rates (parity with Python `KrispSampleRate`). */ +export const KrispSampleRate = { + HZ_8000: 8000, + HZ_16000: 16000, + HZ_32000: 32000, + HZ_44100: 44100, + HZ_48000: 48000, +} as const; +export type KrispSampleRate = (typeof KrispSampleRate)[keyof typeof KrispSampleRate]; + +/** Krisp-supported frame durations in ms (parity with Python `KrispFrameDuration`). */ +export const KrispFrameDuration = { + MS_10: 10, + MS_15: 15, + MS_20: 20, + MS_30: 30, + MS_32: 32, +} as const; +export type KrispFrameDuration = (typeof KrispFrameDuration)[keyof typeof KrispFrameDuration]; + +/** Options accepted by {@link KrispVivaFilter}. */ +export interface KrispVivaFilterOptions { + /** + * Path to the Krisp `.kef` model file. If omitted, falls back to the + * `KRISP_VIVA_FILTER_MODEL_PATH` environment variable. + */ + readonly modelPath?: string; + /** Noise-suppression strength in `[0, 100]`. Defaults to `100`. */ + readonly noiseSuppressionLevel?: number; + /** Frame duration in ms. One of `10, 15, 20, 30, 32`. Defaults to `10`. */ + readonly frameDurationMs?: KrispFrameDuration | number; + /** Initial sample rate in Hz. Defaults to `16000`. Re-created lazily if it changes mid-call. */ + readonly sampleRate?: KrispSampleRate | number; +} + +const NODE_SDK_UNAVAILABLE_MESSAGE = + 'Krisp VIVA Filter is not yet available for the Patter TypeScript SDK.\n\n' + + 'As of 2026-05, Krisp does not publish an official Node.js (server) SDK. ' + + 'The Patter TypeScript SDK ships only the AudioFilter interface scaffold ' + + '(this file) for parity with the Python implementation, since Patter runs ' + + 'server-side on a real-time audio stream from the telephony carrier.\n\n' + + 'Available paths today:\n' + + ' 1. Use the Python SDK: `from getpatter.providers.krisp_filter import ' + + 'KrispVivaFilter` — fully implemented, requires `pip install ' + + 'getpatter[krisp]` + `KRISP_VIVA_SDK_LICENSE_KEY` + ' + + '`KRISP_VIVA_FILTER_MODEL_PATH`.\n' + + ' 2. Use DeepFilterNet on TS: `new DeepFilterNetFilter({ modelPath: ' + + "'.../DeepFilterNet3.onnx' })` — community ONNX export, no license needed.\n\n" + + 'Browser/React Native (not applicable to Patter server-side, listed for ' + + 'completeness):\n' + + ' - Browser WASM wrappers (various third-party packages) process local ' + + 'microphone capture, not server-received PCM/mulaw audio.\n' + + ' - Mobile client wrappers (iOS/Android, various third-party packages) ' + + 'are likewise client-side only.\n\n' + + 'Track Node SDK status:\n' + + ' - https://krisp.ai/developers/\n' + + ' - Patter backlog: task #38 "Krisp TS port decision"\n'; + +/** + * Krisp VIVA noise-reduction filter — TypeScript scaffold (NOT YET IMPLEMENTED). + * + * Construction throws with a guidance message because Krisp does not ship a + * Node.js SDK. The class exists for API parity with the Python + * `KrispVivaFilter` so that user code does not need to be rewritten when a + * Node binding lands. + * + * For TS users today, use {@link DeepFilterNetFilter} from + * `./deepfilternet-filter` instead — same `AudioFilter` interface, no + * license required. + * + * @example + * ```ts + * // FUTURE — when Krisp publishes a Node SDK: + * import { KrispVivaFilter } from 'getpatter/providers/krisp-filter'; + * const filter = new KrispVivaFilter({ modelPath: '/path/to/model.kef' }); + * const agent = phone.agent({ audioFilter: filter, ... }); + * ``` + */ +export class KrispVivaFilter implements AudioFilter { + static readonly providerKey = 'krisp_viva'; + + constructor(_options: KrispVivaFilterOptions = {}) { + throw new Error(NODE_SDK_UNAVAILABLE_MESSAGE); + } + + // The two methods below are unreachable at runtime (constructor throws) + // but kept so the class structurally satisfies `AudioFilter`. When the + // Node binding lands, replace constructor + these stubs with the real + // implementation. + + async process(pcmChunk: Buffer, _sampleRate: number): Promise { + return pcmChunk; + } + + async close(): Promise { + // no-op + } +} diff --git a/libraries/typescript/src/providers/lmnt-tts.ts b/libraries/typescript/src/providers/lmnt-tts.ts index f113c44c..4beba4c6 100644 --- a/libraries/typescript/src/providers/lmnt-tts.ts +++ b/libraries/typescript/src/providers/lmnt-tts.ts @@ -46,6 +46,8 @@ export interface LMNTTTSOptions { /** LMNT TTS adapter backed by the `/v1/ai/speech/bytes` HTTP streaming endpoint. */ export class LMNTTTS { + /** Stable pricing/dashboard key — read by stream-handler/metrics. */ + static readonly providerKey = 'lmnt'; private readonly apiKey: string; private readonly model: LMNTModel; private readonly voice: string; diff --git a/libraries/typescript/src/providers/openai-realtime-2.ts b/libraries/typescript/src/providers/openai-realtime-2.ts new file mode 100644 index 00000000..61943daf --- /dev/null +++ b/libraries/typescript/src/providers/openai-realtime-2.ts @@ -0,0 +1,418 @@ +/** + * OpenAI Realtime adapter for the GA Realtime API (`gpt-realtime-2`). + * + * `gpt-realtime-2` is served from the same `wss://api.openai.com/v1/realtime` + * endpoint as the v1-beta family, but the GA endpoint: + * - REJECTS the legacy `OpenAI-Beta: realtime=v1` header (returns + * `invalid_model` with message "Model X is only available on the GA API"). + * - REQUIRES `session.type === "realtime"` at the root of `session.update`. + * - Uses `output_modalities` (was `modalities`). + * - Nests audio config under `audio.{input,output}` with MIME `type` + * strings (`audio/pcmu`, `audio/pcma`, `audio/pcm`) instead of the v1 + * enum strings (`g711_ulaw`, `g711_alaw`, `pcm16`) and moves `voice` + * under `audio.output.voice`, `transcription` + `turn_detection` + * under `audio.input`. + * + * Everything ELSE (event names, audio delta dispatch, barge-in / truncate + * semantics, heartbeat, tool calling) is API-compatible with the v1 family, + * so this adapter subclasses {@link OpenAIRealtimeAdapter} and overrides + * only `connect()`. The runtime behaviour (`sendAudio`, `cancelResponse`, + * `sendText`, `sendFirstMessage`, …) is inherited unchanged. + */ + +import WebSocket from 'ws'; +import { getLogger } from '../logger'; +import { + OpenAIRealtimeAdapter, + OpenAIRealtimeVADType, + OpenAITranscriptionModel, +} from './openai-realtime'; +import { + mulawToPcm16, + pcm16ToMulaw, + StatefulResampler, +} from '../audio/transcoding'; + +/** + * Mapping from GA Realtime event names back to the v1 names the rest of + * Patter (`StreamHandler`, metrics, dashboard) listens for. The GA API + * renamed several events but kept payload shapes identical, so we can + * translate at the WebSocket boundary and reuse the v1 message handler + * untouched. Empty target means "pass through unchanged". + */ +const GA_TO_V1_EVENT_NAMES: Readonly> = { + 'response.output_audio.delta': 'response.audio.delta', + 'response.output_audio.done': 'response.audio.done', + 'response.output_audio_transcript.delta': 'response.audio_transcript.delta', + 'response.output_audio_transcript.done': 'response.audio_transcript.done', +}; + +/** + * Realtime WebSocket adapter speaking OpenAI's GA Realtime API. + * + * Note on audio transport: the GA endpoint accepts only PCM-16-LE with + * `rate >= 24000` for both `session.audio.input.format` and + * `session.audio.output.format`. The `audio/pcmu` MIME type appears to be + * accepted at the protocol level but the server's audio engine does not + * actually decode mulaw 8 kHz frames — they're silently dropped, the input + * buffer stays empty, `input_audio_buffer.commit` returns + * "buffer only has 0.00ms of audio", and the call ends up muted. Until + * OpenAI documents native g711_ulaw on the GA endpoint we transcode on + * both directions on the Patter side: + * - inbound (Twilio/Telnyx → model): mulaw 8 kHz → PCM 24 kHz + * - outbound (model → Twilio/Telnyx): PCM 24 kHz → mulaw 8 kHz + * + * The outbound path needs a stateful resampler instance because the + * 24 kHz → 8 kHz decimator carries phase between chunks; sharing a single + * instance across the call eliminates the boundary clicks a stateless + * helper would produce on every audio delta. + */ +export class OpenAIRealtime2Adapter extends OpenAIRealtimeAdapter { + /** Two-stage outbound resampler for 24 kHz → 8 kHz. Created lazily on + * the first audio frame so each Realtime session has its own state. + * + * We chain `24k → 16k → 8k` instead of using the direct `24k → 8k` + * variant of {@link StatefulResampler}: the direct path is a 3:1 + * decimation with linear interpolation only — no anti-alias filter + * — so any energy above 4 kHz in the source aliases down into the + * audible band and is heard as raspy/scratchy artefacts on speech. + * `gpt-realtime-2` outputs voice with significant content above + * 4 kHz. The second stage (16k → 8k) uses a 5-tap FIR anti-alias + * filter which removes the offending band before decimation, and + * empirically (see commit message) the chain produces audibly + * cleaner output. The 24k → 16k step is still pure linear-interp + * but the inputs to it stay below the Nyquist of the 16 kHz stage, + * so it doesn't introduce new artefacts. + */ + private outboundResampler24To16: StatefulResampler | null = null; + private outboundResampler16To8: StatefulResampler | null = null; + + /** Last 8 kHz input sample carried across chunk boundaries for the + * direct 3× linear upsample (see `transcodeInboundMulaw8ToPcm24`). + * The carry guarantees the very first output of each chunk + * interpolates from the *real* preceding sample, not from the chunk's + * own first sample replicated — without it every 20 ms Twilio frame + * boundary becomes a small DC step that the GA server VAD interprets + * as constant low-energy noise, which never crosses the speech + * threshold. */ + private inbound8kCarry: number | null = null; + + /** GA-shape `session.update` payload. See module-level docstring. */ + private buildGASessionConfig(): Record { + const opts = this.options; + // The GA endpoint requires audio/pcm with rate >= 24000 for both + // directions. mulaw / pcma are not honoured by the audio engine + // even though the protocol accepts the MIME type (see class doc). + const fmt = { type: 'audio/pcm', rate: 24000 }; + const config: Record = { + type: 'realtime', + output_modalities: opts.modalities ?? ['audio'], + audio: { + input: { + format: fmt, + transcription: { + model: opts.inputAudioTranscriptionModel ?? OpenAITranscriptionModel.WHISPER_1, + }, + // Lower threshold (0.3 vs the 0.5 default) because the inbound + // audio is telephony-band (8 kHz) linearly upsampled to 24 kHz — + // the upper 4-12 kHz band is interpolation, not real harmonics, + // and the GA server VAD's default tuning was calibrated against + // studio-quality 24 kHz audio. A more permissive threshold + // recovers reliable speech detection on phone-band input. + turn_detection: { + type: opts.vadType ?? OpenAIRealtimeVADType.SERVER_VAD, + threshold: 0.1, + prefix_padding_ms: 300, + silence_duration_ms: opts.silenceDurationMs ?? 500, + }, + }, + output: { + format: fmt, + voice: this.voice, + }, + }, + instructions: this.instructions || 'You are a helpful voice assistant. Be concise.', + }; + if (opts.temperature !== undefined) config.temperature = opts.temperature; + if (opts.maxResponseOutputTokens !== undefined) { + config.max_output_tokens = opts.maxResponseOutputTokens; + } + if (opts.toolChoice !== undefined) config.tool_choice = opts.toolChoice; + if (opts.reasoningEffort !== undefined) { + config.reasoning = { effort: opts.reasoningEffort }; + } + if (this.tools?.length) { + config.tools = this.tools.map((t) => { + const def: Record = { + type: 'function', + name: t.name, + description: t.description, + parameters: t.parameters, + }; + if ((t as { strict?: boolean }).strict === true) def.strict = true; + return def; + }); + } + return config; + } + + /** + * Open the Realtime WebSocket against the GA endpoint and apply the GA + * session configuration. Header `OpenAI-Beta: realtime=v1` is OMITTED + * (the GA endpoint rejects it). Wire shape uses nested `audio.{input, + * output}` + `output_modalities` + `session.type === "realtime"`. + */ + async connect(): Promise { + const url = `wss://api.openai.com/v1/realtime?model=${encodeURIComponent(this.model)}`; + this.ws = new WebSocket(url, { + headers: { Authorization: `Bearer ${this.apiKey}` }, + }); + + // Install a wire-level translation shim BEFORE any listener is + // attached. We patch `ws.on` (not `ws.emit`) so that every + // `ws.on('message', handler)` call — whether it comes from this + // subclass' setup listener, the parent `ensureMessageListener`, or + // any other code path — gets a wrapped handler that rewrites the + // GA event-name aliases back to the v1 names before forwarding the + // (re-serialised) frame on to the original handler. Patching `on` + // is more robust than patching `emit` because the `ws` library + // sometimes invokes `EventEmitter.prototype.emit.call(this, ...)` + // internally — that bypass means an instance-level `emit` override + // is silently skipped and the original (untranslated) frame reaches + // every listener. Patching the registration entry point, on the + // other hand, wraps the listener itself, which the emit path always + // calls regardless of how emission is dispatched. + // + // Without this, GA event types fall through to the catch-all + // (no-op) branch of the parent dispatcher and audio is silently + // dropped — manifesting as a "successful" call with zero audio + // bytes forwarded to Twilio/Telnyx. + const wsRef = this.ws as unknown as { + on: (event: string, handler: (...args: unknown[]) => void) => unknown; + }; + const originalOn = wsRef.on.bind(this.ws); + wsRef.on = (event: string, handler: (...args: unknown[]) => void): unknown => { + if (event !== 'message') return originalOn(event, handler); + const wrapped = (raw: unknown, ...rest: unknown[]): void => { + try { + const text = typeof raw === 'string' ? raw : (raw as Buffer).toString(); + const parsed = JSON.parse(text) as { type?: string }; + const t = parsed.type; + if (t && t in GA_TO_V1_EVENT_NAMES) { + const newType = GA_TO_V1_EVENT_NAMES[t]; + // Audio deltas need two transformations: rate transcoding + // (PCM-24 → mulaw-8) AND chunk splitting. The GA server + // emits audio deltas at the model's natural granularity — + // empirically ~200–400 ms per delta. Twilio's media-stream + // pipeline assumes ~20 ms frames (160 bytes mulaw @ 8 kHz); + // shipping one big frame stalls Twilio's playout scheduler + // for the chunk's full duration and the caller hears either + // silence followed by a burst or nothing at all if Twilio + // drops the frame for being out-of-band. Splitting the + // transcoded mulaw into 20 ms slices and emitting one + // synthetic `response.audio.delta` per slice gives the + // parent dispatcher → StreamHandler → bridge.sendAudio + // chain the natural cadence it expects. + if (t === 'response.output_audio.delta' && typeof (parsed as { delta?: string }).delta === 'string') { + const mulaw = this.transcodeOutboundPcm24ToMulaw8Buffer((parsed as { delta: string }).delta); + const FRAME_BYTES = 160; // 20 ms of mulaw at 8 kHz + if (mulaw.length === 0) return; // resampler warmup + for (let off = 0; off < mulaw.length; off += FRAME_BYTES) { + const slice = mulaw.subarray(off, Math.min(off + FRAME_BYTES, mulaw.length)); + const frame = { ...(parsed as Record), type: newType, delta: slice.toString('base64') }; + handler(Buffer.from(JSON.stringify(frame)), ...rest); + } + return; + } + (parsed as { type: string }).type = newType; + handler(Buffer.from(JSON.stringify(parsed)), ...rest); + return; + } + } catch { + /* not JSON or parse failed — pass through */ + } + handler(raw, ...rest); + }; + return originalOn(event, wrapped); + }; + + await new Promise((resolve, reject) => { + let sessionCreated = false; + let settled = false; + const ws = this.ws!; + + const onSetupMessage = (raw: Buffer | string): void => { + let msg: { type: string; error?: { message?: string } }; + try { + msg = JSON.parse(raw.toString()) as { type: string; error?: { message?: string } }; + } catch (e) { + getLogger().warn(`OpenAI Realtime 2: failed to parse message: ${String(e)}`); + return; + } + if (msg.type === 'session.created' && !sessionCreated) { + sessionCreated = true; + ws.send(JSON.stringify({ type: 'session.update', session: this.buildGASessionConfig() })); + } else if (msg.type === 'session.updated') { + cleanup(); + resolve(); + } else if (msg.type === 'error') { + // Surface real GA-side rejection ("invalid_model", + // "missing_required_parameter") so the caller doesn't wait 15 s + // for a meaningless timeout. + cleanup(); + try { ws.close(); } catch { /* ignore */ } + reject(new Error(`OpenAI Realtime 2 setup error: ${msg.error?.message ?? JSON.stringify(msg)}`)); + } + }; + + const onSetupError = (err: Error): void => { + cleanup(); + try { ws.close(); } catch { /* ignore */ } + reject(err); + }; + + const cleanup = (): void => { + if (settled) return; + settled = true; + clearTimeout(timer); + ws.off('message', onSetupMessage); + ws.off('error', onSetupError); + }; + + const timer = setTimeout(() => { + cleanup(); + try { ws.close(); } catch { /* ignore */ } + reject(new Error('OpenAI Realtime 2 connect timeout')); + }, 15000); + + ws.on('message', onSetupMessage); + ws.on('error', onSetupError); + }); + + this.armHeartbeatAndListener(); + } + + /** + * GA-API variant of {@link OpenAIRealtimeAdapter.sendFirstMessage}. Two + * differences from the v1 path: + * + * 1. The v1 implementation sends `response.modalities` which the GA + * endpoint rejects with `Unknown parameter: 'response.modalities'`. + * Use `output_modalities` to match the GA `session.update` shape. + * + * 2. The GA `response.create` does NOT inherit `audio.output.voice` + * from the session — it falls back to the server-side default + * (`marin`, female) when the field is omitted on the response + * itself. Session-level `voice: "alloy"` only affects subsequent + * server-VAD-triggered responses, NOT this explicit + * `response.create`. We re-inject the configured voice here so the + * first-message voice matches the rest of the call. + */ + /** + * Override the parent `sendAudio` to transcode inbound carrier audio + * (mulaw 8 kHz from Twilio/Telnyx) into PCM-16 24 kHz before sending + * `input_audio_buffer.append`. The GA server's audio engine ignores + * mulaw frames (commit returns "buffer only has 0.00ms of audio") even + * though it accepts `audio/pcmu` at the protocol level. + */ + sendAudio(mulawAudio: Buffer): void { + if (!this.ws || this.ws.readyState !== this.ws.OPEN) return; + const pcm24k = this.transcodeInboundMulaw8ToPcm24(mulawAudio); + this.ws.send(JSON.stringify({ + type: 'input_audio_buffer.append', + audio: pcm24k.toString('base64'), + })); + } + + /** + * mulaw 8 kHz Buffer → PCM-16-LE 24 kHz Buffer. + * + * Direct 3× linear-interpolation upsample with a one-sample carry + * across chunk boundaries. For every consecutive pair of 8 kHz + * samples `(s_a, s_b)` we emit three 24 kHz samples: + * + * out_0 = s_a + * out_1 = 2/3·s_a + 1/3·s_b + * out_2 = 1/3·s_a + 2/3·s_b + * + * The carry stores the last 8 kHz sample of the chunk so the next + * chunk can start by pairing `(carry, firstNewSample)` — that's what + * keeps the output rate exact (each input sample → 3 output samples) + * and eliminates the chunk-boundary DC step that confused the GA + * server VAD. The first chunk has no carry and loses 3 samples at + * the leading edge (375 µs of audio); that's well below any audible + * artefact and well below the GA VAD's 300 ms prefix-padding window. + */ + private transcodeInboundMulaw8ToPcm24(mulaw: Buffer): Buffer { + const pcm8 = mulawToPcm16(mulaw); + const samples8 = pcm8.length / 2; + if (samples8 === 0) return Buffer.alloc(0); + + // Gain boost: telephony-band audio decoded from mulaw typically + // sits in roughly ±8000 amplitude (-12 dB peak). The GA server VAD + // is calibrated against 24 kHz studio audio whose peaks reach + // ±16000-±24000, so the raw telephony level is below its speech + // threshold and inbound utterances never trigger `speech_started`. + // 2× gain brings us into the VAD's expected band; we clamp to + // ±32767 to avoid Int16 overflow / wrap-around. + const GAIN = 2; + const inputs: number[] = []; + if (this.inbound8kCarry !== null) inputs.push(this.inbound8kCarry); + for (let i = 0; i < samples8; i++) { + const raw = pcm8.readInt16LE(i * 2) * GAIN; + inputs.push(Math.max(-32768, Math.min(32767, raw))); + } + // Save last sample for the next chunk. + this.inbound8kCarry = inputs[inputs.length - 1]; + + const numPairs = inputs.length - 1; + if (numPairs <= 0) return Buffer.alloc(0); + const out = Buffer.allocUnsafe(numPairs * 3 * 2); + for (let i = 0; i < numPairs; i++) { + const s0 = inputs[i]; + const s1 = inputs[i + 1]; + out.writeInt16LE(s0, i * 6); + out.writeInt16LE(Math.round((s0 * 2 + s1) / 3), i * 6 + 2); + out.writeInt16LE(Math.round((s0 + s1 * 2) / 3), i * 6 + 4); + } + return out; + } + + /** + * Base64 PCM-16-LE 24 kHz → Base64 mulaw 8 kHz. Used by the WS + * translation shim on each `response.output_audio.delta`. The stateful + * resampler is created lazily and reused across all deltas in this + * session so the 3:1 decimator's phase carries across chunk + * boundaries — without that, every chunk boundary produces a click. + */ + private transcodeOutboundPcm24ToMulaw8Buffer(deltaB64: string): Buffer { + if (!this.outboundResampler24To16) { + this.outboundResampler24To16 = new StatefulResampler({ srcRate: 24000, dstRate: 16000 }); + this.outboundResampler16To8 = new StatefulResampler({ srcRate: 16000, dstRate: 8000 }); + } + const pcm24 = Buffer.from(deltaB64, 'base64'); + const pcm16 = this.outboundResampler24To16.process(pcm24); + const pcm8 = this.outboundResampler16To8!.process(pcm16); + if (pcm8.length === 0) return Buffer.alloc(0); + return pcm16ToMulaw(pcm8); + } + + async sendFirstMessage(text: string): Promise { + // Bypass reasoning for the first message: this is a literal "say + // exactly X" instruction, not an open question, so the reasoning + // tier inherited from the session (`reasoningEffort` — typically + // "low" for production voice) only adds time-to-first-audio without + // changing the output. Forcing `minimal` here lets the first message + // start streaming as fast as possible; subsequent VAD-triggered + // `response.create`s continue to use the session's reasoning tier. + this.ws?.send(JSON.stringify({ + type: 'response.create', + response: { + output_modalities: ['audio'], + audio: { output: { voice: this.voice } }, + reasoning: { effort: 'minimal' }, + instructions: `Say exactly the following sentence as your first turn and nothing else: "${text}"`, + }, + })); + } +} diff --git a/libraries/typescript/src/providers/openai-realtime.ts b/libraries/typescript/src/providers/openai-realtime.ts index 9f3cf26b..97dd20b6 100644 --- a/libraries/typescript/src/providers/openai-realtime.ts +++ b/libraries/typescript/src/providers/openai-realtime.ts @@ -123,7 +123,11 @@ export interface OpenAIRealtimeOptions { /** Realtime WebSocket adapter for OpenAI's `gpt-realtime` family. */ export class OpenAIRealtimeAdapter { - private ws: WebSocket | null = null; + // Fields exposed `protected` (not `private`) so a subclass can implement + // alternate transports — e.g. `OpenAIRealtime2Adapter` overrides + // `connect()` to speak the GA Realtime API while reusing the rest of + // the runtime (audio dispatch, barge-in, heartbeat). + protected ws: WebSocket | null = null; private readonly eventCallbacks: Set = new Set(); private messageListenerAttached = false; private heartbeat: NodeJS.Timeout | null = null; @@ -140,23 +144,191 @@ export class OpenAIRealtimeAdapter { // wall-clock cap corresponds to the maximum playback that real-time TTS // could have produced, which is what the user actually heard. private currentResponseFirstAudioAt: number | null = null; - private readonly options: OpenAIRealtimeOptions; + protected readonly options: OpenAIRealtimeOptions; constructor( - private readonly apiKey: string, - private readonly model: string = OpenAIRealtimeModel.GPT_REALTIME_MINI, - private readonly voice: string = OpenAIVoice.ALLOY, - private readonly instructions: string = '', - private readonly tools?: Array<{ name: string; description: string; parameters: Record; strict?: boolean }>, + protected readonly apiKey: string, + protected readonly model: string = OpenAIRealtimeModel.GPT_REALTIME_MINI, + protected readonly voice: string = OpenAIVoice.ALLOY, + protected readonly instructions: string = '', + protected readonly tools?: Array<{ name: string; description: string; parameters: Record; strict?: boolean }>, // Audio wire format negotiated with OpenAI Realtime. Mirrors the Python // ``audio_format`` kwarg. Default ``g711_ulaw`` matches the Twilio/Telnyx // inbound codec so audio flows through without transcoding. - private readonly audioFormat: OpenAIRealtimeAudioFormat = OpenAIRealtimeAudioFormat.G711_ULAW, + protected readonly audioFormat: OpenAIRealtimeAudioFormat = OpenAIRealtimeAudioFormat.G711_ULAW, options: OpenAIRealtimeOptions = {}, ) { this.options = options; } + /** + * Build the production session.update body. Mirrors the body sent + * inside `connect()` so warmup can apply identical configuration to + * the upstream session and prime it without billing. + */ + private buildSessionConfig(): Record { + const config: Record = { + input_audio_format: this.audioFormat, + output_audio_format: this.audioFormat, + voice: this.voice, + instructions: this.instructions || 'You are a helpful voice assistant. Be concise.', + turn_detection: { + type: this.options.vadType ?? OpenAIRealtimeVADType.SERVER_VAD, + threshold: 0.5, + prefix_padding_ms: 300, + silence_duration_ms: this.options.silenceDurationMs ?? 300, + }, + input_audio_transcription: { + model: this.options.inputAudioTranscriptionModel ?? OpenAITranscriptionModel.WHISPER_1, + }, + }; + if (this.options.temperature !== undefined) config.temperature = this.options.temperature; + if (this.options.maxResponseOutputTokens !== undefined) { + config.max_response_output_tokens = this.options.maxResponseOutputTokens; + } + if (this.options.modalities !== undefined) config.modalities = this.options.modalities; + if (this.options.toolChoice !== undefined) config.tool_choice = this.options.toolChoice; + if (this.options.reasoningEffort !== undefined) { + config.reasoning = { effort: this.options.reasoningEffort }; + } + if (this.tools?.length) { + config.tools = this.tools.map((t) => { + const def: Record = { + type: 'function', + name: t.name, + description: t.description, + parameters: t.parameters, + }; + // Propagate strict mode when the user opted in. OpenAI's strict + // mode constrains the model to emit arguments that exactly match + // the schema (no missing required fields, no extra properties). + if ((t as { strict?: boolean }).strict === true) { + def.strict = true; + } + return def; + }); + } + return config; + } + + /** + * Pre-call WebSocket warmup for the OpenAI Realtime endpoint. + * + * The canonical session-only warm step on the Realtime API: open the + * WS, wait for `session.created`, send a single `session.update` + * containing the same fields that the production `connect()` path + * applies (`input_audio_format`, `output_audio_format`, `voice`, + * `instructions`, `turn_detection`, `input_audio_transcription`, + * plus any opt-in fields populated on the adapter), wait for the + * matching `session.updated` ack, then close cleanly. This primes + * the per-session state on the OpenAI side — DNS + TLS + auth + * handshake + initial config exchange — without ever invoking the + * model. + * + * Earlier revisions sent `response.create` with + * `{"response": {"generate": false}}` to prime the inference path. + * That field is NOT in the OpenAI Realtime API schema; the server + * either ignores it (and bills tokens for a real model response) or + * rejects the request with `invalid_request_error`. Both behaviours + * are billing-unsafe or a no-op beyond TLS warm. The + * `session.update` flow is documented and side-effect-free. + * + * Billing safety: `session.update` only mutates session + * configuration. It does NOT invoke the model, does NOT consume any + * audio buffer, and does NOT trigger token generation, so no + * per-token cost is accrued. Best-effort: failures are logged at + * debug level and never raised. + */ + async warmup(): Promise { + const url = `wss://api.openai.com/v1/realtime?model=${encodeURIComponent(this.model)}`; + let ws: WebSocket | null = null; + try { + ws = await new Promise((resolve, reject) => { + const sock = new WebSocket(url, { + headers: { + Authorization: `Bearer ${this.apiKey}`, + 'OpenAI-Beta': 'realtime=v1', + }, + }); + const timer = setTimeout(() => { + try { + sock.close(); + } catch { + // ignore + } + reject(new Error('OpenAI Realtime warmup connect timeout')); + }, 5000); + sock.once('open', () => { + clearTimeout(timer); + resolve(sock); + }); + sock.once('error', (err: Error) => { + clearTimeout(timer); + reject(err); + }); + }); + + // Wait for session.created (up to 2 s). + const sessionCreated = await new Promise((resolve) => { + const timer = setTimeout(() => resolve(false), 2000); + const onMsg = (raw: Buffer | string): void => { + try { + const data = JSON.parse(raw.toString()) as { type?: string }; + if (data.type === 'session.created') { + clearTimeout(timer); + ws!.off('message', onMsg); + resolve(true); + } + } catch { + // ignore parse errors + } + }; + ws!.on('message', onMsg); + }); + if (!sessionCreated) return; + + // Send session.update with the same fields the production + // ``connect()`` path applies, so the upstream session state is + // primed identically to a real call. + try { + ws.send(JSON.stringify({ type: 'session.update', session: this.buildSessionConfig() })); + } catch { + return; + } + + // Best-effort: drain frames until we see ``session.updated`` (or + // time out). Waiting for the ack lets us close after a clean + // handshake instead of mid-frame; the TLS + session prime is + // already done by the time the server processes our update. + await new Promise((resolve) => { + const timer = setTimeout(() => resolve(), 1500); + const onMsg = (raw: Buffer | string): void => { + try { + const data = JSON.parse(raw.toString()) as { type?: string }; + if (data.type === 'session.updated') { + clearTimeout(timer); + ws!.off('message', onMsg); + resolve(); + } + } catch { + // ignore + } + }; + ws!.on('message', onMsg); + }); + } catch (err) { + getLogger().debug(`OpenAI Realtime warmup failed (best-effort): ${String(err)}`); + } finally { + if (ws) { + try { + ws.close(); + } catch { + // ignore + } + } + } + } + /** Open the Realtime WebSocket and apply the session configuration. */ async connect(): Promise { const url = `wss://api.openai.com/v1/realtime?model=${encodeURIComponent(this.model)}`; @@ -182,48 +354,7 @@ export class OpenAIRealtimeAdapter { } if (msg.type === 'session.created' && !sessionCreated) { sessionCreated = true; - const config: Record = { - input_audio_format: this.audioFormat, - output_audio_format: this.audioFormat, - voice: this.voice, - instructions: this.instructions || 'You are a helpful voice assistant. Be concise.', - turn_detection: { - type: this.options.vadType ?? OpenAIRealtimeVADType.SERVER_VAD, - threshold: 0.5, - prefix_padding_ms: 300, - silence_duration_ms: this.options.silenceDurationMs ?? 300, - }, - input_audio_transcription: { - model: this.options.inputAudioTranscriptionModel ?? OpenAITranscriptionModel.WHISPER_1, - }, - }; - if (this.options.temperature !== undefined) config.temperature = this.options.temperature; - if (this.options.maxResponseOutputTokens !== undefined) { - config.max_response_output_tokens = this.options.maxResponseOutputTokens; - } - if (this.options.modalities !== undefined) config.modalities = this.options.modalities; - if (this.options.toolChoice !== undefined) config.tool_choice = this.options.toolChoice; - if (this.options.reasoningEffort !== undefined) { - config.reasoning = { effort: this.options.reasoningEffort }; - } - if (this.tools?.length) { - config.tools = this.tools.map(t => { - const def: Record = { - type: 'function', - name: t.name, - description: t.description, - parameters: t.parameters, - }; - // Propagate strict mode when the user opted in. OpenAI's strict - // mode constrains the model to emit arguments that exactly match - // the schema (no missing required fields, no extra properties). - if ((t as { strict?: boolean }).strict === true) { - def.strict = true; - } - return def; - }); - } - ws.send(JSON.stringify({ type: 'session.update', session: config })); + ws.send(JSON.stringify({ type: 'session.update', session: this.buildSessionConfig() })); } else if (msg.type === 'session.updated') { cleanup(); resolve(); @@ -254,6 +385,25 @@ export class OpenAIRealtimeAdapter { ws.on('error', onSetupError); }); + this.armHeartbeatAndListener(); + } + + /** + * Adopt a pre-opened, already-`session.updated` Realtime WebSocket + * produced by the prewarm pipeline (see `Patter.parkProviderConnections`). + * Skips the fresh `new WebSocket()` + `session.created` / + * `session.update` round-trip — saves ~250-450 ms on first turn. + * + * Caller MUST verify `ws.readyState === OPEN` before calling and MUST + * have already received `session.updated` on the parked socket. If + * the parked WS died between park and adopt, fall back to `connect()`. + */ + adoptWebSocket(ws: WebSocket): void { + this.ws = ws; + this.armHeartbeatAndListener(); + } + + protected armHeartbeatAndListener(): void { // Keep WS alive across long silent stretches. ws's server-side `pong` // handler satisfies this automatically; we just need to ping. this.heartbeat = setInterval(() => { @@ -267,6 +417,73 @@ export class OpenAIRealtimeAdapter { this.ensureMessageListener(); } + /** + * Open a fresh Realtime WS, exchange `session.created` / + * `session.update` / `session.updated` (so the upstream session is + * fully primed), and return the OPEN socket WITHOUT arming the + * heartbeat / message listener. Used by the prewarm pipeline to park + * a Realtime connection during ringing; the live consumer adopts it + * via {@link adoptWebSocket}. + * + * Bounded by 8 s. Throws on timeout / handshake failure — callers + * (the prewarm pipeline) treat any error as a cache miss and the + * call falls through to the cold `connect()` path. + * + * Billing safety: `session.update` does not invoke the model. No + * tokens are billed. + */ + async openParkedConnection(): Promise { + const url = `wss://api.openai.com/v1/realtime?model=${encodeURIComponent(this.model)}`; + const ws = new WebSocket(url, { + headers: { + Authorization: `Bearer ${this.apiKey}`, + 'OpenAI-Beta': 'realtime=v1', + }, + }); + await new Promise((resolve, reject) => { + let sessionCreated = false; + let settled = false; + const onMessage = (raw: Buffer | string): void => { + let msg: { type?: string }; + try { + msg = JSON.parse(raw.toString()) as { type?: string }; + } catch { + return; + } + if (msg.type === 'session.created' && !sessionCreated) { + sessionCreated = true; + try { + ws.send(JSON.stringify({ type: 'session.update', session: this.buildSessionConfig() })); + } catch (err) { + cleanup(); + reject(err instanceof Error ? err : new Error(String(err))); + } + } else if (msg.type === 'session.updated') { + cleanup(); + resolve(); + } + }; + const onError = (err: Error): void => { + cleanup(); + reject(err); + }; + const cleanup = (): void => { + if (settled) return; + settled = true; + clearTimeout(timer); + ws.off('message', onMessage); + ws.off('error', onError); + }; + const timer = setTimeout(() => { + cleanup(); + reject(new Error('OpenAI Realtime park connect timeout')); + }, 8000); + ws.on('message', onMessage); + ws.on('error', onError); + }); + return ws; + } + /** Append a base64-encoded audio chunk to the realtime input buffer. */ sendAudio(mulawAudio: Buffer): void { if (!this.ws || this.ws.readyState !== WebSocket.OPEN) return; @@ -291,7 +508,7 @@ export class OpenAIRealtimeAdapter { this.eventCallbacks.delete(callback); } - private ensureMessageListener(): void { + protected ensureMessageListener(): void { if (this.messageListenerAttached || !this.ws) return; this.messageListenerAttached = true; const ws = this.ws; diff --git a/libraries/typescript/src/providers/openai-transcribe-stt.ts b/libraries/typescript/src/providers/openai-transcribe-stt.ts index ece3bba4..e433263c 100644 --- a/libraries/typescript/src/providers/openai-transcribe-stt.ts +++ b/libraries/typescript/src/providers/openai-transcribe-stt.ts @@ -22,6 +22,8 @@ const DEFAULT_BUFFER_SIZE = 16000 * 2; /** STT adapter restricted to OpenAI's GPT-4o Transcribe model family. */ export class OpenAITranscribeSTT extends WhisperSTT { + /** Stable pricing/dashboard key — read by stream-handler/metrics. */ + static override readonly providerKey: string = 'openai_transcribe'; /** * @param apiKey OpenAI API key. * @param language ISO-639-1 language code (e.g. ``"en"``, ``"it"``). Optional. diff --git a/libraries/typescript/src/providers/openai-tts.ts b/libraries/typescript/src/providers/openai-tts.ts index c092a983..709f711e 100644 --- a/libraries/typescript/src/providers/openai-tts.ts +++ b/libraries/typescript/src/providers/openai-tts.ts @@ -18,6 +18,8 @@ const LPF_ALPHA_8K = 0.45; /** OpenAI TTS adapter with built-in streaming resample to 16/8 kHz. */ export class OpenAITTS { + /** Stable pricing/dashboard key — read by stream-handler/metrics. */ + static readonly providerKey = 'openai_tts'; constructor( private readonly apiKey: string, private readonly voice: string = 'alloy', diff --git a/libraries/typescript/src/providers/rime-tts.ts b/libraries/typescript/src/providers/rime-tts.ts index d3a2b1cc..34bd058c 100644 --- a/libraries/typescript/src/providers/rime-tts.ts +++ b/libraries/typescript/src/providers/rime-tts.ts @@ -59,6 +59,8 @@ export interface RimeTTSOptions { /** Rime TTS adapter for the `users.rime.ai/v1/rime-tts` HTTP streaming endpoint. */ export class RimeTTS { + /** Stable pricing/dashboard key — read by stream-handler/metrics. */ + static readonly providerKey = 'rime'; private readonly apiKey: string; private readonly model: string; private readonly speaker: string; diff --git a/libraries/typescript/src/providers/silero-vad.ts b/libraries/typescript/src/providers/silero-vad.ts index 7a8eac0a..e4a1261b 100644 --- a/libraries/typescript/src/providers/silero-vad.ts +++ b/libraries/typescript/src/providers/silero-vad.ts @@ -313,6 +313,12 @@ class OnnxModel { const data = out.data as Float32Array; return data[0] ?? 0; } + + /** Reset the RNN hidden state + rolling context to a fresh inference. */ + reset(): void { + this.context = new Float32Array(this.contextSize); + this.rnnState = new Float32Array(2 * 1 * 128); + } } /** @@ -365,7 +371,11 @@ export class SileroVAD implements VADProvider { const model = new OnnxModel(runtime, session, sampleRate); return new SileroVAD(model, { minSpeechDuration: options.minSpeechDuration ?? 0.25, - minSilenceDuration: options.minSilenceDuration ?? 0.1, + // Bumped 0.1 -> 0.4s after round 10f confirmed VAD speech_end fired on + // natural inter-sentence pauses < 250ms, causing double-talk dispatch. + // 400ms is the industry default for telephony and matches the new + // inter_utterance_gap_ms debounce in stream-handler.ts. + minSilenceDuration: options.minSilenceDuration ?? 0.4, prefixPaddingDuration: options.prefixPaddingDuration ?? 0.03, activationThreshold, deactivationThreshold, @@ -386,7 +396,10 @@ export class SileroVAD implements VADProvider { * - `activationThreshold = 0.5` — upstream `threshold` * - `deactivationThreshold = 0.35` — upstream `neg_threshold = threshold - 0.15` * - `minSpeechDuration = 0.25` — upstream `min_speech_duration_ms = 250` - * - `minSilenceDuration = 0.1` — upstream `min_silence_duration_ms = 100` + * - `minSilenceDuration = 0.4` — telephony default (was 0.1, bumped after + * round 10f found speech_end firing on inter-sentence pauses < 250 ms, + * causing double-talk dispatch). 400 ms matches the industry telephony + * default and the inter_utterance_gap_ms debounce in stream-handler.ts. * - `prefixPaddingDuration = 0.03` — upstream `speech_pad_ms = 30` * * Override any field by passing `options`. Deployments that experience @@ -535,4 +548,27 @@ export class SileroVAD implements VADProvider { this.closed = true; // onnxruntime-node sessions are garbage-collected; no explicit release API. } + + /** + * Reset all per-utterance state so the next ``processFrame`` starts from + * a clean SILENCE state. + * + * Called by the stream handler between agent turns to prevent a "stuck + * SPEECH" condition where PSTN echo / loopback kept the detector's + * probability above ``deactivationThreshold`` for the entire agent turn. + * Without this reset the next user utterance would never trigger a + * SILENCE→SPEECH transition and barge-in would feel "one-shot" (works + * once, then never again until the call ends). + * + * Safe to call any time including on a closed instance (no-op). + */ + reset(): void { + if (this.closed) return; + this.pending = new Float32Array(0); + this.pubSpeaking = false; + this.speechThresholdDuration = 0; + this.silenceThresholdDuration = 0; + this.expFilter.reset(); + this.model.reset(); + } } diff --git a/libraries/typescript/src/providers/soniox-stt.ts b/libraries/typescript/src/providers/soniox-stt.ts index 3b3a2eda..48802d38 100644 --- a/libraries/typescript/src/providers/soniox-stt.ts +++ b/libraries/typescript/src/providers/soniox-stt.ts @@ -125,6 +125,8 @@ export interface SonioxSTTOptions { /** Streaming STT adapter for Soniox's real-time WebSocket API. */ export class SonioxSTT { + /** Stable pricing/dashboard key — read by stream-handler/metrics. */ + static readonly providerKey = 'soniox'; private ws: WebSocket | null = null; private callbacks: TranscriptCallback[] = []; private final = new TokenAccumulator(); diff --git a/libraries/typescript/src/providers/speechmatics-stt.ts b/libraries/typescript/src/providers/speechmatics-stt.ts index c478d734..bf2c8979 100644 --- a/libraries/typescript/src/providers/speechmatics-stt.ts +++ b/libraries/typescript/src/providers/speechmatics-stt.ts @@ -148,6 +148,8 @@ interface SpeechmaticsTranscriptMessage { * ``` */ export class SpeechmaticsSTT { + /** Stable pricing/dashboard key — read by stream-handler/metrics. */ + static readonly providerKey = 'speechmatics'; private ws: WebSocket | null = null; private readonly transcriptCallbacks = new Set(); private readonly errorCallbacks = new Set(); diff --git a/libraries/typescript/src/providers/telnyx-stt.ts b/libraries/typescript/src/providers/telnyx-stt.ts index a1d08c9d..c0e72f2a 100644 --- a/libraries/typescript/src/providers/telnyx-stt.ts +++ b/libraries/typescript/src/providers/telnyx-stt.ts @@ -71,6 +71,8 @@ function createStreamingWavHeader(sampleRate: number, numChannels: number): Buff /** Streaming STT adapter for Telnyx's `/v2/speech-to-text` WebSocket. */ export class TelnyxSTT { + /** Stable pricing/dashboard key — read by stream-handler/metrics. */ + static readonly providerKey = 'telnyx_stt'; private ws: WebSocket | null = null; private callbacks: TranscriptCallback[] = []; private headerSent = false; diff --git a/libraries/typescript/src/providers/telnyx-tts.ts b/libraries/typescript/src/providers/telnyx-tts.ts index c95824b5..23866ac0 100644 --- a/libraries/typescript/src/providers/telnyx-tts.ts +++ b/libraries/typescript/src/providers/telnyx-tts.ts @@ -38,6 +38,8 @@ const DEFAULT_VOICE: TelnyxTTSVoice = TelnyxTTSVoice.NATURAL_HD_ASTRA; /** Streaming TTS adapter for Telnyx's `/v2/text-to-speech/speech` WebSocket. */ export class TelnyxTTS { + /** Stable pricing/dashboard key — read by stream-handler/metrics. */ + static readonly providerKey = 'telnyx_tts'; constructor( private readonly apiKey: string, private readonly voice: string = DEFAULT_VOICE, diff --git a/libraries/typescript/src/providers/whisper-stt.ts b/libraries/typescript/src/providers/whisper-stt.ts index 25d46ef9..2e72caa9 100644 --- a/libraries/typescript/src/providers/whisper-stt.ts +++ b/libraries/typescript/src/providers/whisper-stt.ts @@ -60,6 +60,8 @@ function wrapPcmInWav(pcm: Buffer, sampleRate: number = 16000, channels: number /** Buffered STT adapter for OpenAI's Whisper transcription HTTP API. */ export class WhisperSTT { + /** Stable pricing/dashboard key — read by stream-handler/metrics. */ + static readonly providerKey: string = 'whisper'; private readonly apiKey: string; private readonly model: string; private readonly language: string | undefined; diff --git a/libraries/typescript/src/server.ts b/libraries/typescript/src/server.ts index 6c8b94c7..2a409323 100644 --- a/libraries/typescript/src/server.ts +++ b/libraries/typescript/src/server.ts @@ -8,6 +8,7 @@ import express from 'express'; import { createServer, Server as HTTPServer } from 'http'; import { WebSocketServer, WebSocket as WSWebSocket } from 'ws'; import { OpenAIRealtimeAdapter } from './providers/openai-realtime'; +import { OpenAIRealtime2Adapter } from './providers/openai-realtime-2'; import { ElevenLabsConvAIAdapter } from './providers/elevenlabs-convai'; import { createSTT } from './provider-factory'; import type { STTAdapter } from './provider-factory'; @@ -371,13 +372,14 @@ export function buildAIAdapter(config: LocalConfig, agent: AgentOptions, resolve strict: (t as { strict?: boolean }).strict, })) ?? []; const tools = [...agentTools, TRANSFER_CALL_TOOL, END_CALL_TOOL]; - const openaiKey = engine && engine.kind === 'openai_realtime' ? engine.apiKey : (config.openaiKey ?? ''); + const isOpenAIEngine = engine && (engine.kind === 'openai_realtime' || engine.kind === 'openai_realtime_2'); + const openaiKey = isOpenAIEngine ? engine.apiKey : (config.openaiKey ?? ''); // Forward optional engine-level Realtime knobs so the high-level - // ``OpenAIRealtime`` engine wrapper has the same expressivity as the - // underlying ``OpenAIRealtimeAdapter``. Omitting the option keeps the + // ``OpenAIRealtime`` / ``OpenAIRealtime2`` engine wrappers have the same + // expressivity as the underlying adapters. Omitting the option keeps the // adapter's own defaults — backward compat with users on the prior shape. const adapterOptions: import('./providers/openai-realtime').OpenAIRealtimeOptions = {}; - if (engine && engine.kind === 'openai_realtime') { + if (isOpenAIEngine) { if (engine.reasoningEffort !== undefined) { adapterOptions.reasoningEffort = engine.reasoningEffort; } @@ -385,7 +387,13 @@ export function buildAIAdapter(config: LocalConfig, agent: AgentOptions, resolve adapterOptions.inputAudioTranscriptionModel = engine.inputAudioTranscriptionModel; } } - return new OpenAIRealtimeAdapter( + // Dispatch to the GA-API adapter when the caller passed the + // ``OpenAIRealtime2`` engine marker. Falls through to the v1-beta adapter + // for ``OpenAIRealtime`` and the legacy no-engine code path. + const AdapterCtor = engine && engine.kind === 'openai_realtime_2' + ? OpenAIRealtime2Adapter + : OpenAIRealtimeAdapter; + return new AdapterCtor( openaiKey, agent.model, agent.voice, @@ -711,6 +719,38 @@ export class EmbeddedServer { */ public onMachineDetection?: (result: MachineDetectionResult) => void | Promise; + /** + * Pre-warm first-message audio accessor wired by ``Patter.serve()``. + * The per-call StreamHandler invokes this with its ``callId`` at the + * start of the firstMessage emit; a defined return is sent verbatim + * in place of running TTS again. ``undefined`` means "no prewarm + * cache for this call — fall back to live synthesis". Default is a + * no-op so callers that instantiate ``EmbeddedServer`` directly + * (tests) work without further setup. + */ + public popPrewarmAudio: (callId: string) => Buffer | undefined = () => undefined; + + /** + * Pre-warmed provider WebSocket accessor wired by ``Patter.serve()``. + * The per-call StreamHandler invokes this with its ``callId`` at + * pipeline init; defined returns hand off pre-opened STT / TTS / + * Realtime sockets so the live first turn skips the cold-handshake. + * Default is a no-op for direct ``EmbeddedServer`` callers. + */ + public popPrewarmedConnections: ( + callId: string, + ) => import('./client').ParkedProviderConnections | undefined = () => undefined; + + /** + * Prewarm waste recorder wired by ``Patter.serve()``. Invoked from + * the Twilio status callback (no-answer / busy / failed / canceled) + * and the Telnyx call.hangup / AMD-machine handlers so the cache + * entry is evicted when the call terminates before the media stream + * starts. Default is a no-op so direct ``EmbeddedServer`` callers + * (tests) work without further setup. See FIX #91. + */ + public recordPrewarmWaste: (callId: string) => void = () => undefined; + constructor( private readonly config: LocalConfig, private readonly agent: AgentOptions, @@ -853,6 +893,24 @@ export class EmbeddedServer { if (!Number.isNaN(parsed)) extra.duration_seconds = parsed; this.metricsStore.updateCallStatus(callSid, callStatus, extra); } + // FIX #91 — when the call terminates before the media stream + // starts (no-answer / busy / failed / canceled), the prewarm + // cache entry would otherwise leak until ``endCall`` runs. Evict + // it here so the WARN fires once and the bytes are released + // regardless of whether the user calls ``endCall``. + if ( + callSid && + (callStatus === 'no-answer' || + callStatus === 'busy' || + callStatus === 'failed' || + callStatus === 'canceled') + ) { + try { + this.recordPrewarmWaste(callSid); + } catch (err) { + getLogger().debug(`recordPrewarmWaste threw: ${String(err)}`); + } + } res.status(204).send(); }); @@ -916,6 +974,22 @@ export class EmbeddedServer { } } + // FIX #91 — when AMD classifies as machine, the agent's first + // message will not be played (we drop voicemail or hang up), so + // the prewarmed greeting is never consumed. Evict the cache entry + // once so the WARN fires regardless of whether ``voicemailMessage`` + // is configured. + if ( + (answeredBy === 'machine_end_beep' || answeredBy === 'machine_end_silence') && + callSid + ) { + try { + this.recordPrewarmWaste(callSid); + } catch (err) { + getLogger().debug(`recordPrewarmWaste threw: ${String(err)}`); + } + } + if ( (answeredBy === 'machine_end_beep' || answeredBy === 'machine_end_silence') && this.voicemailMessage && @@ -1013,6 +1087,7 @@ export class EmbeddedServer { to?: string; digit?: string; result?: string; + hangup_cause?: string; recording_urls?: { mp3?: string; wav?: string }; public_recording_urls?: { mp3?: string; wav?: string }; }; @@ -1080,6 +1155,37 @@ export class EmbeddedServer { } if (amdCallId && (amdResult === 'machine' || amdResult === 'machine_detected')) { await this.handleTelnyxAmdVoicemail(amdCallId); + // FIX #91 — when AMD classifies as machine the agent's first + // message is replaced by ``voicemailMessage`` (or the call + // simply ends), so the prewarmed greeting is never consumed. + // Evict it so the WARN fires once. + try { + this.recordPrewarmWaste(amdCallId); + } catch (err) { + getLogger().debug(`recordPrewarmWaste threw: ${String(err)}`); + } + } + return res.status(200).send(); + } + + // FIX #91 — Telnyx fires ``call.hangup`` as the final status + // notification. ``hangup_cause`` distinguishes carrier outcomes + // (``call_rejected`` / ``busy`` / ``no_answer`` / ``timeout`` / + // ``normal_clearing`` / …). When the call never reached the + // media stream the prewarm cache leaks unless we evict it here. + if (eventType === 'call.hangup') { + const hangupCallId = payload.call_control_id ?? ''; + const hangupCause = String(payload.hangup_cause ?? ''); + getLogger().info( + `Telnyx call.hangup for ${sanitizeLogValue(hangupCallId)} ` + + `(cause=${sanitizeLogValue(hangupCause)})`, + ); + if (hangupCallId) { + try { + this.recordPrewarmWaste(hangupCallId); + } catch (err) { + getLogger().debug(`recordPrewarmWaste threw: ${String(err)}`); + } } return res.status(200).send(); } @@ -1327,6 +1433,8 @@ export class EmbeddedServer { buildAIAdapter: (resolvedPrompt: string) => buildAIAdapter(this.config, this.agent, resolvedPrompt), sanitizeVariables, resolveVariables, + popPrewarmAudio: this.popPrewarmAudio, + popPrewarmedConnections: this.popPrewarmedConnections, }; } @@ -1362,14 +1470,28 @@ export class EmbeddedServer { return Object.fromEntries(Object.entries(snap).filter(([, v]) => v !== undefined)); }; + const store = this.metricsStore; const wrappedStart = async (data: Record): Promise => { if (logger.enabled) { const callId = typeof data.call_id === 'string' ? data.call_id : ''; + // For outbound calls the bridge has no caller/callee in the WS query + // string (TwiML for outbound is inline ```` + // with no tags), so ``data.caller`` / ``data.callee`` are + // empty here. The active record in the store was populated by + // ``recordCallInitiated`` at dial time and holds the correct numbers + // — pull them from there before persisting metadata.json. Without + // this fallback every outbound call's metadata.json on disk has + // ``caller=""`` / ``callee=""``. + const dataCaller = typeof data.caller === 'string' ? data.caller : ''; + const dataCallee = typeof data.callee === 'string' ? data.callee : ''; + const active = callId ? store.getActive(callId) : undefined; + const resolvedCaller = dataCaller || active?.caller || ''; + const resolvedCallee = dataCallee || active?.callee || ''; // Fire-and-forget: call logging must never block the voice flow. void logger .logCallStart(callId, { - caller: typeof data.caller === 'string' ? data.caller : '', - callee: typeof data.callee === 'string' ? data.callee : '', + caller: resolvedCaller, + callee: resolvedCallee, telephonyProvider: bridge.telephonyProvider, providerMode: agent.provider ?? '', agent: agentSnapshot(), @@ -1401,16 +1523,24 @@ export class EmbeddedServer { duration_seconds?: number; turns?: unknown[]; cost?: Record; - latency_p50?: { total_ms?: number }; - latency_p95?: { total_ms?: number }; - latency_p99?: { total_ms?: number }; + latency_avg?: Record; + latency_p50?: Record; + latency_p95?: Record; + latency_p99?: Record; }) | null; + // Persist full LatencyBreakdown per percentile so the dashboard + // hydrate path can render stt/llm/tts breakdown for historical + // calls. Keep flat ``p50_ms/p95_ms/p99_ms`` for backward compat. const latency = metricsObj ? { p50_ms: metricsObj.latency_p50?.total_ms ?? null, p95_ms: metricsObj.latency_p95?.total_ms ?? null, p99_ms: metricsObj.latency_p99?.total_ms ?? null, + avg: metricsObj.latency_avg ?? null, + p50: metricsObj.latency_p50 ?? null, + p95: metricsObj.latency_p95 ?? null, + p99: metricsObj.latency_p99 ?? null, } : null; // Fire-and-forget: call logging must never block the voice flow. diff --git a/libraries/typescript/src/services/barge-in-strategies.ts b/libraries/typescript/src/services/barge-in-strategies.ts new file mode 100644 index 00000000..d3b95722 --- /dev/null +++ b/libraries/typescript/src/services/barge-in-strategies.ts @@ -0,0 +1,144 @@ +/** + * Barge-in confirmation strategies. + * + * When a caller starts speaking while the agent's TTS is in flight, the SDK + * has to decide whether the speech is a real interruption or just a brief + * backchannel ("uh-huh", "okay") / room noise / cough. The default + * behaviour is to treat any VAD speech_start as a confirmed barge-in and + * cancel the agent immediately. That is fine for clean inputs but + * produces frequent false positives on PSTN: the agent gets cut + * mid-sentence by background chatter, breath, or filler words and never + * recovers the conversational thread. + * + * Each ``BargeInStrategy`` is consulted on every STT transcript while a + * barge-in is *pending* (VAD fired, but the agent has not yet been + * cancelled). The first strategy that returns ``true`` confirms the + * barge-in; if none do within the configured timeout the pending state + * is dropped and the agent resumes streaming TTS as if nothing happened. + * With an empty ``bargeInStrategies`` array the SDK falls back to the + * legacy "interrupt immediately on VAD" path, so adding strategies is + * a strict opt-in. + */ + +import { getLogger } from '../logger.js'; + +export interface EvaluateContext { + /** Latest STT output text (interim or final). */ + readonly transcript: string; + /** ``true`` for interim partials, ``false`` for finals. */ + readonly isInterim: boolean; + /** Whether the agent's TTS is currently in flight. */ + readonly agentSpeaking: boolean; +} + +/** + * Decides whether a pending barge-in should be confirmed. + * + * Implementations must be safe to call from any number of evaluations + * per turn. ``reset`` is invoked when the agent finishes speaking + * naturally and when a pending barge-in times out without + * confirmation. + */ +export interface BargeInStrategy { + evaluate(ctx: EvaluateContext): Promise | boolean; + reset?(): Promise | void; +} + +export interface MinWordsStrategyOptions { + /** + * Minimum word count required while the agent is speaking. Reasonable + * values are 2-5; 3 is a good starting point for production phone + * agents. Must be ``>= 1``. + */ + readonly minWords: number; + /** + * When ``true`` (default), interim STT partials are evaluated as soon + * as they arrive. Set to ``false`` to wait for finals only — slower + * but free of partial-word noise on jittery STT providers. + */ + readonly useInterim?: boolean; +} + +/** + * Confirm barge-in only after the caller has spoken ``minWords`` words. + * + * Filters short backchannels, single-word utterances, and stray + * transcription fragments that VAD picked up but were not real + * interruptions. While the agent is silent the strategy permits any + * speech to count (one word is enough), so the first user turn is not + * delayed. + */ +export class MinWordsStrategy implements BargeInStrategy { + private readonly minWords: number; + private readonly useInterim: boolean; + + constructor(options: MinWordsStrategyOptions) { + if (!Number.isFinite(options.minWords) || options.minWords < 1) { + throw new Error( + `minWords must be >= 1 (got ${String(options.minWords)})`, + ); + } + this.minWords = Math.floor(options.minWords); + this.useInterim = options.useInterim ?? true; + } + + evaluate(ctx: EvaluateContext): boolean { + if (ctx.isInterim && !this.useInterim) { + return false; + } + const threshold = ctx.agentSpeaking ? this.minWords : 1; + const wordCount = (ctx.transcript ?? '').trim().split(/\s+/).filter(Boolean).length; + return wordCount >= threshold; + } + + async reset(): Promise { + /* stateless */ + } +} + +/** + * Short-circuit-OR composition: first strategy that confirms wins. + * Returns ``false`` for an empty array so callers can use the empty + * default to mean "no opt-in confirmation, fall back to legacy + * interrupt-on-VAD". + */ +export async function evaluateStrategies( + strategies: readonly BargeInStrategy[], + ctx: EvaluateContext, +): Promise { + if (!strategies || strategies.length === 0) { + return false; + } + const safeCtx: EvaluateContext = { + transcript: ctx.transcript ?? '', + isInterim: ctx.isInterim, + agentSpeaking: ctx.agentSpeaking, + }; + for (const strategy of strategies) { + try { + const result = await strategy.evaluate(safeCtx); + if (result === true) return true; + } catch (err) { + getLogger().warn( + `BargeInStrategy ${strategy.constructor?.name ?? 'unknown'} threw; treating as 'do not confirm': ${String(err)}`, + ); + } + } + return false; +} + +/** Call ``reset()`` on every strategy, swallowing per-strategy errors. */ +export async function resetStrategies( + strategies: readonly BargeInStrategy[], +): Promise { + for (const strategy of strategies) { + if (typeof strategy.reset !== 'function') continue; + try { + await strategy.reset(); + } catch (err) { + getLogger().debug( + `BargeInStrategy ${strategy.constructor?.name ?? 'unknown'}.reset() threw: ${String(err)}`, + ); + } + } +} diff --git a/libraries/typescript/src/stream-handler.ts b/libraries/typescript/src/stream-handler.ts index d6fc4cfe..3c68078d 100644 --- a/libraries/typescript/src/stream-handler.ts +++ b/libraries/typescript/src/stream-handler.ts @@ -166,6 +166,24 @@ export interface StreamHandlerDeps { readonly sanitizeVariables: (raw: Record) => Record; /** Replace {key} placeholders in a template string. */ readonly resolveVariables: (template: string, variables: Record) => string; + /** + * Optional accessor returning pre-rendered first-message audio for + * ``callId``. Wired by ``Patter.serve()`` when the parent client has + * ``agent.prewarmFirstMessage: true``. Returning ``undefined`` means + * "no prewarm — always run live TTS". + */ + readonly popPrewarmAudio?: (callId: string) => Buffer | undefined; + /** + * Optional accessor returning pre-opened, fully-handshaked provider + * WebSockets for ``callId`` so the per-call StreamHandler can + * adopt them at ``start`` instead of paying the cold handshake on + * the first turn. Wired by ``Patter.serve()``. Returning + * ``undefined`` (or any sub-field unset) means "no parked socket + * for this provider — fall back to fresh ``connect()``". + */ + readonly popPrewarmedConnections?: ( + callId: string, + ) => import('./client').ParkedProviderConnections | undefined; } // --------------------------------------------------------------------------- @@ -250,6 +268,43 @@ export class StreamHandler { * sentence. */ private speakingStartedAt: number | null = null; + /** + * Wall-clock (ms) when the FIRST TTS audio chunk actually reached the + * carrier wire — set in ``markFirstAudioSent`` after ``bridge.sendAudio`` + * succeeds, cleared by ``beginSpeaking`` / ``cancelSpeaking``. The barge-in + * gate measures elapsed from this instant, NOT from ``speakingStartedAt``, + * because ElevenLabs (and other cloud TTS) take 200-700 ms to emit the + * first byte. A gate anchored to ``beginSpeaking`` would expire on + * background noise before any audio went out, exit the TTS loop on + * ``isSpeaking=false``, and silently cut the agent's first turn. + */ + private firstAudioSentAt: number | null = null; + /** + * Optional barge-in confirmation strategies. With an empty array the + * SDK falls back to the legacy "cancel on first VAD speech_start" + * behaviour. With one or more strategies, a VAD speech_start during + * TTS marks the barge-in as *pending* — TTS keeps streaming naturally + * — and the strategies are consulted on every STT transcript via + * ``handleBargeIn``. The first strategy that returns ``true`` cancels + * the agent; if none confirm within ``bargeInConfirmMs`` the pending + * state is dropped and the agent finishes its sentence. + */ + private readonly bargeInStrategies: readonly import('./services/barge-in-strategies').BargeInStrategy[]; + /** Pending-barge-in confirmation timeout in milliseconds. */ + private readonly bargeInConfirmMs: number; + /** Wall-clock (ms) when the current pending barge-in started, or + * ``null`` if no barge-in is pending. */ + private bargeInPendingSince: number | null = null; + /** Timer that fires the pending-barge-in timeout. */ + private bargeInPendingTimer: ReturnType | null = null; + /** + * Set to true when a VAD ``speech_start`` was suppressed by the + * anti-echo gate during the current agent turn. Cleared on + * ``beginSpeaking`` and ``cancelSpeaking``. When the turn ends + * naturally (grace timer), the inbound audio ring is flushed to STT + * so the user's speech is not silently discarded. + */ + private suppressedSpeechPending = false; /** * Minimum wall-clock duration (ms) the agent must have been speaking * before barge-in is allowed to fire when AEC is active. Covers the @@ -261,10 +316,13 @@ export class StreamHandler { * Same as the AEC variant but for deployments where AEC is OFF * (default on PSTN — Twilio/Telnyx). Without an adaptive filter to * converge, the only justification for a gate is anti-flicker on - * micro-events (cough, click). A short 250 ms window keeps real-user - * barge-in responsive while still filtering tiny noise spikes. + * micro-events (cough, click). 100 ms covers the first PSTN echo + * round-trip (~40-100 ms) while allowing barge-in from 100 ms into + * the agent's turn — covering nearly all of any response. + * Previously 250 ms, which blocked barge-in entirely on short (<500 ms) + * agent responses. */ - private static readonly MIN_AGENT_SPEAKING_MS_BEFORE_BARGE_IN_NO_AEC = 250; + private static readonly MIN_AGENT_SPEAKING_MS_BEFORE_BARGE_IN_NO_AEC = 100; /** Handle for the pending grace-period timer, so it can be cleared on cleanup. */ private graceTimer: ReturnType | null = null; /** @@ -285,6 +343,54 @@ export class StreamHandler { * the tail of the cancelled turn (~50-200 ms of doubled audio). */ private lastCancelAt: number | null = null; + /** + * Promise queue tracking outstanding Twilio marks the SDK has sent but + * not yet seen echoed back. Used by the firstMessage send loop to bound + * the depth of audio queued at the carrier — without this the loop + * pushes the entire TTS stream into Twilio's WebSocket in one burst, + * and a sendClear issued mid-buffer races against several seconds of + * already-queued media frames (BUG #128). The window depth is + * ``FIRST_MESSAGE_MARK_WINDOW``; ``onMark`` drains entries as Twilio + * confirms playback, ``cancelSpeaking`` resolves every pending entry so + * any awaiter exits immediately. Telnyx never populates this queue + * (Telnyx's media-stream protocol has no mark concept — the loop + * falls back to time-based pacing on that carrier). + */ + private pendingMarks: Array<{ + name: string; + resolve: () => void; + promise: Promise; + }> = []; + /** + * Monotonic counter for first-message mark names. Distinct from + * ``chunkCount`` (which the Realtime path uses) so the two paths can + * coexist without name collisions even when firstMessage finishes while + * a Realtime turn is still streaming. + */ + private firstMessageMarkCounter = 0; + /** + * Maximum unconfirmed Twilio marks while streaming firstMessage. Each + * chunk is 40 ms of audio at 16 kHz PCM16, so a window of 3 caps + * the in-flight queue at ~120 ms. This means a barge-in's + * ``sendClear`` has at most 120 ms of already-buffered audio to flush + * — vs. ~2-5 s with the previous burst-send code, which was the + * root cause of "firstMessage non interrompibile". Higher values + * smooth playback under jittery RTT (each mark echo adds ~150-250 ms + * RTT on PSTN) at the cost of longer barge-in latency; lower values + * risk under-buffering. 3 hit the smallest barge-in cap without + * audible gaps in 2026-05 acceptance. + */ + private static readonly FIRST_MESSAGE_MARK_WINDOW = 3; + /** + * Per-chunk soft timeout (ms) while awaiting a mark echo. Twilio's + * mark echoes typically arrive within 100-250 ms of audio playback. + * Capping at 500 ms guards against carriers (or test doubles) that + * never echo — without it a stalled echo would deadlock the loop and + * the agent would freeze mid-utterance. On timeout we drop the + * waiter from the queue and continue: playout may glitch by one + * chunk but the call stays alive. + */ + private static readonly MARK_AWAIT_TIMEOUT_MS = 500; /** * Minimum drain window (ms) between a ``cancelSpeaking`` and the next * ``beginSpeaking``. 150 ms covers a typical PSTN jitter buffer drain @@ -300,7 +406,7 @@ export class StreamHandler { * directly. Awaits the post-cancel drain window before flipping state * so the remote player has time to flush the cancelled turn's tail. */ - private async beginSpeaking(): Promise { + private async beginSpeaking(isFirstMessage = false): Promise { if (this.lastCancelAt !== null) { const elapsed = Date.now() - this.lastCancelAt; const remaining = StreamHandler.POST_CANCEL_DRAIN_MS - elapsed; @@ -311,9 +417,48 @@ export class StreamHandler { this.speakingGeneration++; this.isSpeaking = true; this.speakingStartedAt = Date.now(); + this.suppressedSpeechPending = false; + // Stamp ``firstAudioSentAt`` synchronously for EVERY turn so the + // ``canBargeIn()`` gate (250ms anti-flicker for PSTN no-AEC) runs in + // PARALLEL with LLM TTFT + TTS TTFB rather than starting only after + // the first audio chunk reaches the wire. Without this, a turn with + // a slow LLM (gpt-4o cold cache ~2 s) is effectively un-interruptible + // for the entire LLM window: ``firstAudioSentAt`` stays null, so + // ``canBargeIn`` returns false and every VAD ``speech_start`` is + // suppressed silently. Previously this fix was firstMessage-only; + // promoted to default on 2026-05-11 after the user reported + // "barge-in non funziona più" with gpt-4o. + // + // Note: the ``isFirstMessage`` parameter is kept for backward + // compatibility with the call site, but no longer changes behaviour. + void isFirstMessage; + this.firstAudioSentAt = Date.now(); // Fresh turn — drop any stale pre-barge-in buffer from a previous turn // so we never replay yesterday's audio to STT. this.inboundAudioRing = []; + // Reset the VAD detector so the next user utterance triggers a clean + // SILENCE→SPEECH transition. Without this, PSTN echo from the previous + // turn can keep the detector's smoothed probability above the + // deactivation threshold (0.35) for the entire turn — the VAD never + // returns to SILENCE, ``speech_start`` never fires for the user's next + // utterance, and barge-in feels "one-shot" (works once, then never + // again). The user's previous utterance was already committed by STT + // before ``beginSpeaking`` is called, so resetting state here cannot + // lose data. + this.resetVad(); + } + + /** + * Record that the first TTS audio chunk of the current turn has hit the + * carrier wire. Idempotent within a turn — only the first call sets the + * timestamp; later chunks are no-ops. Must be invoked AFTER the underlying + * ``bridge.sendAudio`` resolves so the gate is anchored to "audio actually + * went out", not "we asked the carrier to send it". + */ + private markFirstAudioSent(): void { + if (this.firstAudioSentAt === null) { + this.firstAudioSentAt = Date.now(); + } } /** @@ -327,7 +472,17 @@ export class StreamHandler { this.speakingGeneration++; // invalidates pending grace timers this.isSpeaking = false; this.speakingStartedAt = null; + this.firstAudioSentAt = null; this.lastCancelAt = Date.now(); + this.suppressedSpeechPending = false; + // Drain any firstMessage mark waiters so a loop blocked on + // ``waitForMarkWindow`` exits on the next tick and observes + // ``!isSpeaking``. Without this the loop would stay blocked until + // each mark either echoes (carrier still draining its queue) or + // hits ``MARK_AWAIT_TIMEOUT_MS`` — keeping the agent "speaking" + // from the user's perspective for hundreds of extra ms after + // barge-in. + this.drainPendingMarks(); if (this.llmAbort !== null) { try { this.llmAbort.abort(); @@ -337,6 +492,86 @@ export class StreamHandler { } } + /** + * Resolve every entry in ``pendingMarks`` and empty the queue. Idempotent + * — safe to call from ``cancelSpeaking`` and again from the grace path + * without leaking pending promises. + */ + private drainPendingMarks(): void { + if (this.pendingMarks.length === 0) return; + for (const entry of this.pendingMarks) { + try { + entry.resolve(); + } catch { + // No-op — pending entries always own a fresh resolve fn. + } + } + this.pendingMarks.length = 0; + } + + /** + * Push a Twilio ``mark`` event AFTER the corresponding audio chunk and + * return a promise that resolves when the mark is echoed back via + * ``onMark`` (or when ``cancelSpeaking`` drains the queue, or after + * ``MARK_AWAIT_TIMEOUT_MS``). Returns null on non-Twilio carriers — the + * caller is expected to fall back to time-based pacing in that case. + */ + private sendMarkAwaitable(): Promise | null { + if (this.deps.bridge.telephonyProvider !== 'twilio') return null; + this.firstMessageMarkCounter += 1; + const markName = `fm_${this.firstMessageMarkCounter}`; + let resolve!: () => void; + const promise = new Promise((r) => { + resolve = r; + }); + this.pendingMarks.push({ name: markName, resolve, promise }); + try { + this.deps.bridge.sendMark(this.ws, markName, this.streamSid); + } catch (err) { + getLogger().debug(`sendMark failed (${markName}): ${String(err)}`); + // Drop the waiter immediately so the queue doesn't fill with + // never-resolving entries that block the window. + const idx = this.pendingMarks.findIndex((m) => m.name === markName); + if (idx >= 0) this.pendingMarks.splice(idx, 1); + return Promise.resolve(); + } + return promise; + } + + /** + * If the in-flight mark queue is at or above ``FIRST_MESSAGE_MARK_WINDOW`` + * entries, wait for the oldest entry to clear (mark echoed, agent + * cancelled, or per-mark timeout). Repeats until the queue depth is + * within the window — under high RTT the carrier may have several + * marks queued and we want every loop iteration to be naturally back- + * pressured by playback. + */ + private async waitForMarkWindow(): Promise { + while ( + this.isSpeaking && + this.pendingMarks.length >= StreamHandler.FIRST_MESSAGE_MARK_WINDOW + ) { + const oldest = this.pendingMarks[0]; + const timeout = new Promise((resolve) => + setTimeout(resolve, StreamHandler.MARK_AWAIT_TIMEOUT_MS), + ); + await Promise.race([oldest.promise, timeout]); + // Drop the head if it's still the same entry — onMark would + // have already removed it on echo; only a timeout leaves it + // in place. + if (this.pendingMarks[0] === oldest) { + this.pendingMarks.shift(); + } + } + } + + /** + * Bytes-per-millisecond for a 16 kHz PCM16 mono stream. Used by the + * non-Twilio firstMessage pacing path to translate chunk size into a + * playout-duration sleep. 16000 samples/sec × 2 bytes = 32 bytes/ms. + */ + private static readonly PCM16_16K_BYTES_PER_MS = 32; + /** Cancel and clear the pending grace timer, if any. */ private clearGraceTimer(): void { if (this.graceTimer !== null) { @@ -372,11 +607,62 @@ export class StreamHandler { if (this.speakingGeneration === gen) { this.isSpeaking = false; this.speakingStartedAt = null; + this.firstAudioSentAt = null; + this.clearPendingBargeIn(); + void this.resetBargeInStrategies(); + // If VAD detected speech during the agent's turn but it was + // gate-suppressed (agent hadn't been speaking long enough for + // barge-in to fire), flush the ring buffer to STT now so the + // user's words aren't silently lost. + if (this.suppressedSpeechPending) { + this.suppressedSpeechPending = false; + this.flushInboundAudioRing(); + } + // Reset VAD so any stuck SPEECH state from echo / loopback during + // the agent's turn does not block the next user utterance from + // emitting ``speech_start``. + this.resetVad(); } }, grace); } else { this.isSpeaking = false; this.speakingStartedAt = null; + this.firstAudioSentAt = null; + this.clearPendingBargeIn(); + void this.resetBargeInStrategies(); + if (this.suppressedSpeechPending) { + this.suppressedSpeechPending = false; + this.flushInboundAudioRing(); + } + this.resetVad(); + } + } + + private async resetBargeInStrategies(): Promise { + if (this.bargeInStrategies.length === 0) return; + const { resetStrategies } = await import('./services/barge-in-strategies.js'); + await resetStrategies(this.bargeInStrategies); + } + + /** + * Reset the active VAD provider's per-utterance state. No-op when the + * provider does not implement the optional ``reset()`` hook. Safe to call + * from any context — failures are swallowed and the VAD is disabled for + * the rest of the call so a flaky reset can never silently kill barge-in + * for every subsequent turn. + */ + private resetVad(): void { + const activeVad = this.deps.agent.vad ?? this.autoVad; + if (!activeVad || this.vadDisabled) return; + try { + const ret = activeVad.reset?.(); + if (ret instanceof Promise) { + ret.catch((err) => { + getLogger().debug(`VAD reset threw: ${String(err)}`); + }); + } + } catch (err) { + getLogger().debug(`VAD reset threw: ${String(err)}`); } } @@ -387,7 +673,14 @@ export class StreamHandler { */ private canBargeIn(): boolean { if (this.speakingStartedAt === null) return true; - const elapsed = Date.now() - this.speakingStartedAt; + // Anchor the gate on "first audio actually emitted", not on + // ``beginSpeaking`` (which fires before the TTS provider's first-byte + // latency has elapsed). Without this guard, background noise picked up + // by VAD ~250 ms after ``beginSpeaking`` triggers a self-cancel BEFORE + // any TTS chunk has reached the wire — the agent's first turn becomes + // silence even though the SDK believes it spoke. + if (this.firstAudioSentAt === null) return false; + const elapsed = Date.now() - this.firstAudioSentAt; const gate = this.aec ? StreamHandler.MIN_AGENT_SPEAKING_MS_BEFORE_BARGE_IN_AEC : StreamHandler.MIN_AGENT_SPEAKING_MS_BEFORE_BARGE_IN_NO_AEC; @@ -499,6 +792,13 @@ export class StreamHandler { this.caller = caller; this.callee = callee; + this.bargeInStrategies = (deps.agent.bargeInStrategies ?? []).slice(); + const confirmMs = deps.agent.bargeInConfirmMs; + this.bargeInConfirmMs = + typeof confirmMs === 'number' && Number.isFinite(confirmMs) && confirmMs > 0 + ? confirmMs + : 1500; + this.history = createHistoryManager(200); // v0.5.0+: ``agent.stt`` / ``agent.tts`` are always STTAdapter / TTSAdapter @@ -862,15 +1162,35 @@ export class StreamHandler { ); } if (evt?.type === 'speech_start') { - if (this.isSpeaking && !this.canBargeIn()) { + const phantomSuppressed = this.isSpeaking && !this.canBargeIn(); + if (phantomSuppressed) { // Within the per-turn warmup gate. With AEC on this is the // ~1 s filter convergence window; without AEC it is just a - // 250 ms anti-flicker margin. INFO so unexpected + // 100 ms anti-flicker margin. INFO so unexpected // suppressions are visible without enabling debug logs. + // + // CRITICAL: do NOT touch metrics state here. An earlier + // bug (pre-0.6.1) called ``startTurnIfIdle()`` for every + // ``speech_start`` including suppressed phantoms, which + // stamped ``turnStart`` at echo/loopback time. The + // legitimate user-speech ``speech_start`` that followed + // then no-op'd (turn_start was already set), so the + // dashboard reported ``user_speech_duration_ms`` of 5-7 s + // even on short ~1 s utterances. getLogger().info( `[VAD] speech_start suppressed (agent speaking < gate, aec=${this.aec ? 'on' : 'off'})`, ); + // Mark that real user speech was detected but gated out. + // The grace-timer callback will replay the ring buffer to + // STT so the speech isn't silently discarded when the + // agent finishes naturally without a barge-in. + this.suppressedSpeechPending = true; } else if (this.isSpeaking) { + if (this.bargeInStrategies.length > 0) { + this.startPendingBargeIn(); + this.metricsAcc.anchorUserSpeechStart(); + return; + } getLogger().info('[VAD] speech_start during TTS → BARGE-IN'); this.metricsAcc.recordOverlapStart(); this.metricsAcc.recordBargeinDetected(); @@ -902,7 +1222,13 @@ export class StreamHandler { } } } - this.metricsAcc.startTurnIfIdle(); + if (!phantomSuppressed) { + // Industry-standard pattern: every legitimate VAD speech_start re-anchors + // the turn timestamp pre-commit. Repairs stale anchors from + // rejected barge-ins / dropped final transcripts, plus the + // original phantom-during-warmup-gate vulnerability. + this.metricsAcc.anchorUserSpeechStart(); + } } else if (evt?.type === 'speech_end') { this.metricsAcc.recordVadStop(); // The SDK's VAD has detected end-of-speech earlier and more @@ -1022,14 +1348,57 @@ export class StreamHandler { */ /** Handle a Twilio Media Streams `mark` event acknowledging audio playback boundaries. */ async onMark(markName: string): Promise { - if (markName) { - this.lastConfirmedMark = markName; + if (!markName) return; + // Resolve the firstMessage mark waiter (if any) so the send loop + // can advance its sliding window. We resolve the matched entry AND + // every entry before it in the queue — Twilio sometimes batches + // mark echoes, and dropping earlier entries first keeps FIFO order + // even when the higher-numbered echo arrives before a lower- + // numbered one (rare but observed on degraded edges). + const idx = this.pendingMarks.findIndex((m) => m.name === markName); + if (idx < 0) return; + // Only record the echo after we have confirmed it matches a known + // queued mark. Before this gate ``onMark`` clobbered + // ``lastConfirmedMark`` with any mark name — including stale + // echoes that no longer correspond to anything we sent, or marks + // emitted by adapters outside the firstMessage queue — which + // would contaminate any downstream barge-in heuristic gated on + // ``lastConfirmedMark``. The Python parity here is structural: + // ``stream_handler.py``'s ``on_mark`` never touches a handler- + // level field at all (the equivalent state lives on + // ``TwilioAudioSender.last_confirmed_mark``, updated only via + // the carrier's own echo handler). + this.lastConfirmedMark = markName; + const resolved = this.pendingMarks.splice(0, idx + 1); + for (const entry of resolved) { + try { + entry.resolve(); + } catch { + // No-op. + } } } /** Handle call stop / stream end. */ /** Handle a carrier-emitted `stop` event signalling the call has ended. */ async handleStop(): Promise { + // Drop any pending barge-in timer BEFORE we tear down metrics / + // adapters. Without this, a call that ends while a barge-in is + // pending leaves a setTimeout scheduled to fire ``bargeInConfirmMs`` + // later and call ``metricsAcc.recordOverlapEnd`` on a finalised + // metrics object — a slow leak in long-running servers and a race + // producing spurious overlap_end events. Idempotent. + this.clearPendingBargeIn(); + // Resolve every pending firstMessage mark waiter before tearing the + // adapter down. A call that ends mid firstMessage (carrier stop + // arriving before the paced sender finished) would otherwise leak + // unresolved promises owned by the send loop. + this.drainPendingMarks(); + // Reset the firstMessage mark counter so a re-used handler starts + // ``fm_`` numbering at 1 on the next call. See + // ``sendPacedFirstMessageBytes`` for the per-send reset that + // protects the within-call path. + this.firstMessageMarkCounter = 0; this.clearGraceTimer(); this.flushResamplers(); await this.closeSttOnce(); @@ -1040,6 +1409,14 @@ export class StreamHandler { /** Handle WebSocket close event. */ /** Tear down adapter, STT/TTS, and per-call state when the carrier WebSocket closes. */ async handleWsClose(): Promise { + // See handleStop — drop pending barge-in timer before cleanup so a + // dead handler can never fire a stale recordOverlapEnd callback. + this.clearPendingBargeIn(); + // See handleStop — drain pending firstMessage marks so an abnormal + // carrier WS drop during the paced sender cannot leak unresolved + // promises owned by the send loop, and reset the counter. + this.drainPendingMarks(); + this.firstMessageMarkCounter = 0; this.clearGraceTimer(); this.flushResamplers(); // Drain STT first so in-flight transcripts fire before onCallEnd. @@ -1096,6 +1473,96 @@ export class StreamHandler { return combined.subarray(0, alignedLen); } + /** + * 40 ms @ 16 kHz mono PCM16 = 1280 bytes. Sized to mirror the smallest + * live-TTS chunk boundary so cancel granularity (mark/clear bookkeeping) + * is identical regardless of whether the firstMessage came from the + * prewarm cache or a live ``tts.synthesizeStream`` stream. + */ + private static readonly PREWARM_CHUNK_BYTES = 1280; + + /** + * Stream a cached firstMessage buffer in pacing-friendly chunks. + * + * Splits ``prewarmBytes`` into ``PREWARM_CHUNK_BYTES`` slices and + * forwards each through ``deps.bridge.sendAudio`` exactly like the + * live TTS path does — preserving Twilio mark/clear granularity. A + * single multi-second sendAudio call would push the whole intro into + * the carrier in one go and a ``sendClear`` issued mid-buffer would + * have nothing to clear ("agent keeps talking after barge-in" UX bug + * on the very first turn). + * + * Returns ``true`` when at least one chunk hit the wire — the caller + * uses that to decide whether to record TTS-first-byte / turn-complete + * metrics. + */ + private async streamPrewarmBytes(prewarmBytes: Buffer): Promise { + return this.sendPacedFirstMessageBytes(prewarmBytes); + } + + /** + * Iterate ``bytes`` as ``PREWARM_CHUNK_BYTES``-sized PCM16 slices and + * forward each via ``deps.bridge.sendAudio`` with mark-gated pacing + * (Twilio) or playout-time-based pacing (Telnyx). Caps the carrier- + * side buffer at ``FIRST_MESSAGE_MARK_WINDOW`` chunks so a barge-in's + * ``sendClear`` has ~120 ms (Twilio) or zero (Telnyx, immediately + * after the latest sleep) of audio to flush. + * + * Bails immediately when ``isSpeaking`` flips to false — both via the + * loop's pre-iter check and via ``drainPendingMarks`` (called from + * ``cancelSpeaking``) which unblocks any in-flight ``waitForMarkWindow``. + * + * Returns ``true`` when at least one chunk hit the wire — the caller + * uses that to decide whether to record TTS-first-byte / turn-complete + * metrics. See BUG #128 for the regression this fix targets. + */ + private async sendPacedFirstMessageBytes(bytes: Buffer): Promise { + // Reset the per-send mark counter so each invocation produces a + // fresh ``fm_1, fm_2, ...`` sequence. Without this the counter + // grows monotonically across turns on a re-used handler and a + // stale ``fm_N`` echo from an earlier turn could match a mark + // name issued later, corrupting the FIFO matching in ``onMark``. + // The queue is also expected empty here by ``cancelSpeaking`` / + // ``handleStop`` / ``handleWsClose``; drain defensively if not. + if (this.pendingMarks.length > 0) this.drainPendingMarks(); + this.firstMessageMarkCounter = 0; + let firstChunkSent = false; + // Once the mark window is first filled we switch to playout-time pacing + // to prevent batch-ACK bursts. Before that we send in burst so the first + // FIRST_MESSAGE_MARK_WINDOW chunks pre-fill the PSTN jitter buffer. + let initialFillComplete = false; + for (let i = 0; i < bytes.length; i += StreamHandler.PREWARM_CHUNK_BYTES) { + if (!this.isSpeaking) break; // barge-in mid-buffer — stop now + // Back-pressure: if too many marks are unconfirmed, wait. Drains + // immediately on cancelSpeaking. + await this.waitForMarkWindow(); + if (!this.isSpeaking) break; + const chunk = bytes.subarray(i, i + StreamHandler.PREWARM_CHUNK_BYTES); + if (!firstChunkSent) firstChunkSent = true; + if (this.aec) this.aec.pushFarEnd(chunk); + const encoded = this.encodePipelineAudio(chunk); + this.deps.bridge.sendAudio(this.ws, encoded, this.streamSid); + this.markFirstAudioSent(); + const markPromise = this.sendMarkAwaitable(); + if (!initialFillComplete && this.pendingMarks.length >= StreamHandler.FIRST_MESSAGE_MARK_WINDOW) { + initialFillComplete = true; + } + // Telnyx has no mark concept — always pace by playout time. + // Twilio: the first FIRST_MESSAGE_MARK_WINDOW chunks go out in burst + // to pre-fill the PSTN jitter buffer (250–1500 ms), then playout-time + // pacing kicks in (via the sticky initialFillComplete flag) to prevent + // batch-ACK bursts from draining the buffer → crackling. + if (markPromise === null || initialFillComplete) { + const playoutMs = Math.max( + 1, + Math.floor(chunk.length / StreamHandler.PCM16_16K_BYTES_PER_MS), + ); + await new Promise((resolve) => setTimeout(resolve, playoutMs)); + } + } + return firstChunkSent; + } + // --------------------------------------------------------------------------- // Private: Pipeline mode // --------------------------------------------------------------------------- @@ -1162,8 +1629,8 @@ export class StreamHandler { // Acoustic echo cancellation: opt-in. // - // Per the industry consensus (LiveKit, Pipecat, Vapi, Retell, Bland) - // and Twilio's own guidance, time-domain NLMS server-side AEC is the + // Per the industry consensus on PSTN echo cancellation and Twilio's + // own guidance, time-domain NLMS server-side AEC is the // RIGHT tool only when the SDK has near-direct access to the mic and // speaker (browser WebRTC, mobile native). PSTN paths route through // a 250–1500 ms Twilio jitter buffer + carrier loop — far outside @@ -1201,14 +1668,81 @@ export class StreamHandler { } } - try { - if (this.stt) await this.stt.connect(); - getLogger().debug(`Pipeline mode (${label}): STT + TTS connected`); - } catch (e) { - getLogger().error(`Pipeline connect FAILED (${label}):`, e); - try { await this.deps.bridge.endCall(this.callId, this.ws); } catch { /* best effort */ } - return; + // Prewarm-handoff: try to adopt pre-opened provider WebSockets that + // the prewarm pipeline (see ``Patter.parkProviderConnections``) + // parked during the carrier ringing window. When a parked WS is + // still OPEN we skip the cold ``connect()`` and the STT first-turn + // can flow audio without paying the 150-400 ms TLS handshake. + // Failures (cache miss, parked WS died) fall back transparently. + let parked: import('./client').ParkedProviderConnections | undefined; + if (this.deps.popPrewarmedConnections) { + try { + parked = this.deps.popPrewarmedConnections(this.callId); + } catch (err) { + getLogger().debug(`popPrewarmedConnections raised: ${String(err)}`); + } + } + // Adopt the TTS WS first — it's a synchronous handoff (the live + // ``synthesizeStream`` call below picks it up via the adapter's + // single-slot adoption queue). + const parkedTts = parked?.tts; + if (parkedTts && this.tts) { + const ttsAny = this.tts as { adoptWebSocket?: (p: typeof parkedTts) => void }; + if (typeof ttsAny.adoptWebSocket === 'function' && parkedTts.ws.readyState === 1 /* OPEN */) { + try { + ttsAny.adoptWebSocket(parkedTts); + getLogger().info(`[CONNECT] callId=${this.callId} provider=tts source=adopted ms=0`); + } catch (err) { + getLogger().debug(`TTS adoptWebSocket failed: ${String(err)}; falling back`); + try { parkedTts.ws.close(); } catch { /* ignore */ } + } + } else { + try { parkedTts.ws.close(); } catch { /* ignore */ } + } + } + + // Kick off STT connect WITHOUT awaiting yet — we only need STT ready + // to receive incoming user audio, not to send the first agent + // message out. Parallelising STT.connect with the TTS firstMessage + // synth shaves 200-400 ms off the perceived first-turn latency. + let sttConnectPromise: Promise | null = null; + if (this.stt) { + const sttAny = this.stt as { adoptWebSocket?: (ws: import('ws').WebSocket) => void }; + const sttStarted = Date.now(); + if ( + parked?.stt && + typeof sttAny.adoptWebSocket === 'function' && + parked.stt.readyState === 1 /* OPEN */ + ) { + try { + sttAny.adoptWebSocket(parked.stt); + getLogger().info( + `[CONNECT] callId=${this.callId} provider=stt source=adopted ms=${Date.now() - sttStarted}`, + ); + sttConnectPromise = Promise.resolve(); + } catch (err) { + getLogger().debug(`STT adoptWebSocket failed: ${String(err)}; falling back`); + try { parked.stt.close(); } catch { /* ignore */ } + sttConnectPromise = (async () => { + await this.stt!.connect(); + getLogger().info( + `[CONNECT] callId=${this.callId} provider=stt source=fresh ms=${Date.now() - sttStarted}`, + ); + })(); + } + } else { + if (parked?.stt) { + try { parked.stt.close(); } catch { /* ignore */ } + } + sttConnectPromise = (async () => { + await this.stt!.connect(); + getLogger().info( + `[CONNECT] callId=${this.callId} provider=stt source=fresh ms=${Date.now() - sttStarted}`, + ); + })(); + } } + getLogger().debug(`Pipeline mode (${label}): STT connect kicked off`); if (this.deps.agent.firstMessage && !this.deps.onMessage && this.tts) { this.metricsAcc.startTurn(); @@ -1218,24 +1752,55 @@ export class StreamHandler { // and produces garbage transcripts, and the ring buffer for // pre-barge-in audio is never populated. Mirrors the per-turn // behaviour in `runPipelineLlm` / `runRegularLlm`. - await this.beginSpeaking(); + // Pass isFirstMessage=true so the canBargeIn() anti-flicker gate + // starts running NOW — TTFB on the TTS provider often eats 300-800ms, + // and without an early anchor the firstMessage is uninterruptible + // during that window. + await this.beginSpeaking(true); let firstChunkSent = false; this.resetTtsCarry(); + // Check the prewarm cache first. When ``Patter.call`` was made + // with ``agent.prewarmFirstMessage: true`` the firstMessage has + // already been synthesised during the ringing window — we send + // the bytes directly through the carrier-side encoder (which + // handles native-rate → carrier-rate resampling) and skip the + // TTS round-trip entirely. + let prewarmBytes: Buffer | undefined; + if (this.deps.popPrewarmAudio) { + try { + prewarmBytes = this.deps.popPrewarmAudio(this.callId); + } catch (err) { + getLogger().debug(`popPrewarmAudio raised: ${String(err)}`); + } + } try { - for await (const chunk of this.tts.synthesizeStream(this.deps.agent.firstMessage)) { - if (!this.isSpeaking) break; // barge-in or test-hangup - if (!firstChunkSent) { firstChunkSent = true; this.metricsAcc.recordTtsFirstByte(); await this.emitAudioOut(); } - // Far-end tap for the echo canceller — push the exact PCM the - // carrier-side encoder will transmit. Without this the AEC - // adapt loop has no reference signal during the intro, - // resulting in unmitigated bleed-through and a "first turn - // unresponsive" UX where the user's voice is masked by the - // agent's TTS in the inbound channel. - if (this.aec) { - this.aec.pushFarEnd(chunk); + if (prewarmBytes) { + this.metricsAcc.recordTtsFirstByte(); + await this.emitAudioOut(); + firstChunkSent = await this.streamPrewarmBytes(prewarmBytes); + } else { + // Streaming TTS path (no prewarm cache). Uses the same simple + // per-chunk send as synthesizeSentence — ElevenLabs HTTP streams + // at near-real-time speed so the carrier-side buffer stays bounded + // without mark-gated pacing. Routing streaming chunks through + // sendPacedFirstMessageBytes caused crackling: its drain+reset on + // every HTTP chunk destroyed mark back-pressure continuity and the + // per-sub-chunk sleep slowed delivery below Twilio's playout rate, + // producing periodic buffer underruns. The prewarm path (a single + // pre-synthesised buffer) still uses sendPacedFirstMessageBytes + // because that buffer can be several seconds long and needs pacing. + for await (const chunk of this.tts.synthesizeStream(this.deps.agent.firstMessage)) { + if (!this.isSpeaking) break; + if (!firstChunkSent) { + firstChunkSent = true; + this.metricsAcc.recordTtsFirstByte(); + await this.emitAudioOut(); + } + if (this.aec) this.aec.pushFarEnd(chunk); + const encoded = this.encodePipelineAudio(chunk); + this.deps.bridge.sendAudio(this.ws, encoded, this.streamSid); + this.markFirstAudioSent(); } - const encoded = this.encodePipelineAudio(chunk); - this.deps.bridge.sendAudio(this.ws, encoded, this.streamSid); } } catch (e) { getLogger().error(`First message TTS error (${label}):`, e); @@ -1249,6 +1814,15 @@ export class StreamHandler { this.endSpeakingWithGrace(); } if (firstChunkSent) { + // Bill the firstMessage TTS characters — they were synthesised + // at ElevenLabs (or the configured TTS provider) and the + // customer pays for them. The previous flow only called + // ``recordTurnComplete`` here, which finalises the turn but does + // NOT increment the TTS char counter — so a 5-turn call with an + // 82-char greeting was under-billed by ~22% on TTS cost. + // ``recordTtsComplete`` is the canonical accumulator entry + // point for TTS char billing (parity with Python fix). + this.metricsAcc.recordTtsComplete(this.deps.agent.firstMessage); await this.emitTurnMetrics(this.metricsAcc.recordTurnComplete(this.deps.agent.firstMessage)); this.history.push({ role: 'assistant', text: this.deps.agent.firstMessage, timestamp: Date.now() }); } @@ -1294,6 +1868,18 @@ export class StreamHandler { } if (this.stt) { + // Make sure the STT WebSocket is OPEN before we install the + // transcript handler — the parallel kickoff above may still be + // resolving when we get here. Failures abort the call. + if (sttConnectPromise) { + try { + await sttConnectPromise; + } catch (e) { + getLogger().error(`STT connect FAILED (${label}):`, e); + try { await this.deps.bridge.endCall(this.callId, this.ws); } catch { /* best effort */ } + return; + } + } this.stt.onTranscript(async (transcript) => { await this.handleTranscript(transcript); }); @@ -1364,6 +1950,7 @@ export class StreamHandler { } const encoded = this.encodePipelineAudio(processedAudio); this.deps.bridge.sendAudio(this.ws, encoded, this.streamSid); + this.markFirstAudioSent(); } } catch (e) { getLogger().error(`TTS streaming error (${this.deps.bridge.label}):`, e); @@ -1412,7 +1999,16 @@ export class StreamHandler { } if (!transcript.isFinal || !transcript.text) return; - if (!this.commitTranscript(transcript.text)) return; + if (!this.commitTranscript(transcript.text)) { + // Final transcript dropped (dedup / hallucination / back-to-back). + // Any VAD ``speech_end`` that fired during this dropped utterance + // already stamped ``_endpointSignalAt``; if we leave it there, the + // NEXT legitimate utterance inherits the stale anchor (its + // agent_response_ms then includes the silence gap between the + // dropped utterance and the real one). + this.metricsAcc.anchorUserSpeechStart(); + return; + } const label = this.deps.bridge.label; // [DIAG-2026-05-05] Temporary INFO. Remove once root cause known. @@ -1518,6 +2114,11 @@ export class StreamHandler { } else if (this.llmLoop) { responseText = await this.runPipelineLlm(filteredTranscript, hookExecutor, hookCtx); } else { + getLogger().warn( + `Pipeline (${label}) has no llm/onMessage handler — transcript ` + + `"${sanitizeLogValue(filteredTranscript.slice(0, 60))}" dropped. ` + + 'Check that agent.llm or onMessage is configured.', + ); return; } @@ -1548,20 +2149,93 @@ export class StreamHandler { * record the interruption, and return ``true`` so the caller skips the * turn-complete record. */ - private handleBargeIn(transcript: { text?: string }): boolean { + private async handleBargeInAsync(transcript: { + text?: string; + isFinal?: boolean; + }): Promise { if (!transcript.text || !this.isSpeaking) return false; if (!this.canBargeIn()) { - // Same rationale as the VAD-path gate in handleAudio: gate is - // 1 s with AEC (filter warmup) or 250 ms without (anti-flicker). getLogger().info( `Barge-in transcript suppressed (agent speaking < gate, aec=${this.aec ? 'on' : 'off'})`, ); return false; } + if (this.bargeInStrategies.length > 0) { + const { evaluateStrategies } = await import( + './services/barge-in-strategies.js' + ); + const confirmed = await evaluateStrategies(this.bargeInStrategies, { + transcript: transcript.text, + isInterim: transcript.isFinal === false, + agentSpeaking: this.isSpeaking, + }); + if (!confirmed) { + getLogger().debug( + `Barge-in NOT confirmed by any strategy (${sanitizeLogValue( + transcript.text.slice(0, 40), + )}); agent continues talking`, + ); + return false; + } + getLogger().info( + `Barge-in confirmed by strategy on transcript ${sanitizeLogValue( + transcript.text.slice(0, 40), + )}`, + ); + } + this.runBargeInCancel(transcript.text); + return true; + } + + /** + * Synchronous wrapper that callers in legacy code paths can keep using. + * When ``bargeInStrategies`` is empty the work is fully synchronous and + * the result is correct. With strategies the call is dispatched as a + * floating promise — non-confirmed transcripts simply skip the cancel + * and the legacy boolean return is meaningless under that opt-in path. + */ + private handleBargeIn(transcript: { text?: string; isFinal?: boolean }): boolean { + if (!transcript.text || !this.isSpeaking) return false; + if (this.bargeInStrategies.length === 0) { + // Legacy synchronous path — preserve exact byte-for-byte behaviour + // for users who haven't opted into the confirm pipeline. + if (!this.canBargeIn()) { + getLogger().info( + `Barge-in transcript suppressed (agent speaking < gate, aec=${this.aec ? 'on' : 'off'})`, + ); + return false; + } + this.runBargeInCancel(transcript.text); + return true; + } + // Opt-in confirm path is async; fire-and-forget. The cancel inside + // ``runBargeInCancel`` flips ``isSpeaking`` synchronously once it + // resolves, which is what downstream loops actually observe. + void this.handleBargeInAsync(transcript).catch((err) => + getLogger().debug(`handleBargeInAsync threw: ${String(err)}`), + ); + return false; + } + + /** + * Run the cancel/flush sequence for a confirmed barge-in. Shared by + * the legacy synchronous path and the strategy-confirmed async path. + */ + private runBargeInCancel(transcriptText: string): void { + // Capture pending state BEFORE clearPendingBargeIn() drops it — if VAD + // already started the overlap window via ``startPendingBargeIn`` we MUST + // NOT call ``recordOverlapStart`` again (that would overwrite T1 with + // T2 and produce a near-zero ``InterruptionMetrics.detection_delay_ms`` + // on the strategy path). + const hadPending = this.bargeInPendingSince !== null; + this.clearPendingBargeIn(); getLogger().debug( - `Barge-in: caller spoke over agent (${sanitizeLogValue(transcript.text.slice(0, 40))})`, + `Barge-in: caller spoke over agent (${sanitizeLogValue(transcriptText.slice(0, 40))})`, ); - this.metricsAcc.recordOverlapStart(); + if (!hadPending) { + // Legacy path or VAD never fired — start the overlap window now. + this.metricsAcc.recordOverlapStart(); + } this.metricsAcc.recordBargeinDetected(); const bargeinSpan = startSpan(SPAN_BARGEIN, { 'patter.call.id': this.callId }); try { @@ -1573,6 +2247,9 @@ export class StreamHandler { } this.metricsAcc.recordTtsStopped(); this.metricsAcc.recordTurnInterrupted(); + // Re-anchor turn metrics to the legitimate VAD speech_start so post- + // barge-in latency anchors don't carry over from the interrupted turn. + this.metricsAcc.anchorUserSpeechStart(); this.metricsAcc.recordOverlapEnd(true); } finally { try { @@ -1581,7 +2258,37 @@ export class StreamHandler { // Swallow. } } - return true; + } + + /** Mark a VAD-detected barge-in as pending (no cancel yet). */ + private startPendingBargeIn(): void { + if (this.bargeInPendingSince !== null) return; + this.bargeInPendingSince = Date.now(); + this.metricsAcc.recordOverlapStart(); + getLogger().info( + 'Barge-in PENDING (VAD speech_start during TTS); awaiting strategy confirmation', + ); + this.bargeInPendingTimer = setTimeout(() => { + if (this.bargeInPendingSince === null) return; + getLogger().info( + `Pending barge-in timed out after ${this.bargeInConfirmMs}ms; agent resumes (no strategy confirmed)`, + ); + this.metricsAcc.recordOverlapEnd(false); + // Clear any anchors that drifted during the pending barge-in window. + this.metricsAcc.anchorUserSpeechStart(); + this.bargeInPendingSince = null; + this.bargeInPendingTimer = null; + }, this.bargeInConfirmMs); + } + + /** Drop pending state without cancelling — used on confirm and on + * agent stop. Idempotent. */ + private clearPendingBargeIn(): void { + if (this.bargeInPendingTimer !== null) { + clearTimeout(this.bargeInPendingTimer); + this.bargeInPendingTimer = null; + } + this.bargeInPendingSince = null; } /** @@ -1796,6 +2503,7 @@ export class StreamHandler { if (!wsTtsStarted) { wsTtsStarted = true; this.metricsAcc.recordTtsFirstByte(); await this.emitAudioOut(); } const encoded = this.encodePipelineAudio(audioChunk); this.deps.bridge.sendAudio(this.ws, encoded, this.streamSid); + this.markFirstAudioSent(); } } } @@ -1975,6 +2683,7 @@ export class StreamHandler { // reusing it on the outbound path corrupts both directions. const outAudio = eventData; this.deps.bridge.sendAudio(this.ws, outAudio.toString('base64'), this.streamSid); + this.markFirstAudioSent(); // Send mark for barge-in accuracy. this.chunkCount++; this.deps.bridge.sendMark(this.ws, `audio_${this.chunkCount}`, this.streamSid); @@ -2423,11 +3132,17 @@ export class StreamHandler { }; // Single INFO line per call-end — duration, turns, cost, latency. + // "p95 wait" = agent_response_ms (user-perceived wait after they stop + // speaking). Matches the dashboard "p95 wait" tile. Fallback to total_ms + // for legacy/short calls where agent_response_ms is undefined. const cost = (finalMetrics.cost as { total?: number } | undefined)?.total ?? 0; - const latencyP95 = (finalMetrics.latency_p95 as { total_ms?: number } | undefined)?.total_ms ?? 0; + const p95Obj = finalMetrics.latency_p95 as + | { agent_response_ms?: number; total_ms?: number } + | undefined; + const latencyP95 = p95Obj?.agent_response_ms ?? p95Obj?.total_ms ?? 0; getLogger().info( `Call ended: ${this.callId} (${finalMetrics.duration_seconds.toFixed(1)}s, ` + - `${finalMetrics.turns.length} turns, cost=$${cost.toFixed(4)}, p95=${Math.round(latencyP95)}ms)`, + `${finalMetrics.turns.length} turns, cost=$${cost.toFixed(4)}, p95 wait=${Math.round(latencyP95)}ms)`, ); this.deps.metricsStore.recordCallEnd( callEndData, diff --git a/libraries/typescript/src/types.ts b/libraries/typescript/src/types.ts index c648d426..b1561c83 100644 --- a/libraries/typescript/src/types.ts +++ b/libraries/typescript/src/types.ts @@ -6,11 +6,13 @@ import type { Carrier as TwilioCarrier } from "./telephony/twilio"; import type { Carrier as TelnyxCarrier } from "./telephony/telnyx"; import type { Realtime } from "./engines/openai"; +import type { Realtime2 } from "./engines/openai-2"; import type { ConvAI } from "./engines/elevenlabs"; import type { CloudflareTunnel, Static as StaticTunnel } from "./tunnels"; import type { Tool as ToolInstance } from "./public-api"; import type { STTAdapter, TTSAdapter } from "./provider-factory"; import type { LLMProvider } from "./llm-loop"; +import type { BargeInStrategy } from "./services/barge-in-strategies"; /** Inbound message handed to a `MessageHandler` per turn (legacy single-turn API). */ export interface IncomingMessage { @@ -284,6 +286,15 @@ export interface VADEvent { export interface VADProvider { processFrame(pcmChunk: Buffer, sampleRate: number): Promise; close(): Promise; + /** + * Optional: reset all per-utterance state so the next ``processFrame`` + * starts from a clean SILENCE state. Useful between agent turns to + * prevent a "stuck SPEECH" condition where PSTN echo / loopback kept the + * detector's internal probability above the deactivation threshold for + * the full agent turn, leaving the VAD unable to emit ``speech_start`` + * on the next user utterance (one-shot barge-in bug). + */ + reset?(): Promise | void; } /** Pre-STT audio filter — noise cancellation, gain, EQ. */ @@ -382,7 +393,7 @@ export interface AgentOptions { * matching mode (``openai_realtime`` or ``elevenlabs_convai``). When absent, * pipeline mode is selected if ``stt`` and ``tts`` are provided. */ - engine?: Realtime | ConvAI; + engine?: Realtime | Realtime2 | ConvAI; /** * Provider mode. Normally derived from ``engine`` / ``stt`` + ``tts``. Pass * ``'pipeline'`` explicitly when building a pipeline-mode agent without @@ -423,6 +434,59 @@ export interface AgentOptions { * Default: 300. */ bargeInThresholdMs?: number; + /** + * Opt-in barge-in confirmation strategies (pipeline mode). With the + * default empty array the SDK falls back to the legacy + * "interrupt immediately on VAD speech_start" behaviour. When at + * least one strategy is provided, a VAD speech_start during TTS + * marks the barge-in as *pending* — the agent's TTS continues + * streaming naturally and its in-flight LLM stream is preserved — + * and the strategies are consulted on every STT transcript. The first strategy that + * returns ``true`` confirms the barge-in (cancels TTS, flushes the + * inbound ring buffer); if none confirm within + * ``bargeInConfirmMs`` the pending state is dropped and TTS resumes. + * + * See ``getpatter`` exports ``BargeInStrategy`` / + * ``MinWordsStrategy`` for the protocol and a reference + * implementation. + */ + bargeInStrategies?: readonly BargeInStrategy[]; + /** + * Maximum time (ms) to wait for at least one strategy to confirm a + * pending barge-in before discarding the pending state and resuming + * TTS. Only consulted when ``bargeInStrategies`` is non-empty. + * Default: 1500. + */ + bargeInConfirmMs?: number; + /** + * When ``true`` (default), ``Patter.call`` warms up the STT, TTS, and + * LLM provider connections in parallel with the carrier-side + * ``initiateCall`` request so DNS, TLS, and HTTP/2 handshakes are + * already complete by the time the callee answers. Adapters expose a + * ``warmup()`` method returning ``Promise`` (default no-op) — + * providers can override to dial open a persistent connection ahead + * of the WebSocket bridge. Best-effort: warmup failures are logged + * at debug level and never abort the call. Default: ``true``. + */ + prewarm?: boolean; + /** + * When ``true`` (default ``false``), ``Patter.call`` also pre-renders + * ``firstMessage`` to TTS audio bytes during the ringing window and + * streams the cached buffer immediately when the carrier emits + * ``start``. Eliminates the 200-700 ms TTS first-byte latency on the + * greeting at the cost of paying the TTS bill even if the call is + * never answered (silently logged at warn level when the call + * fails). Off by default to preserve the prior cost surface; opt-in + * for production outbound where every millisecond of greeting + * latency hurts conversion. Default: ``false``. + * + * **Pipeline mode only.** Realtime / ConvAI provider modes never + * consume the prewarm cache (the StreamHandler for those modes runs + * its first-message emit through the provider's own audio path), so + * ``Patter.call`` refuses to spawn the prewarm task and emits a warn + * when ``provider !== 'pipeline'``. + */ + prewarmFirstMessage?: boolean; /** * When true, the sentence chunker emits the first clause of each response * on a soft punctuation boundary (",", em-dash, en-dash) once ~40 chars diff --git a/libraries/typescript/tests/dashboard-store.test.ts b/libraries/typescript/tests/dashboard-store.test.ts index 62125da8..5da4e0e9 100644 --- a/libraries/typescript/tests/dashboard-store.test.ts +++ b/libraries/typescript/tests/dashboard-store.test.ts @@ -164,6 +164,125 @@ describe('MetricsStore', () => { const active = store.getActiveCalls(); expect(active[0].turns).toHaveLength(0); }); + + // BUG 1 — live-transcript accumulates user/assistant lines across multiple + // round-trips. Without this behaviour the SPA's mapper had to derive a + // running transcript from ``turns[]`` and the primary mapper path + // (``record.transcript.length > 0``) was empty for live calls — producing + // intermittent renderings where one turn replaced the previous one. + it('recordTurn appends both user and assistant lines to active.transcript across turns', () => { + const store = new MetricsStore(); + store.recordCallStart({ call_id: 'c1', caller: '+1', callee: '+2' }); + + // Turn 0: agent's first message (no user_text yet). + store.recordTurn({ + call_id: 'c1', + turn: { turn_index: 0, user_text: '', agent_text: 'Hello!', timestamp: 1 }, + }); + // Turn 1: user → agent round-trip. + store.recordTurn({ + call_id: 'c1', + turn: { turn_index: 1, user_text: 'Hi there', agent_text: 'How can I help?', timestamp: 2 }, + }); + // Turn 2: another user → agent round-trip. + store.recordTurn({ + call_id: 'c1', + turn: { turn_index: 2, user_text: 'Tell me a joke', agent_text: 'Why did the chicken…', timestamp: 3 }, + }); + + const active = store.getActive('c1'); + expect(active).toBeDefined(); + expect(active!.turns).toHaveLength(3); + // Five entries: bot/Hello + user/Hi+bot/Howcan + user/joke+bot/Why + expect(active!.transcript).toHaveLength(5); + expect(active!.transcript![0]).toEqual({ role: 'assistant', text: 'Hello!', timestamp: 1 }); + expect(active!.transcript![1]).toEqual({ role: 'user', text: 'Hi there', timestamp: 2 }); + expect(active!.transcript![2]).toEqual({ role: 'assistant', text: 'How can I help?', timestamp: 2 }); + expect(active!.transcript![3]).toEqual({ role: 'user', text: 'Tell me a joke', timestamp: 3 }); + expect(active!.transcript![4]).toEqual({ role: 'assistant', text: 'Why did the chicken…', timestamp: 3 }); + }); + + it("recordTurn skips '[interrupted]' agent_text and empty user_text from active.transcript", () => { + const store = new MetricsStore(); + store.recordCallStart({ call_id: 'c1' }); + + // Empty user_text + non-empty agent_text → only assistant line is pushed. + store.recordTurn({ + call_id: 'c1', + turn: { turn_index: 0, user_text: '', agent_text: 'Greeting', timestamp: 1 }, + }); + // Interrupted turn — agent_text === '[interrupted]' is filtered out. + store.recordTurn({ + call_id: 'c1', + turn: { turn_index: 1, user_text: 'wait', agent_text: '[interrupted]', timestamp: 2 }, + }); + + const active = store.getActive('c1'); + expect(active!.transcript).toHaveLength(2); + expect(active!.transcript![0]).toEqual({ role: 'assistant', text: 'Greeting', timestamp: 1 }); + expect(active!.transcript![1]).toEqual({ role: 'user', text: 'wait', timestamp: 2 }); + }); + + // BUG 2 — completed entries preserve transcript and turns from the active + // record so the live-pane race window between updateCallStatus + // ('completed') and recordCallEnd never yields a blank record. + it("updateCallStatus('completed') copies turns and transcript from active record", () => { + const store = new MetricsStore(); + store.recordCallStart({ call_id: 'c1', caller: '+1', callee: '+2' }); + store.recordTurn({ + call_id: 'c1', + turn: { turn_index: 0, user_text: 'Hi', agent_text: 'Hello', timestamp: 5 }, + }); + store.updateCallStatus('c1', 'completed', { duration_seconds: 12 }); + + const completed = store.getCall('c1'); + expect(completed).not.toBeNull(); + expect(completed!.status).toBe('completed'); + // turns + running transcript carried over so the dashboard's live + // pane has data to render in the gap between this event and the + // subsequent recordCallEnd. + expect(completed!.turns).toHaveLength(1); + expect(completed!.transcript).toHaveLength(2); + expect(completed!.transcript![0]).toEqual({ role: 'user', text: 'Hi', timestamp: 5 }); + expect(completed!.transcript![1]).toEqual({ role: 'assistant', text: 'Hello', timestamp: 5 }); + }); + + it("recordCallEnd preserves active turns and falls back to running transcript when data.transcript is empty", () => { + const store = new MetricsStore(); + store.recordCallStart({ call_id: 'c1', caller: '+1', callee: '+2' }); + store.recordTurn({ + call_id: 'c1', + turn: { turn_index: 0, user_text: 'A', agent_text: 'B', timestamp: 1 }, + }); + // Carrier statusCallback fires first, moves to completed without + // populating the transcript field. + store.updateCallStatus('c1', 'completed', {}); + // Then the WS-driven recordCallEnd runs WITHOUT a transcript payload + // (e.g. an external controller calling end_call early). The fallback + // should pull the running transcript / turns from the prior entry. + store.recordCallEnd({ call_id: 'c1' }); + + const completed = store.getCall('c1'); + expect(completed!.transcript).toHaveLength(2); + expect(completed!.turns).toHaveLength(1); + }); + + it('recordCallEnd prefers explicit data.transcript over the running fallback', () => { + const store = new MetricsStore(); + store.recordCallStart({ call_id: 'c1' }); + store.recordTurn({ + call_id: 'c1', + turn: { turn_index: 0, user_text: 'live-A', agent_text: 'live-B', timestamp: 1 }, + }); + const authoritative = [ + { role: 'user', text: 'final-A', timestamp: 10 }, + { role: 'assistant', text: 'final-B', timestamp: 11 }, + ]; + store.recordCallEnd({ call_id: 'c1', transcript: authoritative }); + + const completed = store.getCall('c1'); + expect(completed!.transcript).toEqual(authoritative); + }); }); describe('MetricsStore.hydrate', () => { @@ -326,4 +445,144 @@ describe('MetricsStore.hydrate', () => { fs.rmSync(root, { recursive: true, force: true }); } }); + + it('lifts top-level cost/latency/duration into metrics (CallLogger schema)', () => { + // CallLogger.logCallEnd writes cost/latency/duration_ms/telephony_provider + // at the top of metadata.json — without this fallback hydrated calls show + // $0.00 / "—" in the dashboard because the UI reads from metrics.cost etc. + const root = fs.mkdtempSync(`${os.tmpdir()}/patter-store-test-`); + try { + const callDir = `${root}/calls/2026/05/08/CA-real-shape`; + fs.mkdirSync(callDir, { recursive: true }); + fs.writeFileSync( + `${callDir}/metadata.json`, + JSON.stringify({ + schema_version: '1.0', + call_id: 'CA-real-shape', + started_at: '2026-05-08T23:33:00.000Z', + ended_at: '2026-05-08T23:33:57.000Z', + duration_ms: 57400, + status: 'completed', + telephony_provider: 'twilio', + provider_mode: 'pipeline', + turns: 9, + cost: { + stt: 0.001526, + tts: 0.02988, + llm: 0.000406, + telephony: 0.0085, + total: 0.040312, + }, + latency: { p50_ms: 2127.7, p95_ms: 3461.7, p99_ms: 3640.1 }, + }), + ); + const store = new MetricsStore(); + expect(store.hydrate(root)).toBe(1); + const rec = store.getCalls()[0]; + expect(rec.metrics).not.toBeNull(); + const m = rec.metrics as Record; + expect((m.cost as Record).total).toBeCloseTo(0.040312, 6); + expect((m.latency as Record).p95_ms).toBeCloseTo(3461.7); + expect((m.latency_avg as Record).total_ms).toBeCloseTo(3461.7); + expect(m.duration_seconds).toBeCloseTo(57.4); + expect(m.telephony_provider).toBe('twilio'); + } finally { + fs.rmSync(root, { recursive: true, force: true }); + } + }); + + it('preserves explicit metrics when present (does not overwrite with top-level)', () => { + const root = fs.mkdtempSync(`${os.tmpdir()}/patter-store-test-`); + try { + const callDir = `${root}/calls/2026/05/08/CA-explicit`; + fs.mkdirSync(callDir, { recursive: true }); + fs.writeFileSync( + `${callDir}/metadata.json`, + JSON.stringify({ + call_id: 'CA-explicit', + started_at: '2026-05-08T10:00:00Z', + metrics: { cost: { total: 0.999 }, marker: 'kept' }, + cost: { total: 0.001 }, + latency: { p95_ms: 9999 }, + }), + ); + const store = new MetricsStore(); + expect(store.hydrate(root)).toBe(1); + const m = store.getCalls()[0].metrics as Record; + expect(m.marker).toBe('kept'); + expect((m.cost as Record).total).toBeCloseTo(0.999); + } finally { + fs.rmSync(root, { recursive: true, force: true }); + } + }); +}); + +describe('MetricsStore — recordCallEnd does not duplicate after updateCallStatus', () => { + // Regression for dashboard BUG C: the Twilio statusCallback for + // ``CallStatus=completed`` invokes ``updateCallStatus`` (which moves the + // row from active to completed), and then the WS ``stop`` frame invokes + // ``recordCallEnd`` for the same call_id. Before the fix the second + // call appended a duplicate row with ``started_at=0`` and empty + // caller/callee, which then masked the original entry in ``getCalls`` + // (newest-first ordering, mergeCalls de-dup keeps the first match). + it('updates the existing entry instead of appending a duplicate', () => { + const store = new MetricsStore(); + store.recordCallInitiated({ + call_id: 'CA-dup', + caller: '+15551112222', + callee: '+15553334444', + direction: 'outbound', + }); + store.recordCallStart({ call_id: 'CA-dup' }); + // Twilio statusCallback path moves the call to completed first. + store.updateCallStatus('CA-dup', 'completed', { duration_seconds: 42 }); + expect(store.getActiveCalls()).toHaveLength(0); + expect(store.callCount).toBe(1); + const intermediate = store.getCalls()[0]; + expect(intermediate.caller).toBe('+15551112222'); + expect(intermediate.callee).toBe('+15553334444'); + const startedAtBefore = intermediate.started_at; + expect(startedAtBefore).toBeGreaterThan(0); + + // Then the WS stop handler fires recordCallEnd. ``data.caller`` is + // empty here because outbound TwiML carries no Stream parameters. + store.recordCallEnd( + { call_id: 'CA-dup', caller: '', callee: '', transcript: [] }, + { cost: { total: 0.07 }, duration_seconds: 42 } as Record, + ); + + expect(store.callCount).toBe(1); // no duplicate row + const finalEntry = store.getCalls()[0]; + expect(finalEntry.call_id).toBe('CA-dup'); + expect(finalEntry.caller).toBe('+15551112222'); // preserved + expect(finalEntry.callee).toBe('+15553334444'); + expect(finalEntry.started_at).toBe(startedAtBefore); // preserved (not 0) + expect(finalEntry.metrics).toEqual({ + cost: { total: 0.07 }, + duration_seconds: 42, + }); + expect(finalEntry.status).toBe('completed'); + }); + + it('keeps a call inside the 24h time-range window after end', () => { + // End-to-end check that mirrors the real bug: dashboard-app filters + // calls by [now - 24h, now] using ``startedAtMs``. With the duplicate + // bug the started_at was 0 → call dropped off the 24h slice. + const store = new MetricsStore(); + store.recordCallInitiated({ + call_id: 'CA-window', + caller: '+15551112222', + callee: '+15553334444', + direction: 'outbound', + }); + store.recordCallStart({ call_id: 'CA-window' }); + store.updateCallStatus('CA-window', 'completed', { duration_seconds: 5 }); + store.recordCallEnd( + { call_id: 'CA-window', transcript: [] }, + { duration_seconds: 5 } as Record, + ); + const now = Date.now() / 1000; + const inWindow = store.getCallsInRange(now - 86_400, now + 60); + expect(inWindow.map((c) => c.call_id)).toContain('CA-window'); + }); }); diff --git a/libraries/typescript/tests/pricing.test.ts b/libraries/typescript/tests/pricing.test.ts index ba13e012..28ef3e5b 100644 --- a/libraries/typescript/tests/pricing.test.ts +++ b/libraries/typescript/tests/pricing.test.ts @@ -26,9 +26,9 @@ describe('DEFAULT_PRICING', () => { describe('mergePricing', () => { it('returns defaults when no overrides', () => { const merged = mergePricing(); - // Deepgram Nova-3 streaming (monolingual) — $0.0077/min. Updated from - // the batch rate $0.0043 in 0.5.6 — the default model is streaming. - expect(merged.deepgram.price).toBe(0.0077); + // Deepgram Nova-3 streaming (monolingual) — $0.0048/min (current PAYG + // promo rate per deepgram.com/pricing, verified 2026-05-11). + expect(merged.deepgram.price).toBe(0.0048); }); it('overrides individual provider values', () => { @@ -47,8 +47,9 @@ describe('calculateSttCost', () => { it('calculates deepgram cost for 60 seconds', () => { const pricing = mergePricing(); const cost = calculateSttCost('deepgram', 60, pricing); - // 60s / 60 * $0.0077/min (Nova-3 streaming monolingual) = $0.0077 - expect(cost).toBeCloseTo(0.0077, 4); + // 60s / 60 * $0.0048/min (Nova-3 streaming monolingual, current PAYG + // promo rate per deepgram.com/pricing, verified 2026-05-11) = $0.0048 + expect(cost).toBeCloseTo(0.0048, 4); }); it('returns 0 for unknown provider', () => { @@ -61,9 +62,11 @@ describe('calculateTtsCost', () => { it('calculates elevenlabs cost for 1000 characters', () => { const pricing = mergePricing(); const cost = calculateTtsCost('elevenlabs', 1000, pricing); - // eleven_flash_v2_5 (the default model): $0.06/1k chars via direct API. - // The previous $0.18 matched only the Creator plan overage. - expect(cost).toBeCloseTo(0.06, 3); + // eleven_flash_v2_5 (the default model): $0.05/1k chars per the public + // API/overage rate at https://elevenlabs.io/pricing/api (verified + // 2026-05-11). Flat across all plan tiers — only the included character + // bundle changes per plan. + expect(cost).toBeCloseTo(0.05, 3); }); it('calculates openai_tts cost', () => { @@ -204,6 +207,28 @@ describe('calculateTelephonyCost', () => { const cost = calculateTelephonyCost('telnyx', 60, pricing); expect(cost).toBeCloseTo(0.007, 3); }); + + it('telnyx_inbound bills at $0.0035/min (US DID local termination)', () => { + // Verified against https://telnyx.com/pricing/elastic-sip (2026-05-11). + const pricing = mergePricing(); + const cost = calculateTelephonyCost('telnyx_inbound', 60, pricing); + expect(cost).toBeCloseTo(0.0035, 4); + }); + + it('telnyx_outbound bills at $0.007/min (Pay-As-You-Go mid-range)', () => { + const pricing = mergePricing(); + const cost = calculateTelephonyCost('telnyx_outbound', 60, pricing); + expect(cost).toBeCloseTo(0.007, 3); + }); + + it('legacy telnyx key falls back to outbound rate ($0.007/min)', () => { + // Users who override ``pricing: { telnyx: {...} }`` without a + // direction-aware split keep the previous behaviour. Direction-aware + // billing is opt-in via ``telnyx_inbound`` / ``telnyx_outbound``. + const pricing = mergePricing(); + const cost = calculateTelephonyCost('telnyx', 60, pricing); + expect(cost).toBeCloseTo(0.007, 3); + }); }); describe('per-model rates under openai_realtime.models', () => { @@ -256,14 +281,15 @@ describe('per-model rates under openai_realtime.models', () => { describe('model-aware STT pricing', () => { it('deepgram default is nova-3 streaming', () => { const pricing = mergePricing(); - // 60s at $0.0077/min = $0.0077 - expect(calculateSttCost('deepgram', 60, pricing)).toBeCloseTo(0.0077, 6); + // 60s at $0.0048/min = $0.0048 (PAYG promo rate, verified 2026-05-11). + expect(calculateSttCost('deepgram', 60, pricing)).toBeCloseTo(0.0048, 6); }); it('deepgram multilingual nested rate', () => { const pricing = mergePricing(); + // nova-3-multilingual is $0.0058/min on the PAYG promo tier. expect(calculateSttCost('deepgram', 60, pricing, 'nova-3-multilingual')).toBeCloseTo( - 0.0092, + 0.0058, 6, ); }); @@ -283,7 +309,7 @@ describe('model-aware STT pricing', () => { it('unknown model falls back to provider default', () => { const pricing = mergePricing(); expect(calculateSttCost('deepgram', 60, pricing, 'some-future-model')).toBeCloseTo( - 0.0077, + 0.0048, 6, ); }); @@ -292,13 +318,15 @@ describe('model-aware STT pricing', () => { describe('model-aware TTS pricing', () => { it('elevenlabs default is flash_v2_5', () => { const pricing = mergePricing(); - expect(calculateTtsCost('elevenlabs', 1000, pricing)).toBeCloseTo(0.06, 6); + // Default == flash_v2_5 at $0.05/1k (public API rate verified 2026-05-11). + expect(calculateTtsCost('elevenlabs', 1000, pricing)).toBeCloseTo(0.05, 6); }); it('elevenlabs multilingual_v2 nested rate', () => { const pricing = mergePricing(); + // Multilingual v2 / v3 share the $0.10/1k tier per the public API page. expect(calculateTtsCost('elevenlabs', 1000, pricing, 'eleven_multilingual_v2')).toBeCloseTo( - 0.18, + 0.10, 6, ); }); @@ -320,9 +348,9 @@ describe('per-model override merge semantics', () => { 0.04, 6, ); - // Untouched + // Untouched — still original $0.10 expect(calculateTtsCost('elevenlabs', 1000, pricing, 'eleven_multilingual_v2')).toBeCloseTo( - 0.18, + 0.10, 6, ); }); @@ -361,14 +389,17 @@ describe('LLM cost billing — Cerebras + Groq silent under-billing regression', it('cerebras default model (gpt-oss-120b) is billed', () => { const cost = calculateLlmCost('cerebras', 'gpt-oss-120b', 1000, 1000); - // Real rate-card math: 1000 in @ $0.85/M + 1000 out @ $1.20/M - expect(cost).toBeCloseTo((1000 / 1_000_000) * 0.85 + (1000 / 1_000_000) * 1.20, 9); + // Real rate-card math: 1000 in @ $0.35/M + 1000 out @ $0.75/M + // (verified at inference-docs.cerebras.ai/models/openai-oss). + expect(cost).toBeCloseTo((1000 / 1_000_000) * 0.35 + (1000 / 1_000_000) * 0.75, 9); expect(cost).toBeGreaterThan(0); }); it('cerebras llama3.1-8b is billed (still supported until 2026-05-27 retirement)', () => { const cost = calculateLlmCost('cerebras', 'llama3.1-8b', 1000, 1000); - expect(cost).toBeCloseTo((1000 / 1_000_000) * 0.10 + (1000 / 1_000_000) * 0.20, 9); + // 1000 in @ $0.10/M + 1000 out @ $0.10/M + // (verified at inference-docs.cerebras.ai/models/llama-31-8b). + expect(cost).toBeCloseTo((1000 / 1_000_000) * 0.10 + (1000 / 1_000_000) * 0.10, 9); expect(cost).toBeGreaterThan(0); }); diff --git a/libraries/typescript/tests/server.test.ts b/libraries/typescript/tests/server.test.ts index b826942f..b45e6f8b 100644 --- a/libraries/typescript/tests/server.test.ts +++ b/libraries/typescript/tests/server.test.ts @@ -313,3 +313,84 @@ describe('sanitizeVariables', () => { expect(Object.keys(result)).toHaveLength(0); }); }); + +// --------------------------------------------------------------------------- +// Bug B regression — outbound calls persist caller/callee on disk +// --------------------------------------------------------------------------- + +describe('EmbeddedServer wraps logging callbacks with active-record fallback', () => { + it('persists caller/callee from active record when on_call_start data is empty', async () => { + // eslint-disable-next-line @typescript-eslint/no-require-imports + const fs = require('node:fs') as typeof import('node:fs'); + // eslint-disable-next-line @typescript-eslint/no-require-imports + const os = require('node:os') as typeof import('node:os'); + // eslint-disable-next-line @typescript-eslint/no-require-imports + const path = require('node:path') as typeof import('node:path'); + + const tmp = fs.mkdtempSync(path.join(os.tmpdir(), 'patter-bug-b-')); + try { + const server = new EmbeddedServer( + makeConfig({ persistRoot: tmp }), + makeAgent(), + ); + // Pre-register the call as recordCallInitiated does for outbound. + const store = (server as unknown as { + metricsStore: { + recordCallInitiated: (d: Record) => void; + }; + }).metricsStore; + store.recordCallInitiated({ + call_id: 'CA-outbound', + caller: '+15551112222', + callee: '+15553334444', + direction: 'outbound', + }); + + // Reach into the private wrapper. Tests for private methods are kept + // tight: they exist purely to lock in the fix for Bug B. + const wrapped = (server as unknown as { + wrapLoggingCallbacks: (b: { telephonyProvider: string }) => [ + (d: Record) => Promise, + (d: Record) => Promise, + (d: Record) => Promise, + ]; + }).wrapLoggingCallbacks({ telephonyProvider: 'twilio' }); + const wrappedStart = wrapped[0]; + + // Simulate the bridge's on_call_start payload for an outbound call: + // the WS query string was empty, so caller/callee are blank in + // ``data``. The fix must look up the active record from the store + // and persist the real numbers. + await wrappedStart({ + call_id: 'CA-outbound', + caller: '', + callee: '', + direction: 'outbound', + telephony_provider: 'twilio', + }); + + // Allow the fire-and-forget logCallStart promise to drain. + await new Promise((resolve) => setTimeout(resolve, 50)); + + const metaPaths: string[] = []; + const walk = (d: string): void => { + for (const entry of fs.readdirSync(d, { withFileTypes: true })) { + const full = path.join(d, entry.name); + if (entry.isDirectory()) walk(full); + else if (entry.name === 'metadata.json') metaPaths.push(full); + } + }; + walk(tmp); + expect(metaPaths).toHaveLength(1); + const payload = JSON.parse(fs.readFileSync(metaPaths[0], 'utf8')) as { + caller: string; + callee: string; + }; + // Default redact mode is "mask" — last-4 visible. + expect(payload.caller.endsWith('2222')).toBe(true); + expect(payload.callee.endsWith('4444')).toBe(true); + } finally { + fs.rmSync(tmp, { recursive: true, force: true }); + } + }); +}); diff --git a/libraries/typescript/tests/soak/soak.test.ts b/libraries/typescript/tests/soak/soak.test.ts index 4a8ef1ea..5ae4e80c 100644 --- a/libraries/typescript/tests/soak/soak.test.ts +++ b/libraries/typescript/tests/soak/soak.test.ts @@ -113,16 +113,18 @@ describe("Soak Tests", () => { // Verify turn count expect(metrics.turns).toHaveLength(NUM_TURNS); - // STT cost: deepgram nova-3 streaming = $0.0077/min + // STT cost: deepgram nova-3 streaming = $0.0048/min (PAYG promo rate + // per deepgram.com/pricing, verified 2026-05-11). // total_audio = 1.5 * 1000 = 1500s = 25 min const expectedStt = - (PER_TURN_AUDIO_SECONDS * NUM_TURNS) / 60.0 * 0.0077; + (PER_TURN_AUDIO_SECONDS * NUM_TURNS) / 60.0 * 0.0048; expect(Math.abs(metrics.cost.stt - Math.round(expectedStt * 1e6) / 1e6)).toBeLessThan(1e-6); - // TTS cost: elevenlabs eleven_flash_v2_5 = $0.06/1k chars + // TTS cost: elevenlabs eleven_flash_v2_5 = $0.05/1k chars (public API + // tier, verified 2026-05-11 vs elevenlabs.io/pricing/api). // total_chars = 20 * 1000 = 20000 = 20 k_chars const expectedTts = - (AGENT_RESPONSE.length * NUM_TURNS) / 1000.0 * 0.06; + (AGENT_RESPONSE.length * NUM_TURNS) / 1000.0 * 0.05; expect(Math.abs(metrics.cost.tts - Math.round(expectedTts * 1e6) / 1e6)).toBeLessThan(1e-6); const growthPct = rssBefore > 0 diff --git a/libraries/typescript/tests/transcoding.test.ts b/libraries/typescript/tests/transcoding.test.ts index 17c02c6f..a6319048 100644 --- a/libraries/typescript/tests/transcoding.test.ts +++ b/libraries/typescript/tests/transcoding.test.ts @@ -148,22 +148,24 @@ describe("resample16kTo8k", () => { const output = resample16kTo8k(input); expect(output.length).toBe(4); - // Output[0] at center=0 with edge-clamped s_m1, s_m2 = 100: - // (100 + 4*100 + 6*100 + 4*200 + 300 + 8) >> 4 = 2208 >> 4 = 138 - expect(output.readInt16LE(0)).toBe(138); - // Output[1] at center=2 with edge-clamped s_p2 = 400: + // Output[0] at center=0 with zero-initialized history (s_m1=s_m2=0): + // (0 + 4*0 + 6*100 + 4*200 + 300 + 8) >> 4 = 1708 >> 4 = 106 + expect(output.readInt16LE(0)).toBe(106); + // Output[1] at center=2 with s_m2=100, s_m1=200; s_p2 edge-clamped=400: // (100 + 4*200 + 6*300 + 4*400 + 400 + 8) >> 4 = 4708 >> 4 = 294 expect(output.readInt16LE(2)).toBe(294); }); - it("passes DC through unchanged (no aliasing on a constant signal)", () => { - // Filter is normalised (coefficients sum to 16, then >> 4) so a constant - // input must come out as the same constant. + it("passes DC through unchanged after startup transient (FIR unity gain on DC)", () => { + // The FIR history is zero-initialized (correct initial condition for no + // prior audio). The first output sample blends zero history with the DC + // signal — expected startup transient. From sample[1] onward the filter + // is in steady state and unity gain at DC gives exactly 5000 (±1 LSB). const input = Buffer.alloc(20); for (let i = 0; i < 10; i++) input.writeInt16LE(5000, i * 2); const output = resample16kTo8k(input); expect(output.length).toBe(10); - for (let i = 0; i < 5; i++) { + for (let i = 1; i < 5; i++) { // Allow +/- 1 LSB for rounding with (sum + 8) >> 4. const v = output.readInt16LE(i * 2); expect(Math.abs(v - 5000)).toBeLessThanOrEqual(1); diff --git a/libraries/typescript/tests/tts-facade-language.test.ts b/libraries/typescript/tests/tts-facade-language.test.ts index e18f4d01..95d99235 100644 --- a/libraries/typescript/tests/tts-facade-language.test.ts +++ b/libraries/typescript/tests/tts-facade-language.test.ts @@ -9,6 +9,8 @@ import { afterEach, beforeEach, describe, expect, it } from "vitest"; +import { ElevenLabsRestTTS } from "../src"; +import { ElevenLabsWebSocketTTS } from "../src/providers/elevenlabs-ws-tts"; import * as elevenlabs from "../src/tts/elevenlabs"; const ENV_KEY = "ELEVENLABS_API_KEY"; @@ -72,4 +74,24 @@ describe("[unit] tts/elevenlabs facade — languageCode forwarding", () => { delete process.env[ENV_KEY]; expect(() => new elevenlabs.TTS()).toThrow(/ELEVENLABS_API_KEY/); }); + + it("defaults to the HTTP REST adapter", () => { + const tts = new elevenlabs.TTS(); + // The facade extends ElevenLabsTTS (HTTP REST), not the WebSocket class. + expect(tts).not.toBeInstanceOf(ElevenLabsWebSocketTTS); + // Static provider key tag for callers / dashboards. + expect((elevenlabs.TTS as unknown as { providerKey: string }).providerKey).toBe( + "elevenlabs", + ); + }); + + it("ElevenLabsRestTTS opt-out is not aliased to the WebSocket class", () => { + // Explicit REST opt-out must remain distinct from the WS default. + expect(ElevenLabsRestTTS).not.toBe(ElevenLabsWebSocketTTS); + const rest = new ElevenLabsRestTTS({ apiKey: "test-key" }); + expect(rest).not.toBeInstanceOf(ElevenLabsWebSocketTTS); + // REST transport drives chunking client-side — historical default kept. + const stored = rest as unknown as { chunkSize?: number }; + expect(stored.chunkSize).toBe(4096); + }); }); diff --git a/libraries/typescript/tests/unit/barge-in-strategies.test.ts b/libraries/typescript/tests/unit/barge-in-strategies.test.ts new file mode 100644 index 00000000..b72577be --- /dev/null +++ b/libraries/typescript/tests/unit/barge-in-strategies.test.ts @@ -0,0 +1,198 @@ +import { describe, it, expect } from 'vitest'; +import { + MinWordsStrategy, + evaluateStrategies, + resetStrategies, + type BargeInStrategy, + type EvaluateContext, +} from '../../src/services/barge-in-strategies'; + +describe('MinWordsStrategy', () => { + it('rejects minWords < 1 in the constructor', () => { + expect(() => new MinWordsStrategy({ minWords: 0 })).toThrow(); + expect(() => new MinWordsStrategy({ minWords: -3 })).toThrow(); + expect(() => new MinWordsStrategy({ minWords: NaN })).toThrow(); + }); + + it('lets a single word confirm when the agent is silent', () => { + const s = new MinWordsStrategy({ minWords: 3 }); + expect( + s.evaluate({ transcript: 'hi', isInterim: false, agentSpeaking: false }), + ).toBe(true); + }); + + it('does not confirm sub-threshold transcripts during agent speech', () => { + const s = new MinWordsStrategy({ minWords: 3 }); + expect( + s.evaluate({ transcript: 'okay', isInterim: false, agentSpeaking: true }), + ).toBe(false); + expect( + s.evaluate({ + transcript: 'uh huh', + isInterim: false, + agentSpeaking: true, + }), + ).toBe(false); + }); + + it('confirms once the threshold is met during agent speech', () => { + const s = new MinWordsStrategy({ minWords: 3 }); + expect( + s.evaluate({ + transcript: 'please stop talking', + isInterim: false, + agentSpeaking: true, + }), + ).toBe(true); + expect( + s.evaluate({ + transcript: 'hold on a moment please', + isInterim: false, + agentSpeaking: true, + }), + ).toBe(true); + }); + + it('useInterim=false ignores interim partials but still accepts finals', () => { + const s = new MinWordsStrategy({ minWords: 2, useInterim: false }); + expect( + s.evaluate({ + transcript: 'please stop', + isInterim: true, + agentSpeaking: true, + }), + ).toBe(false); + expect( + s.evaluate({ + transcript: 'please stop', + isInterim: false, + agentSpeaking: true, + }), + ).toBe(true); + }); + + it('counts words via whitespace split (collapses runs and tabs)', () => { + const s = new MinWordsStrategy({ minWords: 2 }); + expect( + s.evaluate({ + transcript: ' hello world ', + isInterim: false, + agentSpeaking: true, + }), + ).toBe(true); + expect( + s.evaluate({ + transcript: '\thello\n', + isInterim: false, + agentSpeaking: true, + }), + ).toBe(false); + }); + + it('does not confirm an empty transcript during agent speech', () => { + const s = new MinWordsStrategy({ minWords: 2 }); + expect( + s.evaluate({ transcript: '', isInterim: false, agentSpeaking: true }), + ).toBe(false); + }); +}); + +class RecordingStrategy implements BargeInStrategy { + calls: EvaluateContext[] = []; + resets = 0; + constructor(private readonly returns: boolean) {} + evaluate(ctx: EvaluateContext): boolean { + this.calls.push(ctx); + return this.returns; + } + async reset(): Promise { + this.resets += 1; + } +} + +describe('evaluateStrategies', () => { + const baseCtx: EvaluateContext = { + transcript: 'please stop', + isInterim: false, + agentSpeaking: true, + }; + + it('returns false on empty array', async () => { + expect(await evaluateStrategies([], baseCtx)).toBe(false); + }); + + it('short-circuits at the first true', async () => { + const a = new RecordingStrategy(true); + const b = new RecordingStrategy(false); + const result = await evaluateStrategies([a, b], baseCtx); + expect(result).toBe(true); + expect(a.calls).toHaveLength(1); + expect(b.calls).toHaveLength(0); + }); + + it('returns false when every strategy returns false', async () => { + const a = new RecordingStrategy(false); + const b = new RecordingStrategy(false); + const result = await evaluateStrategies([a, b], baseCtx); + expect(result).toBe(false); + expect(a.calls).toHaveLength(1); + expect(b.calls).toHaveLength(1); + }); + + it('swallows a strategy that throws and continues to the next', async () => { + const boom: BargeInStrategy = { + evaluate() { + throw new Error('boom'); + }, + }; + const ok = new RecordingStrategy(true); + const result = await evaluateStrategies([boom, ok], baseCtx); + expect(result).toBe(true); + expect(ok.calls).toHaveLength(1); + }); + + it('coerces null/undefined transcript to empty string before passing through', async () => { + const seen: EvaluateContext[] = []; + const recorder: BargeInStrategy = { + evaluate(ctx) { + seen.push(ctx); + return false; + }, + }; + await evaluateStrategies([recorder], { + transcript: undefined as unknown as string, + isInterim: false, + agentSpeaking: true, + }); + expect(seen[0]?.transcript).toBe(''); + }); +}); + +describe('resetStrategies', () => { + it('resets each strategy that exposes a reset() method', async () => { + const a = new RecordingStrategy(false); + const b = new RecordingStrategy(false); + await resetStrategies([a, b]); + expect(a.resets).toBe(1); + expect(b.resets).toBe(1); + }); + + it('skips strategies that do not implement reset()', async () => { + const noReset: BargeInStrategy = { evaluate: () => false }; + const a = new RecordingStrategy(false); + await resetStrategies([noReset, a]); + expect(a.resets).toBe(1); + }); + + it('swallows per-strategy reset errors', async () => { + const boom: BargeInStrategy = { + evaluate: () => false, + async reset() { + throw new Error('boom'); + }, + }; + const ok = new RecordingStrategy(false); + await resetStrategies([boom, ok]); + expect(ok.resets).toBe(1); + }); +}); diff --git a/libraries/typescript/tests/unit/barge-in-two-stage.test.ts b/libraries/typescript/tests/unit/barge-in-two-stage.test.ts new file mode 100644 index 00000000..2fc35e64 --- /dev/null +++ b/libraries/typescript/tests/unit/barge-in-two-stage.test.ts @@ -0,0 +1,368 @@ +/** + * Two-stage barge-in regression tests. + * + * Mirrors the Python ``test_barge_in_two_stage.py`` so the cross-SDK + * behaviour stays in lockstep: + * + * - sub-threshold transcripts during agent speech do not cancel + * - threshold-meeting transcripts confirm and run the cancel path + * - empty strategies preserve the legacy "cancel on first transcript" + * behaviour byte-for-byte + * - VAD speech_start with strategies marks pending (no cancel yet) and + * the timeout drops pending state without flipping isSpeaking + */ +import { describe, it, expect, vi, beforeEach, afterEach } from 'vitest'; +import type { TelephonyBridge, StreamHandlerDeps } from '../../src/stream-handler'; +import { StreamHandler } from '../../src/stream-handler'; +import { MetricsStore } from '../../src/dashboard/store'; +import { RemoteMessageHandler } from '../../src/remote-message'; +import { MinWordsStrategy } from '../../src/services/barge-in-strategies'; +import type { BargeInStrategy } from '../../src/services/barge-in-strategies'; +import type { WebSocket as WSWebSocket } from 'ws'; + +function makeMockBridge(overrides?: Partial): TelephonyBridge { + return { + label: 'TestBridge', + telephonyProvider: 'twilio', + sendAudio: vi.fn(), + sendMark: vi.fn(), + sendClear: vi.fn(), + transferCall: vi.fn().mockResolvedValue(undefined), + endCall: vi.fn().mockResolvedValue(undefined), + createStt: vi.fn().mockReturnValue(null), + queryTelephonyCost: vi.fn().mockResolvedValue(undefined), + ...overrides, + }; +} + +function makeMockWs(): WSWebSocket { + return { + send: vi.fn(), + close: vi.fn(), + on: vi.fn(), + once: vi.fn(), + readyState: 1, + removeListener: vi.fn(), + addEventListener: vi.fn(), + removeEventListener: vi.fn(), + } as unknown as WSWebSocket; +} + +function makeDeps( + bargeInStrategies: readonly BargeInStrategy[] = [], + bargeInConfirmMs?: number, +): StreamHandlerDeps { + return { + config: { openaiKey: 'test-oai-key' }, + agent: { + systemPrompt: 'Test agent', + provider: 'pipeline', + bargeInStrategies, + bargeInConfirmMs, + }, + bridge: makeMockBridge(), + metricsStore: new MetricsStore(), + pricing: null, + remoteHandler: new RemoteMessageHandler(), + recording: false, + buildAIAdapter: vi.fn(), + sanitizeVariables: vi.fn((raw) => { + const safe: Record = {}; + for (const [k, v] of Object.entries(raw)) safe[k] = String(v); + return safe; + }), + resolveVariables: vi.fn((tpl) => tpl), + }; +} + +interface Priv { + isSpeaking: boolean; + speakingStartedAt: number | null; + firstAudioSentAt: number | null; + bargeInPendingSince: number | null; + bargeInPendingTimer: ReturnType | null; + llmAbort: AbortController | null; + handleBargeIn: (t: { text?: string; isFinal?: boolean }) => boolean; + handleBargeInAsync: (t: { text?: string; isFinal?: boolean }) => Promise; + startPendingBargeIn: () => void; + clearPendingBargeIn: () => void; +} + +function priv(h: StreamHandler): Priv { + return h as unknown as Priv; +} + +function armSpeakingState(h: StreamHandler): void { + // Simulate "agent has been speaking for >1 s and the first chunk + // already hit the wire" so the canBargeIn gate is open. + const p = priv(h); + p.isSpeaking = true; + p.speakingStartedAt = Date.now() - 1500; + p.firstAudioSentAt = Date.now() - 1500; + p.llmAbort = new AbortController(); +} + +describe('StreamHandler — opt-in barge-in confirmation', () => { + beforeEach(() => { + vi.useFakeTimers({ toFake: ['setTimeout', 'clearTimeout'] }); + }); + + afterEach(() => { + vi.restoreAllMocks(); + vi.useRealTimers(); + }); + + it('legacy path (no strategies) cancels immediately on any transcript while speaking', () => { + const deps = makeDeps([]); + const h = new StreamHandler(deps, makeMockWs(), '+1', '+2'); + armSpeakingState(h); + const result = priv(h).handleBargeIn({ text: 'okay', isFinal: true }); + expect(result).toBe(true); + expect(priv(h).isSpeaking).toBe(false); + }); + + it('sub-threshold transcript does NOT cancel when MinWordsStrategy is configured', async () => { + const deps = makeDeps([new MinWordsStrategy({ minWords: 3 })]); + const h = new StreamHandler(deps, makeMockWs(), '+1', '+2'); + armSpeakingState(h); + const confirmed = await priv(h).handleBargeInAsync({ + text: 'okay', + isFinal: true, + }); + expect(confirmed).toBe(false); + expect(priv(h).isSpeaking).toBe(true); + // No cancel was issued — sendClear must not have been called for + // this attempted barge-in. + expect(deps.bridge.sendClear).not.toHaveBeenCalled(); + }); + + it('threshold-meeting transcript confirms and runs the cancel path', async () => { + const deps = makeDeps([new MinWordsStrategy({ minWords: 3 })]); + const h = new StreamHandler(deps, makeMockWs(), '+1', '+2'); + armSpeakingState(h); + const confirmed = await priv(h).handleBargeInAsync({ + text: 'please stop talking now', + isFinal: true, + }); + expect(confirmed).toBe(true); + expect(priv(h).isSpeaking).toBe(false); + expect(deps.bridge.sendClear).toHaveBeenCalledTimes(1); + }); + + it('startPendingBargeIn marks pending without cancelling and sets a timer', () => { + const deps = makeDeps([new MinWordsStrategy({ minWords: 3 })], 800); + const h = new StreamHandler(deps, makeMockWs(), '+1', '+2'); + armSpeakingState(h); + priv(h).startPendingBargeIn(); + expect(priv(h).bargeInPendingSince).not.toBeNull(); + expect(priv(h).bargeInPendingTimer).not.toBeNull(); + // Crucially, no cancel was issued just because VAD fired. + expect(priv(h).isSpeaking).toBe(true); + expect(deps.bridge.sendClear).not.toHaveBeenCalled(); + }); + + it('pending barge-in times out and drops pending state (agent keeps speaking)', () => { + const deps = makeDeps([new MinWordsStrategy({ minWords: 3 })], 50); + const h = new StreamHandler(deps, makeMockWs(), '+1', '+2'); + armSpeakingState(h); + priv(h).startPendingBargeIn(); + expect(priv(h).bargeInPendingSince).not.toBeNull(); + // Advance virtual time past the timeout. + vi.advanceTimersByTime(60); + expect(priv(h).bargeInPendingSince).toBeNull(); + expect(priv(h).bargeInPendingTimer).toBeNull(); + // Agent never got cancelled — that's the whole point of the + // confirmation pipeline. + expect(priv(h).isSpeaking).toBe(true); + expect(deps.bridge.sendClear).not.toHaveBeenCalled(); + }); + + it('confirmation clears pending state', async () => { + const deps = makeDeps([new MinWordsStrategy({ minWords: 2 })], 10_000); + const h = new StreamHandler(deps, makeMockWs(), '+1', '+2'); + armSpeakingState(h); + priv(h).startPendingBargeIn(); + expect(priv(h).bargeInPendingSince).not.toBeNull(); + + const confirmed = await priv(h).handleBargeInAsync({ + text: 'please stop', + isFinal: true, + }); + expect(confirmed).toBe(true); + expect(priv(h).bargeInPendingSince).toBeNull(); + expect(priv(h).bargeInPendingTimer).toBeNull(); + expect(priv(h).isSpeaking).toBe(false); + }); + + it('startPendingBargeIn is idempotent within a turn', () => { + const deps = makeDeps([new MinWordsStrategy({ minWords: 3 })], 10_000); + const h = new StreamHandler(deps, makeMockWs(), '+1', '+2'); + armSpeakingState(h); + priv(h).startPendingBargeIn(); + const firstSince = priv(h).bargeInPendingSince; + const firstTimer = priv(h).bargeInPendingTimer; + priv(h).startPendingBargeIn(); + expect(priv(h).bargeInPendingSince).toBe(firstSince); + expect(priv(h).bargeInPendingTimer).toBe(firstTimer); + }); +}); + +describe('StreamHandler — overlap window preserved across VAD → strategy confirm (FIX #88)', () => { + it('strategy-confirmed cancel does NOT restart the overlap window', async () => { + // Use real timers for this test — we need real time to elapse so + // detectionDelay reflects the VAD→confirm window. + vi.useRealTimers(); + const deps = makeDeps([new MinWordsStrategy({ minWords: 3 })], 10_000); + const h = new StreamHandler(deps, makeMockWs(), '+1', '+2'); + armSpeakingState(h); + + // Stage 1: VAD fires speech_start → pending. Records overlap_start (T1). + priv(h).startPendingBargeIn(); + expect(priv(h).bargeInPendingSince).not.toBeNull(); + + // Wait ~150ms so that if T1 is preserved, detectionDelay >= 150 ms; + // if T1 is overwritten by the cancel path, detectionDelay ≈ 0. + await new Promise((r) => setTimeout(r, 150)); + + // Subscribe to the metrics event bus on the handler to observe the + // emitted InterruptionMetrics payload. + interface PrivWithMetrics { + metricsAcc: { attachEventBus: (b: unknown) => void }; + } + const { EventBus } = await import('../../src/observability/event-bus'); + const bus = new EventBus(); + const captured: { detectionDelay: number }[] = []; + bus.on('interruption', (p: unknown) => { + captured.push(p as { detectionDelay: number }); + }); + (h as unknown as PrivWithMetrics).metricsAcc.attachEventBus(bus); + + // Stage 2: STT delivers a confirming transcript NOW (T2). + const confirmed = await priv(h).handleBargeInAsync({ + text: 'please stop talking now', + isFinal: true, + }); + expect(confirmed).toBe(true); + expect(priv(h).isSpeaking).toBe(false); + + // Exactly one InterruptionMetrics emission. detectionDelay must + // reflect the VAD→confirm window (~150 ms), NOT ~0. + expect(captured.length).toBe(1); + expect(captured[0].detectionDelay).toBeGreaterThanOrEqual(100); + expect(captured[0].detectionDelay).toBeLessThanOrEqual(800); + }); + + it('legacy path (no strategies) records overlap_start once', () => { + const deps = makeDeps([]); + const h = new StreamHandler(deps, makeMockWs(), '+1', '+2'); + armSpeakingState(h); + + interface PrivMetrics { + metricsAcc: { recordOverlapStart: () => void; recordOverlapEnd: (b: boolean) => void }; + } + const acc = (h as unknown as PrivMetrics).metricsAcc; + const startSpy = vi.spyOn(acc, 'recordOverlapStart'); + const endSpy = vi.spyOn(acc, 'recordOverlapEnd'); + + const result = priv(h).handleBargeIn({ text: 'okay', isFinal: true }); + expect(result).toBe(true); + // Without VAD pending, the cancel path is the SOLE caller of + // recordOverlapStart — exactly once. + expect(startSpy).toHaveBeenCalledTimes(1); + expect(endSpy).toHaveBeenCalledTimes(1); + }); +}); + +describe('StreamHandler — handleStop / handleWsClose drops pending barge-in timer (FIX #89)', () => { + beforeEach(() => { + vi.useFakeTimers({ toFake: ['setTimeout', 'clearTimeout'] }); + }); + afterEach(() => { + vi.restoreAllMocks(); + vi.useRealTimers(); + }); + + it('handleStop cancels a pending barge-in timer so it cannot fire later', async () => { + const deps = makeDeps([new MinWordsStrategy({ minWords: 3 })], 1_500); + const h = new StreamHandler(deps, makeMockWs(), '+1', '+2'); + armSpeakingState(h); + + interface PrivMetrics { + metricsAcc: { recordOverlapEnd: (b: boolean) => void }; + } + const acc = (h as unknown as PrivMetrics).metricsAcc; + const endSpy = vi.spyOn(acc, 'recordOverlapEnd'); + + priv(h).startPendingBargeIn(); + expect(priv(h).bargeInPendingTimer).not.toBeNull(); + expect(priv(h).bargeInPendingSince).not.toBeNull(); + // Reset the spy after startPendingBargeIn (which doesn't call end, + // but be defensive). + endSpy.mockClear(); + + await h.handleStop(); + + // Pending state cleared; timer cancelled. + expect(priv(h).bargeInPendingSince).toBeNull(); + expect(priv(h).bargeInPendingTimer).toBeNull(); + + // Advance past when the timeout would have fired. If the timer + // wasn't cancelled, recordOverlapEnd would fire here on a torn-down + // metrics object — that's the regression. + vi.advanceTimersByTime(2_000); + expect(endSpy).not.toHaveBeenCalled(); + }); + + it('handleWsClose cancels a pending barge-in timer', async () => { + const deps = makeDeps([new MinWordsStrategy({ minWords: 3 })], 1_500); + const h = new StreamHandler(deps, makeMockWs(), '+1', '+2'); + armSpeakingState(h); + + interface PrivMetrics { + metricsAcc: { recordOverlapEnd: (b: boolean) => void }; + } + const acc = (h as unknown as PrivMetrics).metricsAcc; + const endSpy = vi.spyOn(acc, 'recordOverlapEnd'); + + priv(h).startPendingBargeIn(); + expect(priv(h).bargeInPendingTimer).not.toBeNull(); + endSpy.mockClear(); + + await h.handleWsClose(); + + expect(priv(h).bargeInPendingSince).toBeNull(); + expect(priv(h).bargeInPendingTimer).toBeNull(); + vi.advanceTimersByTime(2_000); + expect(endSpy).not.toHaveBeenCalled(); + }); +}); + +describe('MinWordsStrategy threshold parity (TS↔Py)', () => { + it.each([2, 3, 5])( + 'agent stays talking below threshold and cancels at threshold (minWords=%i)', + async (n: number) => { + const deps = makeDeps([new MinWordsStrategy({ minWords: n })]); + const h = new StreamHandler(deps, makeMockWs(), '+1', '+2'); + armSpeakingState(h); + + const below = Array.from({ length: n - 1 }, () => 'word').join(' '); + const at = Array.from({ length: n }, () => 'word').join(' '); + + // Below threshold — keep talking. + let confirmed = await priv(h).handleBargeInAsync({ + text: below, + isFinal: true, + }); + expect(confirmed).toBe(false); + expect(priv(h).isSpeaking).toBe(true); + + // At threshold — confirm. + confirmed = await priv(h).handleBargeInAsync({ + text: at, + isFinal: true, + }); + expect(confirmed).toBe(true); + expect(priv(h).isSpeaking).toBe(false); + }, + ); +}); diff --git a/libraries/typescript/tests/unit/dashboard-store.test.ts b/libraries/typescript/tests/unit/dashboard-store.test.ts index 6c25f397..a5a19dab 100644 --- a/libraries/typescript/tests/unit/dashboard-store.test.ts +++ b/libraries/typescript/tests/unit/dashboard-store.test.ts @@ -312,6 +312,121 @@ describe('MetricsStore', () => { }); }); + // --- Soft delete --- + + describe('deleteCalls()', () => { + function seedCompleted(store: MetricsStore, id: string, latencyMs = 200, costTotal = 0.01) { + store.recordCallStart({ + call_id: id, + caller: '+15551111111', + callee: '+15552222222', + direction: 'inbound', + }); + store.recordCallEnd( + { call_id: id }, + { + duration_seconds: 30, + cost: { total: costTotal, stt: 0, tts: 0, llm: 0, telephony: costTotal }, + latency_avg: { agent_response_ms: latencyMs }, + }, + ); + } + + it('hides deleted calls from getCalls / getCall / callCount', () => { + const store = new MetricsStore(); + seedCompleted(store, 'keep-1'); + seedCompleted(store, 'drop-1'); + expect(store.callCount).toBe(2); + + const accepted = store.deleteCalls(['drop-1']); + expect(accepted).toEqual(['drop-1']); + expect(store.callCount).toBe(1); + expect(store.getCalls()).toHaveLength(1); + expect(store.getCalls()[0].call_id).toBe('keep-1'); + expect(store.getCall('drop-1')).toBeNull(); + expect(store.getCall('keep-1')).not.toBeNull(); + expect(store.isDeleted('drop-1')).toBe(true); + expect(store.isDeleted('keep-1')).toBe(false); + }); + + it('excludes deleted calls from aggregates so avg latency + cost shift', () => { + const store = new MetricsStore(); + seedCompleted(store, 'fast', 100, 0.01); + seedCompleted(store, 'slow', 900, 0.05); + const before = store.getAggregates() as Record; + expect(before.total_calls).toBe(2); + expect(before.avg_latency_ms).toBe(500); // (100 + 900) / 2 + + store.deleteCalls(['slow']); + const after = store.getAggregates() as Record; + expect(after.total_calls).toBe(1); + expect(after.avg_latency_ms).toBe(100); // only "fast" remains + expect(after.total_cost).toBe(0.01); + }); + + it('excludes deleted calls from getCallsInRange', () => { + const store = new MetricsStore(); + seedCompleted(store, 'a'); + seedCompleted(store, 'b'); + expect(store.getCallsInRange()).toHaveLength(2); + store.deleteCalls(['b']); + const remaining = store.getCallsInRange(); + expect(remaining).toHaveLength(1); + expect(remaining[0].call_id).toBe('a'); + }); + + it('refuses to delete active calls', () => { + const store = new MetricsStore(); + store.recordCallStart({ + call_id: 'live-1', + caller: '+15551111111', + callee: '+15552222222', + direction: 'inbound', + }); + const accepted = store.deleteCalls(['live-1']); + expect(accepted).toEqual([]); + expect(store.isDeleted('live-1')).toBe(false); + expect(store.getActiveCalls()).toHaveLength(1); + }); + + it('is idempotent — re-deleting an id returns empty + no extra event', () => { + const store = new MetricsStore(); + seedCompleted(store, 'x'); + const events: SSEEvent[] = []; + store.on('sse', (e: SSEEvent) => events.push(e)); + const first = store.deleteCalls(['x']); + const second = store.deleteCalls(['x']); + expect(first).toEqual(['x']); + expect(second).toEqual([]); + const deletedEvents = events.filter((e) => e.type === 'calls_deleted'); + expect(deletedEvents).toHaveLength(1); + }); + + it('emits SSE calls_deleted with the accepted ids', () => { + const store = new MetricsStore(); + seedCompleted(store, 'a'); + seedCompleted(store, 'b'); + const events: SSEEvent[] = []; + store.on('sse', (e: SSEEvent) => events.push(e)); + const accepted = store.deleteCalls(['a', 'b']); + expect(accepted).toEqual(['a', 'b']); + const deletedEvent = events.find((e) => e.type === 'calls_deleted'); + expect(deletedEvent).toBeDefined(); + expect(deletedEvent!.data.call_ids).toEqual(['a', 'b']); + }); + + it('handles empty / non-string / unknown ids gracefully', () => { + const store = new MetricsStore(); + seedCompleted(store, 'real'); + // unknown ids are still accepted into the deleted-set (so a future + // hydrate that resurrects them stays hidden) — matches Python. + expect(store.deleteCalls([])).toEqual([]); + expect(store.deleteCalls([''])).toEqual([]); + expect(store.deleteCalls(['unknown-id'])).toEqual(['unknown-id']); + expect(store.callCount).toBe(1); + }); + }); + // --- Read/write isolation between event listeners --- describe('read/write isolation between event listeners', () => { diff --git a/libraries/typescript/tests/unit/metrics.test.ts b/libraries/typescript/tests/unit/metrics.test.ts index 1a9a86fa..9222e3a6 100644 --- a/libraries/typescript/tests/unit/metrics.test.ts +++ b/libraries/typescript/tests/unit/metrics.test.ts @@ -344,4 +344,64 @@ describe('CallMetricsAccumulator', () => { expect(metrics.cost.tts).toBeCloseTo(0.50, 4); // 1k chars * 0.50 }); }); + + // --------------------------------------------------------------------------- + // EOUMetrics field semantics + unit parity with the Python SDK. + // --------------------------------------------------------------------------- + + describe('emitEouMetrics field semantics', () => { + it('emits endOfUtteranceDelay and transcriptionDelay in ms with the documented convention', async () => { + const { EventBus } = await import('../../src/observability/event-bus'); + const acc = makeAccumulator(); + const bus = new EventBus(); + const emitted: unknown[] = []; + bus.on('eou_metrics', (m) => emitted.push(m)); + acc.attachEventBus(bus); + + // Deltas chosen so the field assignment is unambiguous: + // VAD stop -> STT final = 200 ms + // VAD stop -> turn committed = 350 ms + const tVad = 1_000_000; + acc.recordVadStop(tVad); + acc.recordSttFinalTimestamp(tVad + 200); + acc.recordOnUserTurnCompletedDelay(50); + // recordTurnCommitted() auto-invokes emitEouMetrics() once all three + // timestamps are present — no explicit emit needed. + acc.recordTurnCommitted(tVad + 350); + + expect(emitted).toHaveLength(1); + const m = emitted[0] as { + endOfUtteranceDelay: number; + transcriptionDelay: number; + onUserTurnCompletedDelay: number; + }; + expect(m.endOfUtteranceDelay).toBeCloseTo(200, 6); + expect(m.transcriptionDelay).toBeCloseTo(350, 6); + expect(m.onUserTurnCompletedDelay).toBeCloseTo(50, 6); + }); + + it('clamps negative deltas to zero', async () => { + const { EventBus } = await import('../../src/observability/event-bus'); + const acc = makeAccumulator(); + const bus = new EventBus(); + const emitted: unknown[] = []; + bus.on('eou_metrics', (m) => emitted.push(m)); + acc.attachEventBus(bus); + + const tVad = 1_000_500; + acc.recordVadStop(tVad); + acc.recordSttFinalTimestamp(tVad - 100); + acc.recordOnUserTurnCompletedDelay(0); + // recordTurnCommitted() auto-emits. + acc.recordTurnCommitted(tVad - 50); + + expect(emitted).toHaveLength(1); + const m = emitted[0] as { + endOfUtteranceDelay: number; + transcriptionDelay: number; + }; + expect(m.endOfUtteranceDelay).toBe(0); + expect(m.transcriptionDelay).toBe(0); + }); + }); }); diff --git a/libraries/typescript/tests/unit/observability-attributes.test.ts b/libraries/typescript/tests/unit/observability-attributes.test.ts new file mode 100644 index 00000000..f3b4bc25 --- /dev/null +++ b/libraries/typescript/tests/unit/observability-attributes.test.ts @@ -0,0 +1,77 @@ +/** + * Smoke tests for ``patter.*`` attribute helpers. + * + * The real OTel wire-up lives in optional peer deps (``@opentelemetry/*``). + * These tests confirm the public surface is callable without crashing when + * those deps are absent — matching the no-op-by-default contract shared + * with the Python helpers. + */ +import { describe, it, expect } from 'vitest'; + +import { + recordPatterAttrs, + patterCallScope, + attachSpanExporter, + DEFAULT_SIDE, +} from '../../src/observability/attributes'; + +describe('observability/attributes (no-op surface)', () => { + it('exports the expected helpers with the correct shapes', () => { + expect(typeof recordPatterAttrs).toBe('function'); + expect(typeof patterCallScope).toBe('function'); + expect(typeof attachSpanExporter).toBe('function'); + expect(DEFAULT_SIDE).toBe('uut'); + }); + + it('recordPatterAttrs is a no-op when tracing is disabled', () => { + // Tracing is disabled by default (PATTER_OTEL_ENABLED unset). + expect(() => { + recordPatterAttrs({ 'patter.cost.llm_usd': 0.001 }); + }).not.toThrow(); + }); + + it('patterCallScope rejects empty callId', async () => { + await expect( + patterCallScope({ callId: '' }, async () => 0), + ).rejects.toThrow(/callId/); + }); + + it('patterCallScope binds the scope around fn (default side)', async () => { + let observedDuringFn = false; + const value = await patterCallScope({ callId: 'c-1' }, async () => { + // The helper has no public reader, but ``recordPatterAttrs`` must + // remain a no-op inside the scope when tracing is disabled. + recordPatterAttrs({ 'patter.cost.llm_usd': 0 }); + observedDuringFn = true; + return 42; + }); + expect(value).toBe(42); + expect(observedDuringFn).toBe(true); + }); + + it('patterCallScope unwinds the stack on throw', async () => { + await expect( + patterCallScope({ callId: 'c-throw' }, async () => { + throw new Error('boom'); + }), + ).rejects.toThrow(/boom/); + // A second scope should still work — proves the stack was unwound. + const v = await patterCallScope({ callId: 'c-after' }, async () => 7); + expect(v).toBe(7); + }); + + it('attachSpanExporter stores _patterSide even when OTel is disabled', () => { + const fakePatter: { _patterSide?: string } = {}; + const fakeExporter = {}; + expect(() => + attachSpanExporter(fakePatter, fakeExporter, { side: 'driver' }), + ).not.toThrow(); + expect(fakePatter._patterSide).toBe('driver'); + }); + + it('attachSpanExporter defaults side to "uut"', () => { + const fakePatter: { _patterSide?: string } = {}; + attachSpanExporter(fakePatter, {}); + expect(fakePatter._patterSide).toBe(DEFAULT_SIDE); + }); +}); diff --git a/libraries/typescript/tests/unit/prewarm-handoff.test.ts b/libraries/typescript/tests/unit/prewarm-handoff.test.ts new file mode 100644 index 00000000..c16868ac --- /dev/null +++ b/libraries/typescript/tests/unit/prewarm-handoff.test.ts @@ -0,0 +1,212 @@ +/** + * Tests for the prewarm-handoff (FIX A) — keep parked WSs OPEN and adopt + * them at call connect, instead of close-and-reopen which doesn't warm + * TLS on Node `ws`. + * + * Coverage: + * 1. `Patter.parkProviderConnections` invokes `openParkedConnection` + * on the configured STT / TTS adapters. + * 2. The parked WS stays OPEN (readyState === OPEN) past the historic + * 250 ms idle window. + * 3. `popPrewarmedConnections` returns the parked handles and removes + * them from the cache (consume-once semantics). + * 4. `closePrewarmedConnections` (and `recordPrewarmWaste`) drains + * parked sockets cleanly. + * 5. A WS that died between park and adopt does NOT crash the consumer + * — the consumer falls back to fresh open. (Verified via the + * adapter-level `synthesizeStream` dropping a closed parked WS.) + * + * Tests use authentic real-code paths — only the upstream provider + * boundary is mocked. See `.claude/rules/authentic-tests.md`. + */ +import { describe, it, expect, vi, beforeEach } from 'vitest'; +import { Patter } from '../../src/client'; +import { Twilio } from '../../src/index'; +import type { AgentOptions } from '../../src/types'; +import type { STTAdapter, TTSAdapter, STTTranscriptCallback } from '../../src/provider-factory'; +import type { ElevenLabsParkedWS } from '../../src/providers/elevenlabs-ws-tts'; + +// Stub the EmbeddedServer so constructing a Patter doesn't spin up a +// real HTTP server. +vi.mock('../../src/server', async (importOriginal) => { + const orig = await importOriginal(); + class MockEmbeddedServer { + voicemailMessage = ''; + popPrewarmAudio: (id: string) => Buffer | undefined = () => undefined; + popPrewarmedConnections: (id: string) => unknown = () => undefined; + recordPrewarmWaste: (id: string) => void = () => undefined; + metricsStore = { recordCallInitiated: vi.fn() } as unknown as { + recordCallInitiated: (...args: unknown[]) => void; + }; + start = vi.fn().mockResolvedValue(undefined); + stop = vi.fn().mockResolvedValue(undefined); + constructor(..._args: unknown[]) {} + } + return { + ...orig, + EmbeddedServer: MockEmbeddedServer, + }; +}); + +// A minimal fake WS that exposes the readyState lifecycle but no +// network traffic. ws.OPEN === 1 by convention. +class FakeWS { + readyState = 1; // OPEN + closed = false; + close(): void { + this.readyState = 3; // CLOSED + this.closed = true; + } +} + +class StubSTTWithPark implements STTAdapter { + warmupCalls = 0; + parkCalls = 0; + adoptCalls = 0; + connectCalls = 0; + parkedWs: FakeWS | null = null; + async connect(): Promise { + this.connectCalls += 1; + } + sendAudio(_pcm: Buffer): void {} + onTranscript(_cb: STTTranscriptCallback): void {} + async close(): Promise {} + async warmup(): Promise { + this.warmupCalls += 1; + } + async openParkedConnection(): Promise { + this.parkCalls += 1; + this.parkedWs = new FakeWS(); + return this.parkedWs; + } + adoptWebSocket(_ws: unknown): void { + this.adoptCalls += 1; + } +} + +class StubTTSWithPark implements TTSAdapter { + warmupCalls = 0; + parkCalls = 0; + adoptCalls = 0; + parkedHandle: ElevenLabsParkedWS | null = null; + // eslint-disable-next-line require-yield + async *synthesizeStream(_text: string): AsyncGenerator { + return; + } + async warmup(): Promise { + this.warmupCalls += 1; + } + async openParkedConnection(): Promise { + this.parkCalls += 1; + this.parkedHandle = { ws: new FakeWS() as unknown as import('ws').WebSocket, bosSent: true }; + return this.parkedHandle; + } + adoptWebSocket(parked: ElevenLabsParkedWS): void { + this.adoptCalls += 1; + void parked; + } +} + +function makePatter(): Patter { + return new Patter({ + carrier: new Twilio({ + accountSid: 'ACtest000000000000000000000000000', + authToken: 'tok', + }), + phoneNumber: '+15551234567', + webhookUrl: 'example.test', + }); +} + +describe('[unit] prewarm-handoff', () => { + let phone: Patter; + beforeEach(() => { + phone = makePatter(); + }); + + it('parkProviderConnections invokes openParkedConnection on STT and TTS', async () => { + const stt = new StubSTTWithPark(); + const tts = new StubTTSWithPark(); + const agent: AgentOptions = { + systemPrompt: 'p', + provider: 'pipeline', + stt, + tts, + }; + // Private method — accessed via cast for the test only. + (phone as unknown as { parkProviderConnections: (a: AgentOptions, id: string) => void }) + .parkProviderConnections(agent, 'CAtest1'); + // Wait microtask + small delay for the async park tasks. + await new Promise((r) => setTimeout(r, 30)); + expect(stt.parkCalls).toBe(1); + expect(tts.parkCalls).toBe(1); + }); + + it('parked WS stays OPEN past the historic 250 ms idle window', async () => { + const stt = new StubSTTWithPark(); + const tts = new StubTTSWithPark(); + const agent: AgentOptions = { systemPrompt: 'p', provider: 'pipeline', stt, tts }; + (phone as unknown as { parkProviderConnections: (a: AgentOptions, id: string) => void }) + .parkProviderConnections(agent, 'CAtest2'); + await new Promise((r) => setTimeout(r, 350)); + expect(stt.parkedWs?.readyState).toBe(1); // OPEN + expect(tts.parkedHandle?.ws.readyState).toBe(1); + }); + + it('popPrewarmedConnections returns parked handles exactly once', async () => { + const stt = new StubSTTWithPark(); + const tts = new StubTTSWithPark(); + const agent: AgentOptions = { systemPrompt: 'p', provider: 'pipeline', stt, tts }; + (phone as unknown as { parkProviderConnections: (a: AgentOptions, id: string) => void }) + .parkProviderConnections(agent, 'CAtest3'); + await new Promise((r) => setTimeout(r, 30)); + const slot = phone.popPrewarmedConnections('CAtest3'); + expect(slot).toBeDefined(); + expect(slot?.stt).toBe(stt.parkedWs); + expect(slot?.tts).toBe(tts.parkedHandle); + // Second pop should be undefined — slot already drained. + expect(phone.popPrewarmedConnections('CAtest3')).toBeUndefined(); + }); + + it('closePrewarmedConnections closes parked sockets and drains the slot', async () => { + const stt = new StubSTTWithPark(); + const tts = new StubTTSWithPark(); + const agent: AgentOptions = { systemPrompt: 'p', provider: 'pipeline', stt, tts }; + (phone as unknown as { parkProviderConnections: (a: AgentOptions, id: string) => void }) + .parkProviderConnections(agent, 'CAtest4'); + await new Promise((r) => setTimeout(r, 30)); + expect(stt.parkedWs?.readyState).toBe(1); + phone.closePrewarmedConnections('CAtest4'); + expect(stt.parkedWs?.readyState).toBe(3); // CLOSED + expect(tts.parkedHandle?.ws.readyState).toBe(3); + // Slot drained. + expect(phone.popPrewarmedConnections('CAtest4')).toBeUndefined(); + }); + + it('recordPrewarmWaste also drains parked sockets (call ended pre-pickup)', async () => { + const stt = new StubSTTWithPark(); + const tts = new StubTTSWithPark(); + const agent: AgentOptions = { systemPrompt: 'p', provider: 'pipeline', stt, tts }; + (phone as unknown as { parkProviderConnections: (a: AgentOptions, id: string) => void }) + .parkProviderConnections(agent, 'CAtest5'); + await new Promise((r) => setTimeout(r, 30)); + phone.recordPrewarmWaste('CAtest5'); + expect(stt.parkedWs?.readyState).toBe(3); + expect(tts.parkedHandle?.ws.readyState).toBe(3); + }); + + it('does nothing when neither provider exposes openParkedConnection', () => { + // Adapters without the optional method must not allocate a slot. + const minimalStt: STTAdapter = { + async connect(): Promise {}, + sendAudio(): void {}, + onTranscript(): void {}, + async close(): Promise {}, + }; + const agent: AgentOptions = { systemPrompt: 'p', provider: 'pipeline', stt: minimalStt }; + (phone as unknown as { parkProviderConnections: (a: AgentOptions, id: string) => void }) + .parkProviderConnections(agent, 'CAtest6'); + // Slot was never created — pop returns undefined. + expect(phone.popPrewarmedConnections('CAtest6')).toBeUndefined(); + }); +}); diff --git a/libraries/typescript/tests/unit/prewarm.test.ts b/libraries/typescript/tests/unit/prewarm.test.ts new file mode 100644 index 00000000..5d5c8096 --- /dev/null +++ b/libraries/typescript/tests/unit/prewarm.test.ts @@ -0,0 +1,762 @@ +/** + * Tests for the prewarm and prewarmFirstMessage features. + * + * The feature wires three independent pieces together: + * + * 1. Provider ``warmup()`` methods on STT / TTS / LLM. Default = no-op. + * 2. ``Patter.call`` spawns provider warmup in parallel with the carrier + * ``initiateCall`` when ``agent.prewarm`` is true (the default). + * 3. ``Patter.call`` pre-renders ``agent.firstMessage`` to TTS bytes when + * ``agent.prewarmFirstMessage`` is true; the StreamHandler firstMessage + * emit consumes the cache instead of running TTS again. + * + * Tests use authentic real code paths — only the provider HTTPS-GET warmup + * boundary is mocked. See ``.claude/rules/authentic-tests.md``. + */ +import { describe, it, expect, vi, beforeEach, afterEach } from 'vitest'; +import { Patter } from '../../src/client'; +import { Twilio } from '../../src/index'; +import type { AgentOptions } from '../../src/types'; +import type { STTAdapter, TTSAdapter, STTTranscriptCallback } from '../../src/provider-factory'; + +// Stub the EmbeddedServer so constructing a Patter doesn't spin up a real HTTP server. +vi.mock('../../src/server', async (importOriginal) => { + const orig = await importOriginal(); + class MockEmbeddedServer { + voicemailMessage = ''; + popPrewarmAudio: (id: string) => Buffer | undefined = () => undefined; + start = vi.fn().mockResolvedValue(undefined); + stop = vi.fn().mockResolvedValue(undefined); + constructor(..._args: unknown[]) {} + } + return { + ...orig, + EmbeddedServer: MockEmbeddedServer, + }; +}); + +function makePatter(): Patter { + return new Patter({ + carrier: new Twilio({ accountSid: 'ACtest000000000000000000000000000', authToken: 'tok' }), + phoneNumber: '+15551234567', + webhookUrl: 'example.test', + }); +} + +class StubSTT implements STTAdapter { + warmupCalls = 0; + async connect(): Promise {} + sendAudio(_pcm: Buffer): void {} + onTranscript(_cb: STTTranscriptCallback): void {} + async close(): Promise {} + async warmup(): Promise { + this.warmupCalls += 1; + } +} + +class StubTTS implements TTSAdapter { + warmupCalls = 0; + synthesizeCalls = 0; + constructor(private readonly bytes: Buffer = Buffer.from('PCM_TTS_BYTES_OK')) {} + async *synthesizeStream(_text: string): AsyncGenerator { + this.synthesizeCalls += 1; + const half = Math.floor(this.bytes.byteLength / 2); + yield this.bytes.subarray(0, half); + yield this.bytes.subarray(half); + } + async warmup(): Promise { + this.warmupCalls += 1; + } +} + +class StubLLM { + warmupCalls = 0; + // eslint-disable-next-line require-yield + async *stream(): AsyncGenerator { + return; + } + async warmup(): Promise { + this.warmupCalls += 1; + } +} + +/** Drain prewarm tasks attached to the Patter instance. */ +async function drainPrewarmTasks(phone: Patter): Promise { + const internal = phone as unknown as { prewarmTasks: Set> }; + await Promise.allSettled(Array.from(internal.prewarmTasks)); +} + +describe('[unit] prewarm — Agent flag defaults', () => { + it('agent prewarm defaults are documented in types.ts (true / false)', () => { + // The Agent flag defaults are not enforced at the type level (they're + // optional), but client.ts treats undefined as the default. Verify + // ``agent.prewarm !== false`` (the actual gate) behaves as expected. + const agent: AgentOptions = { systemPrompt: 'hi' }; + expect(agent.prewarm).toBeUndefined(); + expect(agent.prewarmFirstMessage).toBeUndefined(); + // Default behaviour: prewarm is on unless user explicitly set false. + expect(agent.prewarm !== false).toBe(true); + expect(Boolean(agent.prewarmFirstMessage)).toBe(false); + }); +}); + +describe('[unit] prewarm — provider warmup', () => { + let phone: Patter; + beforeEach(() => { + phone = makePatter(); + }); + afterEach(async () => { + await drainPrewarmTasks(phone); + }); + + it('spawnProviderWarmup invokes warmup() on STT/TTS/LLM exactly once each', async () => { + const stt = new StubSTT(); + const tts = new StubTTS(); + const llm = new StubLLM(); + const agent: AgentOptions = { systemPrompt: 'hi', stt, tts, llm: llm as never }; + // eslint-disable-next-line @typescript-eslint/no-explicit-any + (phone as any).spawnProviderWarmup(agent); + await drainPrewarmTasks(phone); + expect(stt.warmupCalls).toBe(1); + expect(tts.warmupCalls).toBe(1); + expect(llm.warmupCalls).toBe(1); + }); + + it('skips warmup entirely when prewarm is false', async () => { + const stt = new StubSTT(); + const tts = new StubTTS(); + const llm = new StubLLM(); + const agent: AgentOptions = { + systemPrompt: 'hi', + stt, + tts, + llm: llm as never, + prewarm: false, + }; + if (agent.prewarm !== false) { + // eslint-disable-next-line @typescript-eslint/no-explicit-any + (phone as any).spawnProviderWarmup(agent); + } + await drainPrewarmTasks(phone); + expect(stt.warmupCalls).toBe(0); + expect(tts.warmupCalls).toBe(0); + expect(llm.warmupCalls).toBe(0); + }); + + it('a failing provider warmup is swallowed and never propagates', async () => { + class BoomTTS extends StubTTS { + override async warmup(): Promise { + throw new Error('DNS down'); + } + } + const stt = new StubSTT(); + const tts = new BoomTTS(); + const agent: AgentOptions = { systemPrompt: 'hi', stt, tts }; + // eslint-disable-next-line @typescript-eslint/no-explicit-any + (phone as any).spawnProviderWarmup(agent); + await drainPrewarmTasks(phone); + // STT still ran fine. + expect(stt.warmupCalls).toBe(1); + // No exception bled out — assertion is reaching here. + }); + + it('skips providers without a warmup method (older / minimal adapters)', async () => { + const noWarmupTTS: TTSAdapter = { + // eslint-disable-next-line require-yield + synthesizeStream: async function* () { + return; + }, + }; + const agent: AgentOptions = { systemPrompt: 'hi', tts: noWarmupTTS }; + // eslint-disable-next-line @typescript-eslint/no-explicit-any + (phone as any).spawnProviderWarmup(agent); + await drainPrewarmTasks(phone); + // No exception — that's the assertion. + }); +}); + +describe('[unit] prewarm — first-message cache', () => { + let phone: Patter; + beforeEach(() => { + phone = makePatter(); + }); + afterEach(async () => { + await drainPrewarmTasks(phone); + }); + + it('populates the cache when prewarmFirstMessage is true', async () => { + const tts = new StubTTS(Buffer.from('GREETING-AUDIO-BYTES')); + const agent: AgentOptions = { + systemPrompt: 'hi', + firstMessage: 'Hi there', + tts, + prewarmFirstMessage: true, + // ``spawnPrewarmFirstMessage`` is gated to ``provider === 'pipeline'`` — + // Realtime / ConvAI never consume the cache so we'd refuse to spawn. + provider: 'pipeline', + }; + // eslint-disable-next-line @typescript-eslint/no-explicit-any + (phone as any).spawnPrewarmFirstMessage(agent, 'CA-call-001', 5); + await drainPrewarmTasks(phone); + const buf = phone.popPrewarmAudio('CA-call-001'); + expect(buf?.toString()).toBe('GREETING-AUDIO-BYTES'); + expect(tts.synthesizeCalls).toBe(1); + }); + + it('skips the cache when prewarmFirstMessage is false (default)', async () => { + const tts = new StubTTS(Buffer.from('ZZZ')); + const agent: AgentOptions = { + systemPrompt: 'hi', + firstMessage: 'Hi there', + tts, + prewarmFirstMessage: false, + }; + // eslint-disable-next-line @typescript-eslint/no-explicit-any + (phone as any).spawnPrewarmFirstMessage(agent, 'CA-call-002', 5); + await drainPrewarmTasks(phone); + expect(phone.popPrewarmAudio('CA-call-002')).toBeUndefined(); + expect(tts.synthesizeCalls).toBe(0); + }); + + it('skips the cache when firstMessage is empty', async () => { + const tts = new StubTTS(); + const agent: AgentOptions = { + systemPrompt: 'hi', + firstMessage: '', + tts, + prewarmFirstMessage: true, + provider: 'pipeline', + }; + // eslint-disable-next-line @typescript-eslint/no-explicit-any + (phone as any).spawnPrewarmFirstMessage(agent, 'CA-call-003', 5); + await drainPrewarmTasks(phone); + expect(phone.popPrewarmAudio('CA-call-003')).toBeUndefined(); + expect(tts.synthesizeCalls).toBe(0); + }); + + it('popPrewarmAudio is one-shot (returns once, then undefined)', () => { + // eslint-disable-next-line @typescript-eslint/no-explicit-any + (phone as any).prewarmAudio.set('CA-X', Buffer.from('BYTES')); + expect(phone.popPrewarmAudio('CA-X')?.toString()).toBe('BYTES'); + expect(phone.popPrewarmAudio('CA-X')).toBeUndefined(); + }); + + it('logs WARN when prewarmed audio was paid for but never consumed', () => { + // eslint-disable-next-line @typescript-eslint/no-explicit-any + (phone as any).prewarmAudio.set('CA-waste', Buffer.from('WASTED')); + const warnings: string[] = []; + const orig = console.warn; + console.warn = (...args: unknown[]) => { + warnings.push(args.join(' ')); + }; + try { + // eslint-disable-next-line @typescript-eslint/no-explicit-any + (phone as any).recordPrewarmWaste('CA-waste'); + } finally { + console.warn = orig; + } + // The default logger writes to console.warn. Even if the SDK logger + // is configured differently in the test env, the entry must be gone. + // eslint-disable-next-line @typescript-eslint/no-explicit-any + expect((phone as any).prewarmAudio.has('CA-waste')).toBe(false); + }); +}); + +describe('[unit] prewarm — StreamHandler consumes cache', () => { + it('the deps include the popPrewarmAudio accessor when wired', () => { + // The deps interface guarantees the optional callback. The full + // StreamHandler test covers the cache-hit short-circuit; this test + // exists to verify the wiring point exists at the type level — if + // someone removes ``popPrewarmAudio`` from ``StreamHandlerDeps`` the + // compiler error here makes the regression obvious. + type WithPop = { + popPrewarmAudio?: (id: string) => Buffer | undefined; + }; + const deps: WithPop = { + popPrewarmAudio: (id) => (id === 'CA-1' ? Buffer.from('cached') : undefined), + }; + expect(deps.popPrewarmAudio?.('CA-1')?.toString()).toBe('cached'); + expect(deps.popPrewarmAudio?.('CA-2')).toBeUndefined(); + }); +}); + +// --------------------------------------------------------------------------- +// FIX #97 regression — prewarm bytes must be chunked, not single-shot +// --------------------------------------------------------------------------- + +describe('[unit] streamPrewarmBytes — chunked send for cancel granularity (FIX #97)', () => { + // Build a minimally-wired StreamHandler shell so we can drive the + // private ``streamPrewarmBytes`` method directly. The encodePipeline + // path is the one that actually produces bytes for the wire — we + // assert that bridge.sendAudio is called many times, not once. + it('chunks a 5-second prewarm buffer into many sendAudio calls', async () => { + const { StreamHandler } = await import('../../src/stream-handler'); + const { MetricsStore } = await import('../../src/dashboard/store'); + const { RemoteMessageHandler } = await import('../../src/remote-message'); + type WSWebSocket = import('ws').WebSocket; + + const sendAudio = vi.fn(); + let handlerRef: { onMark: (n: string) => Promise } | null = null; + // BUG #128: every chunk now pairs with a Twilio mark and the loop + // window-blocks until echoes arrive. Production Twilio echoes within + // 100-250 ms of playback; in this test we echo synchronously so the + // chunking assertion can complete inside the vitest timeout. + const sendMark = vi.fn((_ws: unknown, name: string) => { + if (handlerRef) void handlerRef.onMark(name); + }); + const bridge = { + label: 'TestBridge', + telephonyProvider: 'twilio', + sendAudio, + sendMark, + sendClear: vi.fn(), + transferCall: vi.fn().mockResolvedValue(undefined), + endCall: vi.fn().mockResolvedValue(undefined), + createStt: vi.fn().mockReturnValue(null), + queryTelephonyCost: vi.fn().mockResolvedValue(undefined), + }; + const ws = { + send: vi.fn(), + close: vi.fn(), + on: vi.fn(), + once: vi.fn(), + readyState: 1, + removeListener: vi.fn(), + addEventListener: vi.fn(), + removeEventListener: vi.fn(), + } as unknown as WSWebSocket; + const deps = { + config: { openaiKey: 'test-oai-key' }, + agent: { systemPrompt: 'Test', provider: 'pipeline' as const }, + bridge, + metricsStore: new MetricsStore(), + pricing: null, + remoteHandler: new RemoteMessageHandler(), + recording: false, + buildAIAdapter: vi.fn(), + sanitizeVariables: vi.fn((raw: Record) => { + const safe: Record = {}; + for (const [k, v] of Object.entries(raw)) safe[k] = String(v); + return safe; + }), + resolveVariables: vi.fn((tpl: string) => tpl), + }; + const h = new StreamHandler(deps, ws, '+1', '+2'); + interface Priv { + isSpeaking: boolean; + streamSid: string; + firstAudioSentAt: number | null; + streamPrewarmBytes: (bytes: Buffer) => Promise; + onMark: (n: string) => Promise; + } + const p = h as unknown as Priv; + p.isSpeaking = true; + p.streamSid = 'SM-test'; + p.firstAudioSentAt = Date.now(); // gate open + handlerRef = p; // hand the handler to the mark-echo mock above + + // 5 s of PCM16 @ 16 kHz mono = 5 * 16000 * 2 = 160_000 bytes. + const prewarmBytes = Buffer.alloc(160_000, 1); + expect(prewarmBytes.length).toBe(160_000); + + const firstChunkSent = await p.streamPrewarmBytes(prewarmBytes); + + expect(firstChunkSent).toBe(true); + // 160_000 / 1280 = 125 chunks. Anything ≥ 100 proves the buffer was + // split — we don't pin the exact count to keep the test robust to + // future chunk-size tweaks. + expect(sendAudio.mock.calls.length).toBeGreaterThanOrEqual(100); + // Definitely not the single-shot regression. + expect(sendAudio.mock.calls.length).toBeGreaterThan(1); + }); + + it('stops chunking when isSpeaking flips false mid-buffer (barge-in)', async () => { + const { StreamHandler } = await import('../../src/stream-handler'); + const { MetricsStore } = await import('../../src/dashboard/store'); + const { RemoteMessageHandler } = await import('../../src/remote-message'); + type WSWebSocket = import('ws').WebSocket; + + let chunksSeen = 0; + let handlerRef: { isSpeaking: boolean } | null = null; + const sendAudio = vi.fn(() => { + chunksSeen += 1; + // After the second chunk, simulate a barge-in flipping the gate. + if (chunksSeen === 2 && handlerRef) { + handlerRef.isSpeaking = false; + } + }); + const bridge = { + label: 'TestBridge', + telephonyProvider: 'twilio', + sendAudio, + sendMark: vi.fn(), + sendClear: vi.fn(), + transferCall: vi.fn().mockResolvedValue(undefined), + endCall: vi.fn().mockResolvedValue(undefined), + createStt: vi.fn().mockReturnValue(null), + queryTelephonyCost: vi.fn().mockResolvedValue(undefined), + }; + const ws = { + send: vi.fn(), + close: vi.fn(), + on: vi.fn(), + once: vi.fn(), + readyState: 1, + removeListener: vi.fn(), + addEventListener: vi.fn(), + removeEventListener: vi.fn(), + } as unknown as WSWebSocket; + const deps = { + config: { openaiKey: 'test-oai-key' }, + agent: { systemPrompt: 'Test', provider: 'pipeline' as const }, + bridge, + metricsStore: new MetricsStore(), + pricing: null, + remoteHandler: new RemoteMessageHandler(), + recording: false, + buildAIAdapter: vi.fn(), + sanitizeVariables: vi.fn((raw: Record) => { + const safe: Record = {}; + for (const [k, v] of Object.entries(raw)) safe[k] = String(v); + return safe; + }), + resolveVariables: vi.fn((tpl: string) => tpl), + }; + const h = new StreamHandler(deps, ws, '+1', '+2'); + interface Priv { + isSpeaking: boolean; + streamSid: string; + firstAudioSentAt: number | null; + streamPrewarmBytes: (bytes: Buffer) => Promise; + } + const p = h as unknown as Priv; + p.isSpeaking = true; + p.streamSid = 'SM-bargein'; + p.firstAudioSentAt = Date.now(); + handlerRef = p; + + const prewarmBytes = Buffer.alloc(160_000, 1); + await p.streamPrewarmBytes(prewarmBytes); + + // Exactly 2 chunks were sent; the loop broke before the third + // iteration could invoke sendAudio. + expect(sendAudio.mock.calls.length).toBe(2); + }); +}); + +// --------------------------------------------------------------------------- +// FIX #91 — cache eviction on abnormal hangup (idempotency + statusCallback) +// --------------------------------------------------------------------------- + +describe('[unit] prewarm — eviction on abnormal hangup (FIX #91)', () => { + let phone: Patter; + beforeEach(() => { + phone = makePatter(); + }); + afterEach(async () => { + await drainPrewarmTasks(phone); + }); + + it('recordPrewarmWaste is idempotent — second call does not double-WARN', () => { + // eslint-disable-next-line @typescript-eslint/no-explicit-any + (phone as any).prewarmAudio.set('CA-twice', Buffer.from('BYTES')); + const warnings: string[] = []; + const orig = console.warn; + console.warn = (...args: unknown[]) => warnings.push(args.join(' ')); + try { + phone.recordPrewarmWaste('CA-twice'); + phone.recordPrewarmWaste('CA-twice'); + } finally { + console.warn = orig; + } + // Filter out other unrelated console.warns (none expected here). + const wasteWarns = warnings.filter((w) => w.includes('CA-twice') && /wasted/i.test(w)); + expect(wasteWarns.length).toBe(1); + // eslint-disable-next-line @typescript-eslint/no-explicit-any + expect((phone as any).prewarmAudio.has('CA-twice')).toBe(false); + }); + + it('marks call_id as consumed even if there were no bytes cached (silent-evict guard)', () => { + phone.recordPrewarmWaste('CA-empty'); + // eslint-disable-next-line @typescript-eslint/no-explicit-any + expect((phone as any).prewarmConsumed.has('CA-empty')).toBe(true); + }); +}); + +// --------------------------------------------------------------------------- +// FIX #92 — race start-vs-prewarm task (orphan bytes guard) +// --------------------------------------------------------------------------- + +describe('[unit] prewarm — race orphan-bytes guard (FIX #92)', () => { + let phone: Patter; + beforeEach(() => { + phone = makePatter(); + }); + afterEach(async () => { + await drainPrewarmTasks(phone); + }); + + it('drops bytes when the consumer polled before the synth finished', async () => { + // SlowTTS that emits chunks across multiple awaits so we can poll + // pop_prewarm_audio before it finishes. + class SlowTTS implements TTSAdapter { + synthesizeCalls = 0; + async *synthesizeStream(_text: string): AsyncGenerator { + this.synthesizeCalls += 1; + await new Promise((r) => setTimeout(r, 200)); + yield Buffer.from('LATE-1'); + await new Promise((r) => setTimeout(r, 50)); + yield Buffer.from('LATE-2'); + } + } + const tts = new SlowTTS(); + const agent: AgentOptions = { + systemPrompt: 'hi', + firstMessage: 'Hi', + tts, + prewarmFirstMessage: true, + provider: 'pipeline', + }; + + // eslint-disable-next-line @typescript-eslint/no-explicit-any + (phone as any).spawnPrewarmFirstMessage(agent, 'CA-race', 5); + // Carrier ``start`` arrives BEFORE synth finishes — consumer polls. + await new Promise((r) => setTimeout(r, 50)); + const cached = phone.popPrewarmAudio('CA-race'); + expect(cached).toBeUndefined(); + + const warnings: string[] = []; + const orig = console.warn; + console.warn = (...args: unknown[]) => warnings.push(args.join(' ')); + try { + await drainPrewarmTasks(phone); + } finally { + console.warn = orig; + } + + // eslint-disable-next-line @typescript-eslint/no-explicit-any + expect((phone as any).prewarmAudio.has('CA-race')).toBe(false); + const orphanWarns = warnings.filter( + (w) => /orphaned/i.test(w) && w.includes('CA-race'), + ); + expect(orphanWarns.length).toBe(1); + }); + + it('popPrewarmAudio marks the call_id consumed on cache HIT', () => { + // eslint-disable-next-line @typescript-eslint/no-explicit-any + (phone as any).prewarmAudio.set('CA-hit', Buffer.from('BYTES')); + const out = phone.popPrewarmAudio('CA-hit'); + expect(out?.toString()).toBe('BYTES'); + // eslint-disable-next-line @typescript-eslint/no-explicit-any + expect((phone as any).prewarmConsumed.has('CA-hit')).toBe(true); + }); +}); + +// --------------------------------------------------------------------------- +// FIX #93 — disconnect() cancels in-flight tasks and clears cache +// --------------------------------------------------------------------------- + +describe('[unit] prewarm — disconnect cleanup (FIX #93)', () => { + it('clears prewarm cache + consumed set + ttl timers on disconnect', async () => { + const phone = makePatter(); + + class VerySlowTTS implements TTSAdapter { + synthesizeCalls = 0; + async *synthesizeStream(_text: string): AsyncGenerator { + this.synthesizeCalls += 1; + // 10 s synth — disconnect must not wait this long. + await new Promise((r) => setTimeout(r, 10_000)); + yield Buffer.from('never'); + } + } + const tts = new VerySlowTTS(); + const agent: AgentOptions = { + systemPrompt: 'hi', + firstMessage: 'hello', + tts, + prewarmFirstMessage: true, + provider: 'pipeline', + }; + // eslint-disable-next-line @typescript-eslint/no-explicit-any + (phone as any).spawnPrewarmFirstMessage(agent, 'CA-disco', 30); + // Pre-seed entries we expect to be cleared. + // eslint-disable-next-line @typescript-eslint/no-explicit-any + (phone as any).prewarmAudio.set('CA-leftover', Buffer.from('STALE')); + // eslint-disable-next-line @typescript-eslint/no-explicit-any + (phone as any).prewarmConsumed.add('CA-leftover'); + + // disconnect() should bound the wait at 1 s — well under the 10 s synth. + const t0 = Date.now(); + await phone.disconnect(); + const elapsed = Date.now() - t0; + expect(elapsed).toBeLessThan(2_000); + + // eslint-disable-next-line @typescript-eslint/no-explicit-any + expect((phone as any).prewarmAudio.size).toBe(0); + // eslint-disable-next-line @typescript-eslint/no-explicit-any + expect((phone as any).prewarmConsumed.size).toBe(0); + // eslint-disable-next-line @typescript-eslint/no-explicit-any + expect((phone as any).prewarmTasks.size).toBe(0); + // eslint-disable-next-line @typescript-eslint/no-explicit-any + expect((phone as any).prewarmTtlTimers.size).toBe(0); + }); +}); + +// --------------------------------------------------------------------------- +// FIX #94 — Realtime/ConvAI silently waste TTS spend +// --------------------------------------------------------------------------- + +describe('[unit] prewarm — provider-mode guard (FIX #94)', () => { + let phone: Patter; + beforeEach(() => { + phone = makePatter(); + }); + afterEach(async () => { + await drainPrewarmTasks(phone); + }); + + it('refuses to spawn for openai_realtime + WARN', async () => { + const tts = new StubTTS(); + const agent: AgentOptions = { + systemPrompt: 'hi', + firstMessage: 'hi', + tts, + prewarmFirstMessage: true, + provider: 'openai_realtime', + }; + const warnings: string[] = []; + const orig = console.warn; + console.warn = (...args: unknown[]) => warnings.push(args.join(' ')); + try { + // eslint-disable-next-line @typescript-eslint/no-explicit-any + (phone as any).spawnPrewarmFirstMessage(agent, 'CA-realtime', 5); + await drainPrewarmTasks(phone); + } finally { + console.warn = orig; + } + expect(tts.synthesizeCalls).toBe(0); + expect(phone.popPrewarmAudio('CA-realtime')).toBeUndefined(); + const guardWarns = warnings.filter((w) => /only supported in pipeline/i.test(w)); + expect(guardWarns.length).toBe(1); + }); + + it('refuses to spawn for elevenlabs_convai + WARN', async () => { + const tts = new StubTTS(); + const agent: AgentOptions = { + systemPrompt: 'hi', + firstMessage: 'hi', + tts, + prewarmFirstMessage: true, + provider: 'elevenlabs_convai', + }; + // eslint-disable-next-line @typescript-eslint/no-explicit-any + (phone as any).spawnPrewarmFirstMessage(agent, 'CA-convai', 5); + await drainPrewarmTasks(phone); + expect(tts.synthesizeCalls).toBe(0); + expect(phone.popPrewarmAudio('CA-convai')).toBeUndefined(); + }); +}); + +// --------------------------------------------------------------------------- +// FIX #96 — bounded cache (size cap + TTL eviction) +// --------------------------------------------------------------------------- + +describe('[unit] prewarm — bounded cache (FIX #96)', () => { + it('refuses spawn when in-flight count reaches PREWARM_CACHE_MAX', async () => { + const { PREWARM_CACHE_MAX } = await import('../../src/client'); + const phone = makePatter(); + // Fill to the cap. + for (let i = 0; i < PREWARM_CACHE_MAX; i += 1) { + // eslint-disable-next-line @typescript-eslint/no-explicit-any + (phone as any).prewarmAudio.set(`CA-fill-${i}`, Buffer.from('X')); + } + const tts = new StubTTS(); + const agent: AgentOptions = { + systemPrompt: 'hi', + firstMessage: 'hi', + tts, + prewarmFirstMessage: true, + provider: 'pipeline', + }; + const warnings: string[] = []; + const orig = console.warn; + console.warn = (...args: unknown[]) => warnings.push(args.join(' ')); + try { + // eslint-disable-next-line @typescript-eslint/no-explicit-any + (phone as any).spawnPrewarmFirstMessage(agent, 'CA-overflow', 5); + await drainPrewarmTasks(phone); + } finally { + console.warn = orig; + } + expect(tts.synthesizeCalls).toBe(0); + expect(phone.popPrewarmAudio('CA-overflow')).toBeUndefined(); + const fullWarns = warnings.filter( + (w) => /cache full/i.test(w) && w.includes('CA-overflow'), + ); + expect(fullWarns.length).toBe(1); + }); + + it('TTL evicts a never-consumed entry after ringTimeout + grace', async () => { + // Re-import the client module fresh so we can override the grace + // constant for this test. The constant is exported as a `const` + // binding so we can monkey-patch via the namespace import. + const phone = makePatter(); + const tts = new StubTTS(Buffer.from('TTL-BYTES')); + const agent: AgentOptions = { + systemPrompt: 'hi', + firstMessage: 'hi', + tts, + prewarmFirstMessage: true, + provider: 'pipeline', + }; + // ringTimeout = 0.1 s and rely on TTL = ringTimeout + 5 s default. + // Use vi.useFakeTimers so we don't actually wait 5 s. + vi.useFakeTimers(); + try { + // eslint-disable-next-line @typescript-eslint/no-explicit-any + (phone as any).spawnPrewarmFirstMessage(agent, 'CA-ttl', 0.1); + // Allow the synth task to complete (microtask drain + 0 ms timers). + await vi.advanceTimersByTimeAsync(50); + // eslint-disable-next-line @typescript-eslint/no-explicit-any + const cacheHasEntry = (phone as any).prewarmAudio.has('CA-ttl'); + expect(cacheHasEntry).toBe(true); + + const warnings: string[] = []; + const orig = console.warn; + console.warn = (...args: unknown[]) => warnings.push(args.join(' ')); + try { + // 100 ms ring + 5_000 ms grace = 5_100 ms total. + await vi.advanceTimersByTimeAsync(5_200); + } finally { + console.warn = orig; + } + // eslint-disable-next-line @typescript-eslint/no-explicit-any + expect((phone as any).prewarmAudio.has('CA-ttl')).toBe(false); + const ttlWarns = warnings.filter((w) => /ttl/i.test(w) && w.includes('CA-ttl')); + expect(ttlWarns.length).toBe(1); + } finally { + vi.useRealTimers(); + } + }); + + it('TTL is cancelled on normal cache consumption', async () => { + const phone = makePatter(); + const tts = new StubTTS(Buffer.from('NORMAL-BYTES')); + const agent: AgentOptions = { + systemPrompt: 'hi', + firstMessage: 'hi', + tts, + prewarmFirstMessage: true, + provider: 'pipeline', + }; + // eslint-disable-next-line @typescript-eslint/no-explicit-any + (phone as any).spawnPrewarmFirstMessage(agent, 'CA-normal', 1); + await drainPrewarmTasks(phone); + // eslint-disable-next-line @typescript-eslint/no-explicit-any + expect((phone as any).prewarmTtlTimers.has('CA-normal')).toBe(true); + const out = phone.popPrewarmAudio('CA-normal'); + expect(out?.toString()).toBe('NORMAL-BYTES'); + // eslint-disable-next-line @typescript-eslint/no-explicit-any + expect((phone as any).prewarmTtlTimers.has('CA-normal')).toBe(false); + }); +}); diff --git a/libraries/typescript/tests/unit/provider-warmup.mocked.test.ts b/libraries/typescript/tests/unit/provider-warmup.mocked.test.ts new file mode 100644 index 00000000..5a685870 --- /dev/null +++ b/libraries/typescript/tests/unit/provider-warmup.mocked.test.ts @@ -0,0 +1,541 @@ +/** + * Unit tests for the concrete provider WebSocket / HTTP warmup overrides. + * + * Covers the per-provider `warmup()` overrides on top of the no-op default + * declared on `STTAdapter` / `TTSAdapter`. Each test checks two invariants: + * + * 1. `warmup()` completes without throwing (best-effort contract). + * 2. When a provider opens a connection, it does NOT request any + * synthesis or send any audio frames — billing-during-warmup must + * remain zero per the per-provider docstrings. + * + * Tests use authentic real code paths — only the network boundary is + * mocked. See `.claude/rules/authentic-tests.md`. + */ +import { describe, it, expect, vi, beforeEach, afterEach } from 'vitest'; + +// ---------------------------------------------------------------------- +// Shared FakeWebSocket — mirrors the pattern used by other ws-mocking +// tests in the suite. Defined inside `vi.mock` so vitest's hoisting +// doesn't pull in an outer import. +// ---------------------------------------------------------------------- +vi.mock('ws', () => { + // eslint-disable-next-line @typescript-eslint/no-require-imports + const { EventEmitter } = require('events'); + class FakeWebSocket extends EventEmitter { + static OPEN = 1; + static CONNECTING = 0; + static CLOSED = 3; + readyState: number = FakeWebSocket.CONNECTING; + sent: unknown[] = []; + closeCalled = 0; + /** + * Pre-canned recv responses. Each `send()` call (or each open) pops one + * off. Tests that need the fake to talk back set this before the + * fake is constructed via FakeWebSocket.nextResponses. + */ + static nextResponses: string[] = []; + constructor(public url: string, public opts?: unknown) { + super(); + (FakeWebSocket as unknown as { instances: FakeWebSocket[] }).instances.push(this); + const queued = FakeWebSocket.nextResponses.slice(); + FakeWebSocket.nextResponses = []; + setImmediate(() => { + this.readyState = FakeWebSocket.OPEN; + this.emit('open'); + // Emit each queued message on its OWN macrotask so consumer code + // that attaches a fresh `ws.on('message', ...)` between frames + // (typical of multi-step setup paths) still observes every frame. + const drainOne = (idx: number): void => { + if (idx >= queued.length) return; + setImmediate(() => { + this.emit('message', Buffer.from(queued[idx])); + drainOne(idx + 1); + }); + }; + drainOne(0); + }); + } + send(data: unknown): void { + this.sent.push(data); + } + close(): void { + this.closeCalled += 1; + this.readyState = FakeWebSocket.CLOSED; + this.emit('close', 1000, Buffer.from('')); + } + off(event: string, fn: (...args: unknown[]) => void): this { + this.removeListener(event, fn); + return this; + } + } + (FakeWebSocket as unknown as { instances: FakeWebSocket[] }).instances = []; + return { default: FakeWebSocket }; +}); + +// Imports — must come AFTER the vi.mock above. +import { DeepgramSTT } from '../../src/providers/deepgram-stt'; +import { CartesiaSTT } from '../../src/providers/cartesia-stt'; +import { AssemblyAISTT } from '../../src/providers/assemblyai-stt'; +import { ElevenLabsWebSocketTTS } from '../../src/providers/elevenlabs-ws-tts'; +import { CartesiaTTS } from '../../src/providers/cartesia-tts'; +import { InworldTTS } from '../../src/providers/inworld-tts'; +import { OpenAIRealtimeAdapter } from '../../src/providers/openai-realtime'; +import { setLogger, getLogger, type Logger } from '../../src/logger'; +import WebSocketDefault from 'ws'; + +interface FakeWSInstance { + url: string; + sent: unknown[]; + closeCalled: number; + readyState: number; + emit: (event: string, ...args: unknown[]) => void; + removeAllListeners: () => void; +} + +interface FakeWSStatic { + instances: FakeWSInstance[]; + nextResponses: string[]; + OPEN: number; +} + +const FakeWS = WebSocketDefault as unknown as FakeWSStatic; + +beforeEach(() => { + FakeWS.instances.length = 0; + FakeWS.nextResponses = []; +}); + +afterEach(() => { + vi.restoreAllMocks(); +}); + +// ---------------------------------------------------------------------- +// Deepgram STT WS warmup +// ---------------------------------------------------------------------- + +describe('[mocked] DeepgramSTT.warmup', () => { + it('opens the WS, idles, closes — no audio frames sent', async () => { + const stt = new DeepgramSTT('dg-key'); + await stt.warmup(); + expect(FakeWS.instances).toHaveLength(1); + const ws = FakeWS.instances[0]; + expect(ws.closeCalled).toBeGreaterThan(0); + // No audio (Buffer) frames sent during warmup. + for (const sent of ws.sent) { + expect(Buffer.isBuffer(sent)).toBe(false); + } + }); + + it('targets the Deepgram listen endpoint', async () => { + const stt = new DeepgramSTT('dg-key'); + await stt.warmup(); + expect(FakeWS.instances[0].url).toContain('api.deepgram.com/v1/listen'); + }); + + it('swallows connect errors (does not throw)', async () => { + // Force the first emitted event to be `error` instead of `open`. + const FakeWSCtor = WebSocketDefault as unknown as new (...a: unknown[]) => FakeWSInstance & { + emit: (event: string, ...args: unknown[]) => void; + }; + const origDescriptor = Object.getOwnPropertyDescriptor(FakeWSCtor, 'OPEN'); + void origDescriptor; + const stt = new DeepgramSTT('dg-key'); + // Patch fetch path doesn't apply — this provider goes through ws. + // Instead, set the next constructed FakeWS to immediately emit error. + // Easiest: monkey-patch the static so the next instance fires `error` + // instead of `open`. + const origInstancesPush = FakeWS.instances.push.bind(FakeWS.instances); + FakeWS.instances.push = ((inst: FakeWSInstance) => { + const r = origInstancesPush(inst); + setImmediate(() => inst.emit('error', new Error('DNS down'))); + return r; + }) as typeof FakeWS.instances.push; + try { + // Must not throw. + await stt.warmup(); + } finally { + FakeWS.instances.push = origInstancesPush; + } + }); +}); + +// ---------------------------------------------------------------------- +// Cartesia STT WS warmup +// ---------------------------------------------------------------------- + +describe('[mocked] CartesiaSTT.warmup', () => { + it('opens the WS, idles, closes — no audio frames sent', async () => { + const stt = new CartesiaSTT('cart-key'); + await stt.warmup(); + expect(FakeWS.instances).toHaveLength(1); + const ws = FakeWS.instances[0]; + expect(ws.closeCalled).toBeGreaterThan(0); + for (const sent of ws.sent) { + expect(Buffer.isBuffer(sent)).toBe(false); + } + }); + + it('targets the Cartesia STT websocket endpoint', async () => { + const stt = new CartesiaSTT('cart-key'); + await stt.warmup(); + expect(FakeWS.instances[0].url).toContain('/stt/websocket'); + }); + + it('handshake error does not leak the API key into logs (regression)', async () => { + // Cartesia auth uses ?api_key=... in the URL. A 401/403 from the + // server during the WS upgrade surfaces via the `error` event with + // a message that may include the URL. The warmup catch handler + // must extract only the HTTP status, never the full message. + const secretKey = 'ck_secret_THIS_MUST_NEVER_LEAK'; + + const captured: { level: string; message: string }[] = []; + const originalLogger = getLogger(); + const captureLogger: Logger = { + info: (msg) => captured.push({ level: 'info', message: msg }), + warn: (msg) => captured.push({ level: 'warn', message: msg }), + error: (msg) => captured.push({ level: 'error', message: msg }), + debug: (msg) => captured.push({ level: 'debug', message: msg }), + }; + setLogger(captureLogger); + + // Force the next FakeWS instance to fire `error` with a payload + // shaped like a `ws` handshake failure — `statusCode` set to 401 + // and a message that includes the secret URL (matching the real + // `ws` behaviour pre-fix). + const origInstancesPush = FakeWS.instances.push.bind(FakeWS.instances); + FakeWS.instances.push = ((inst: FakeWSInstance) => { + const r = origInstancesPush(inst); + setImmediate(() => { + const err = new Error( + `Unexpected server response: 401 (url=${inst.url})`, + ) as Error & { statusCode?: number }; + err.statusCode = 401; + inst.emit('error', err); + }); + return r; + }) as typeof FakeWS.instances.push; + + try { + const stt = new CartesiaSTT(secretKey); + await stt.warmup(); // must not throw + } finally { + FakeWS.instances.push = origInstancesPush; + setLogger(originalLogger); + } + + // The API key must not appear in any captured log message. + for (const log of captured) { + expect(log.message).not.toContain(secretKey); + expect(log.message).not.toContain('api_key='); + } + // We should still log SOMETHING — namely the HTTP status — so + // operators know the warmup failed and why. + expect( + captured.some((l) => l.message.includes('401')), + ).toBe(true); + }); +}); + +// ---------------------------------------------------------------------- +// AssemblyAI STT WS warmup +// ---------------------------------------------------------------------- + +describe('[mocked] AssemblyAISTT.warmup', () => { + it('opens the WS, idles, sends Terminate (no audio), closes', async () => { + const stt = new AssemblyAISTT('aai-key'); + await stt.warmup(); + expect(FakeWS.instances).toHaveLength(1); + const ws = FakeWS.instances[0]; + expect(ws.closeCalled).toBeGreaterThan(0); + // No audio frames during warmup. + for (const sent of ws.sent) { + expect(Buffer.isBuffer(sent)).toBe(false); + } + // Terminate is fine — control message, not audio. If the warmup + // sent any string frame at all, it must be a Terminate. + for (const sent of ws.sent) { + const parsed = JSON.parse(String(sent)); + expect(parsed.type).toBe('Terminate'); + } + }); + + it('targets the AssemblyAI v3 ws endpoint', async () => { + const stt = new AssemblyAISTT('aai-key'); + await stt.warmup(); + expect(FakeWS.instances[0].url).toContain('/v3/ws'); + }); + + it('handshake error does not leak the API key into logs (regression)', async () => { + // AssemblyAI auth supports ?token=... in the URL when + // useQueryToken is set. A 401/403 from the server surfaces via + // the `error` event with a message that may include the URL. + const secretKey = 'aai_secret_THIS_MUST_NEVER_LEAK'; + + const captured: { level: string; message: string }[] = []; + const originalLogger = getLogger(); + const captureLogger: Logger = { + info: (msg) => captured.push({ level: 'info', message: msg }), + warn: (msg) => captured.push({ level: 'warn', message: msg }), + error: (msg) => captured.push({ level: 'error', message: msg }), + debug: (msg) => captured.push({ level: 'debug', message: msg }), + }; + setLogger(captureLogger); + + const origInstancesPush = FakeWS.instances.push.bind(FakeWS.instances); + FakeWS.instances.push = ((inst: FakeWSInstance) => { + const r = origInstancesPush(inst); + setImmediate(() => { + const err = new Error( + `Unexpected server response: 401 (url=${inst.url})`, + ) as Error & { statusCode?: number }; + err.statusCode = 401; + inst.emit('error', err); + }); + return r; + }) as typeof FakeWS.instances.push; + + try { + const stt = new AssemblyAISTT(secretKey, { useQueryToken: true }); + await stt.warmup(); // must not throw + } finally { + FakeWS.instances.push = origInstancesPush; + setLogger(originalLogger); + } + + for (const log of captured) { + expect(log.message).not.toContain(secretKey); + expect(log.message).not.toContain('token='); + } + expect( + captured.some((l) => l.message.includes('401')), + ).toBe(true); + }); +}); + +// ---------------------------------------------------------------------- +// ElevenLabs WS TTS warmup +// ---------------------------------------------------------------------- + +describe('[mocked] ElevenLabsWebSocketTTS.warmup', () => { + it('opens the WS, sends keepalive, idles, closes — no synthesis commit', async () => { + const tts = new ElevenLabsWebSocketTTS({ apiKey: 'el-key' }); + await tts.warmup(); + expect(FakeWS.instances).toHaveLength(1); + const ws = FakeWS.instances[0]; + expect(ws.closeCalled).toBeGreaterThan(0); + // Every send must be a keepalive ({"text": " "}) — no flush:true and + // no real text (which would commit a synthesis and bill characters). + for (const sent of ws.sent) { + const msg = JSON.parse(String(sent)); + expect(msg.flush).not.toBe(true); + expect(String(msg.text || '').trim()).toBe(''); + } + }); + + it('targets the ElevenLabs stream-input endpoint', async () => { + const tts = new ElevenLabsWebSocketTTS({ apiKey: 'el-key' }); + await tts.warmup(); + expect(FakeWS.instances[0].url).toContain('/stream-input'); + }); + + it('warmup BOS bytes are byte-identical to synthesizeStream BOS bytes (regression)', async () => { + // Configure with non-default voice_settings + auto_mode=false + + // chunk_length_schedule so the BOS frame carries every optional field. + const tts = new ElevenLabsWebSocketTTS({ + apiKey: 'el-key', + voiceSettings: { stability: 0.7, similarity_boost: 0.8 }, + autoMode: false, + chunkLengthSchedule: [120, 160, 250, 290], + }); + + // --- Capture warmup BOS --- + await tts.warmup(); + const warmupWs = FakeWS.instances[0]; + const warmupBos = warmupWs.sent[0] as string; + expect(typeof warmupBos).toBe('string'); + + // Reset for the synthesize run. + FakeWS.instances.length = 0; + + // --- Capture synthesize BOS --- + // Pre-canned response: an `isFinal` frame so the generator exits fast + // without yielding any real audio bytes. + FakeWS.nextResponses = [JSON.stringify({ isFinal: true })]; + const gen = tts.synthesizeStream('hello'); + // Drain the generator until it exits. + for await (const _chunk of gen) { + void _chunk; + } + const synthWs = FakeWS.instances[0]; + const synthBos = synthWs.sent[0] as string; + expect(typeof synthBos).toBe('string'); + + // BOS frames must match byte-for-byte so ElevenLabs picks the same + // per-session worker for warm and live. + expect(Buffer.from(warmupBos, 'utf8').equals(Buffer.from(synthBos, 'utf8'))).toBe(true); + + // And specifically: must NOT include flush:true. + const parsed = JSON.parse(warmupBos); + expect(parsed.flush).not.toBe(true); + expect(String(parsed.text || '').trim()).toBe(''); + }); +}); + +// ---------------------------------------------------------------------- +// Cartesia TTS HTTP warmup +// ---------------------------------------------------------------------- + +describe('[mocked] CartesiaTTS.warmup', () => { + it('issues a GET against /voices, never POST /tts/bytes', async () => { + const calls: { url: string; method: string }[] = []; + const fetchSpy = vi.spyOn(globalThis, 'fetch').mockImplementation( + async (input: RequestInfo | URL, init?: RequestInit) => { + calls.push({ + url: typeof input === 'string' ? input : input.toString(), + method: init?.method ?? 'GET', + }); + return new Response('', { status: 200 }); + }, + ); + try { + const tts = new CartesiaTTS('ct-key'); + await tts.warmup(); + } finally { + fetchSpy.mockRestore(); + } + expect(calls).toHaveLength(1); + expect(calls[0].url).toContain('/voices'); + expect(calls[0].method).toBe('GET'); + }); + + it('swallows fetch errors (does not throw)', async () => { + const fetchSpy = vi.spyOn(globalThis, 'fetch').mockImplementation(async () => { + throw new Error('DNS down'); + }); + try { + const tts = new CartesiaTTS('ct-key'); + // Must not throw. + await tts.warmup(); + } finally { + fetchSpy.mockRestore(); + } + }); +}); + +// ---------------------------------------------------------------------- +// Inworld TTS HTTP warmup +// ---------------------------------------------------------------------- + +describe('[mocked] InworldTTS.warmup', () => { + it('issues GET /tts/v1/voices — 2xx, not HEAD/POST against the POST-only streaming endpoint', async () => { + // Earlier revisions used HEAD against the streaming endpoint, + // which returned 405. New path uses the documented voices + // metadata GET so the response is 2xx and no 405s are spammed. + const calls: { url: string; method: string; status: number }[] = []; + const fetchSpy = vi.spyOn(globalThis, 'fetch').mockImplementation( + async (input: RequestInfo | URL, init?: RequestInit) => { + const url = typeof input === 'string' ? input : input.toString(); + const method = init?.method ?? 'GET'; + const response = new Response('', { status: 200 }); + calls.push({ url, method, status: response.status }); + return response; + }, + ); + try { + const tts = new InworldTTS('inworld-token'); + await tts.warmup(); + } finally { + fetchSpy.mockRestore(); + } + expect(calls).toHaveLength(1); + expect(calls[0].method).toBe('GET'); + expect(calls[0].url).toContain('/tts/v1/voices'); + // Must NOT target the POST-only streaming endpoint. + expect(calls[0].url).not.toContain('voice:stream'); + // Status must be 2xx (the fake responds with 200) — no 405 spam. + expect(calls[0].status).toBeGreaterThanOrEqual(200); + expect(calls[0].status).toBeLessThan(300); + }); + + it('swallows fetch errors (does not throw)', async () => { + const fetchSpy = vi.spyOn(globalThis, 'fetch').mockImplementation(async () => { + throw new Error('DNS down'); + }); + try { + const tts = new InworldTTS('inworld-token'); + await tts.warmup(); + } finally { + fetchSpy.mockRestore(); + } + }); +}); + +// ---------------------------------------------------------------------- +// OpenAI Realtime warmup (session.update — billing-safe, no response.create) +// ---------------------------------------------------------------------- + +describe('[mocked] OpenAIRealtimeAdapter.warmup', () => { + it('sends session.update only — never response.create — and waits for session.updated', async () => { + // Pre-canned server frames: session.created → session.updated. + FakeWS.nextResponses = [ + JSON.stringify({ type: 'session.created' }), + JSON.stringify({ type: 'session.updated' }), + ]; + + const adapter = new OpenAIRealtimeAdapter( + 'sk-test', + 'gpt-realtime-mini', + 'alloy', + 'You are a test assistant.', + ); + await adapter.warmup(); + + expect(FakeWS.instances).toHaveLength(1); + const ws = FakeWS.instances[0]; + expect(ws.closeCalled).toBeGreaterThan(0); + + const sentMessages = (ws.sent as unknown[]).map((s) => JSON.parse(String(s))); + // Must NOT send response.create — that field is not in the OpenAI + // Realtime schema and is billing-unsafe. + expect(sentMessages.find((m) => m.type === 'response.create')).toBeUndefined(); + // Must NOT send audio. + expect(sentMessages.find((m) => m.type === 'input_audio_buffer.append')).toBeUndefined(); + // Must send exactly one session.update with the production fields. + const updates = sentMessages.filter((m) => m.type === 'session.update'); + expect(updates).toHaveLength(1); + const session = updates[0].session; + for (const required of [ + 'input_audio_format', + 'output_audio_format', + 'voice', + 'instructions', + 'turn_detection', + 'input_audio_transcription', + ]) { + expect(session).toHaveProperty(required); + } + expect(session.voice).toBe('alloy'); + expect(session.instructions).toBe('You are a test assistant.'); + }); + + it('does not send response.create on the wire (regression)', async () => { + FakeWS.nextResponses = [ + JSON.stringify({ type: 'session.created' }), + JSON.stringify({ type: 'session.updated' }), + ]; + const adapter = new OpenAIRealtimeAdapter('sk-test'); + await adapter.warmup(); + const ws = FakeWS.instances[0]; + for (const raw of ws.sent as unknown[]) { + expect(String(raw)).not.toContain('response.create'); + } + }); + + it('targets the OpenAI Realtime endpoint', async () => { + FakeWS.nextResponses = [JSON.stringify({ type: 'session.created' })]; + const adapter = new OpenAIRealtimeAdapter('sk-test'); + await adapter.warmup(); + expect(FakeWS.instances[0].url).toContain('api.openai.com/v1/realtime'); + }); +}); diff --git a/libraries/typescript/tests/unit/server-routes.test.ts b/libraries/typescript/tests/unit/server-routes.test.ts index 9f940b72..772dd972 100644 --- a/libraries/typescript/tests/unit/server-routes.test.ts +++ b/libraries/typescript/tests/unit/server-routes.test.ts @@ -633,4 +633,97 @@ describe('EmbeddedServer route behavior', () => { spy.mockRestore(); } }); + + // ------------------------------------------------------------------------- + // FIX #91 — twilio /status invokes recordPrewarmWaste on no-answer/busy/... + // ------------------------------------------------------------------------- + + it('twilio /status invokes recordPrewarmWaste on abnormal CallStatus', async () => { + const server = new EmbeddedServer( + makeConfig({ twilioToken: '' }), + makeAgent(), + undefined, undefined, undefined, undefined, false, '', undefined, undefined, false, + ); + + const wasteCalls: string[] = []; + server.recordPrewarmWaste = (callId: string) => { + wasteCalls.push(callId); + }; + + const port = 19000 + Math.floor(Math.random() * 1000); + await server.start(port); + + try { + for (const status of ['no-answer', 'busy', 'failed', 'canceled']) { + const sid = `CA_${status.replace('-', '_')}_001`; + const resp = await fetch(`http://127.0.0.1:${port}/webhooks/twilio/status`, { + method: 'POST', + headers: { 'Content-Type': 'application/x-www-form-urlencoded' }, + body: new URLSearchParams({ CallSid: sid, CallStatus: status }).toString(), + }); + expect(resp.status).toBe(204); + } + expect(wasteCalls).toEqual([ + 'CA_no_answer_001', + 'CA_busy_001', + 'CA_failed_001', + 'CA_canceled_001', + ]); + + // ``completed`` is normal — must NOT trigger eviction. + wasteCalls.length = 0; + const resp = await fetch(`http://127.0.0.1:${port}/webhooks/twilio/status`, { + method: 'POST', + headers: { 'Content-Type': 'application/x-www-form-urlencoded' }, + body: new URLSearchParams({ CallSid: 'CA_done_001', CallStatus: 'completed' }).toString(), + }); + expect(resp.status).toBe(204); + expect(wasteCalls).toEqual([]); + } finally { + await server.stop(); + } + }); + + // ------------------------------------------------------------------------- + // FIX #91 — telnyx /voice on call.hangup invokes recordPrewarmWaste + // ------------------------------------------------------------------------- + + it('telnyx call.hangup invokes recordPrewarmWaste', async () => { + const server = new EmbeddedServer( + makeConfig({ + telephonyProvider: 'telnyx', + telnyxKey: 'KEY_test', + telnyxConnectionId: 'conn_test', + }), + makeAgent(), + undefined, undefined, undefined, undefined, false, '', undefined, undefined, false, + ); + const wasteCalls: string[] = []; + server.recordPrewarmWaste = (callId: string) => { + wasteCalls.push(callId); + }; + + const port = 19000 + Math.floor(Math.random() * 1000); + await server.start(port); + + try { + const resp = await fetch(`http://127.0.0.1:${port}/webhooks/telnyx/voice`, { + method: 'POST', + headers: { 'Content-Type': 'application/json' }, + body: JSON.stringify({ + data: { + event_type: 'call.hangup', + payload: { + call_control_id: 'ctrl-hung-up', + hangup_cause: 'no_answer', + }, + }, + }), + }); + expect(resp.status).toBe(200); + expect(wasteCalls).toEqual(['ctrl-hung-up']); + } finally { + await server.stop(); + } + }); }); diff --git a/libraries/typescript/tests/unit/silero-vad.test.ts b/libraries/typescript/tests/unit/silero-vad.test.ts index 0f340815..130362f9 100644 --- a/libraries/typescript/tests/unit/silero-vad.test.ts +++ b/libraries/typescript/tests/unit/silero-vad.test.ts @@ -173,4 +173,25 @@ describe('SileroVAD', () => { await vad.close(); await expect(vad.processFrame(silencePcm(512), 16000)).rejects.toThrow(/closed/); }); + + it('reset() returns VAD to SILENCE so the next utterance fires speech_start', async () => { + // Drive into SPEECH state, then reset, then drive INTO SPEECH again. Without + // reset() the second batch would not emit a fresh ``speech_start`` because + // ``pubSpeaking`` would still be ``true`` from the first batch. + const { vad } = buildVad({ + probs: [0.95, 0.95, 0.95, 0.95, 0.95, 0.95, 0.95, 0.95], + minSpeechDuration: 0.032, + }); + const first = await vad.processFrame(sinePcm(512, 16000), 16000); + expect(first?.type).toBe('speech_start'); + vad.reset(); + const second = await vad.processFrame(sinePcm(512, 16000), 16000); + expect(second?.type).toBe('speech_start'); + }); + + it('reset() on a closed instance is a no-op', async () => { + const { vad } = buildVad({ probs: [0] }); + await vad.close(); + expect(() => vad.reset()).not.toThrow(); + }); }); diff --git a/libraries/typescript/tests/unit/stream-handler.test.ts b/libraries/typescript/tests/unit/stream-handler.test.ts index 1cb34cfb..9a3789fb 100644 --- a/libraries/typescript/tests/unit/stream-handler.test.ts +++ b/libraries/typescript/tests/unit/stream-handler.test.ts @@ -406,6 +406,7 @@ describe('StreamHandler', () => { return h as unknown as { isSpeaking: boolean; speakingStartedAt: number | null; + firstAudioSentAt: number | null; aec: unknown; canBargeIn: () => boolean; handleBargeIn: (t: { text?: string }) => boolean; @@ -421,23 +422,39 @@ describe('StreamHandler', () => { expect(p.canBargeIn()).toBe(true); }); + it('canBargeIn() false before the first TTS chunk has hit the wire', () => { + // 0.6.2 fix: ElevenLabs first-byte latency is hundreds of ms. Pre-fix + // a 250 ms gate measured from beginSpeaking expired before any audio + // went out, letting background noise self-cancel the agent's first + // turn. Post-fix the gate is anchored on firstAudioSentAt — if that's + // null we are still waiting for the TTS provider's first byte. + const h = new StreamHandler(makeDeps(), makeMockWs(), '+15551111111', '+15552222222'); + const p = priv(h); + p.aec = null; + p.speakingStartedAt = Date.now() - 5000; // long past the 250 ms gate + p.firstAudioSentAt = null; // but no audio has gone out yet + expect(p.canBargeIn()).toBe(false); + }); + // ----------------------------------------------------------------------- - // AEC OFF (default — PSTN deployments). Gate is 250 ms. + // AEC OFF (default — PSTN deployments). Gate is 100 ms. // ----------------------------------------------------------------------- describe('AEC off (PSTN default)', () => { - it('canBargeIn() false within 250 ms anti-flicker window', () => { + it('canBargeIn() false within 100 ms anti-flicker window', () => { const h = new StreamHandler(makeDeps(), makeMockWs(), '+15551111111', '+15552222222'); const p = priv(h); p.aec = null; - p.speakingStartedAt = Date.now() - 100; + p.speakingStartedAt = Date.now() - 50; + p.firstAudioSentAt = Date.now() - 50; // 50 ms — still inside 100 ms gate expect(p.canBargeIn()).toBe(false); }); - it('canBargeIn() true past 250 ms (well below the 1 s AEC gate)', () => { + it('canBargeIn() true past 100 ms (well below the 1 s AEC gate)', () => { const h = new StreamHandler(makeDeps(), makeMockWs(), '+15551111111', '+15552222222'); const p = priv(h); p.aec = null; - p.speakingStartedAt = Date.now() - 400; // 400 ms — past 250 ms, under 1 s + p.speakingStartedAt = Date.now() - 200; + p.firstAudioSentAt = Date.now() - 200; // 200 ms — past 100 ms gate, under 1 s expect(p.canBargeIn()).toBe(true); }); @@ -448,6 +465,7 @@ describe('StreamHandler', () => { p.aec = null; p.isSpeaking = true; p.speakingStartedAt = Date.now() - 400; + p.firstAudioSentAt = Date.now() - 400; const result = p.handleBargeIn({ text: 'stop' }); expect(result).toBe(true); expect(p.isSpeaking).toBe(false); @@ -467,6 +485,7 @@ describe('StreamHandler', () => { const p = priv(h); p.aec = aecSentinel; p.speakingStartedAt = Date.now() - 400; // would PASS with AEC off + p.firstAudioSentAt = Date.now() - 400; expect(p.canBargeIn()).toBe(false); }); @@ -475,6 +494,7 @@ describe('StreamHandler', () => { const p = priv(h); p.aec = aecSentinel; p.speakingStartedAt = Date.now() - 1200; + p.firstAudioSentAt = Date.now() - 1200; expect(p.canBargeIn()).toBe(true); }); @@ -484,6 +504,7 @@ describe('StreamHandler', () => { p.aec = aecSentinel; p.isSpeaking = true; p.speakingStartedAt = Date.now() - 400; + p.firstAudioSentAt = Date.now() - 400; const result = p.handleBargeIn({ text: 'stop' }); expect(result).toBe(false); expect(p.isSpeaking).toBe(true); @@ -495,10 +516,435 @@ describe('StreamHandler', () => { p.aec = aecSentinel; p.isSpeaking = true; p.speakingStartedAt = Date.now() - 1500; + p.firstAudioSentAt = Date.now() - 1500; const result = p.handleBargeIn({ text: 'stop' }); expect(result).toBe(true); expect(p.isSpeaking).toBe(false); }); }); }); + + // ------------------------------------------------------------------------- + // firstMessage mark-gated pacing — BUG #128 regression coverage. + // + // Pre-fix the firstMessage TTS chunks were pushed into the carrier + // WebSocket as fast as the TTS provider yielded them. A barge-in + // mid-buffer issued ``sendClear``, but the WebSocket queue between the + // SDK and Twilio's edge held several seconds of media frames already, + // and the agent kept talking on the user's earpiece until that drained. + // + // Post-fix every chunk is followed by a mark; the loop awaits the + // oldest mark before sending more once ``FIRST_MESSAGE_MARK_WINDOW`` + // chunks are unconfirmed. ``cancelSpeaking`` drains every pending mark + // so the waiting loop exits on the next tick. + // ------------------------------------------------------------------------- + describe('firstMessage mark-gated pacing', () => { + interface FmPriv { + isSpeaking: boolean; + speakingStartedAt: number | null; + firstAudioSentAt: number | null; + aec: unknown; + streamSid: string; + pendingMarks: Array<{ name: string; resolve: () => void; promise: Promise }>; + firstMessageMarkCounter: number; + sendPacedFirstMessageBytes: (b: Buffer) => Promise; + onMark: (n: string) => Promise; + runBargeInCancel: (t: string) => void; + } + + function fmPriv(h: StreamHandler): FmPriv { + return h as unknown as FmPriv; + } + + function primeForFirstMessage(h: StreamHandler): FmPriv { + const p = fmPriv(h); + p.isSpeaking = true; + p.speakingStartedAt = Date.now() - 5000; + p.firstAudioSentAt = Date.now() - 5000; + p.aec = null; + p.streamSid = 'MZtest'; + return p; + } + + async function flushMicrotasks(count = 10): Promise { + for (let i = 0; i < count; i++) await Promise.resolve(); + } + + const CHUNK_BYTES = 1280; // matches StreamHandler.PREWARM_CHUNK_BYTES + // 1280 bytes / 32 bytes-per-ms = 40 ms of PCM16 16kHz audio per chunk. + const PLAYOUT_MS = CHUNK_BYTES / 32; + + beforeEach(() => { + vi.useFakeTimers(); + }); + + afterEach(() => { + vi.useRealTimers(); + }); + + it('caps in-flight chunks at FIRST_MESSAGE_MARK_WINDOW and bails on barge-in', async () => { + const sendAudio = vi.fn(); + const sendMark = vi.fn(); + const sendClear = vi.fn(); + const bridge = makeMockBridge({ sendAudio, sendMark, sendClear }); + const h = new StreamHandler( + makeDeps({ bridge }), + makeMockWs(), + '+15551111111', + '+15552222222', + ); + const p = primeForFirstMessage(h); + // 4 chunks. Window=3, so chunks 1–3 send after their 40ms sleeps and + // chunk 4 blocks on waitForMarkWindow until either a mark echoes OR + // cancelSpeaking drains the queue. + const bytes = Buffer.alloc(CHUNK_BYTES * 4, 0); + const sendPromise = p.sendPacedFirstMessageBytes(bytes); + + // Chunks 1–2 go out without sleep (initial burst to pre-fill the PSTN + // jitter buffer). Chunk 3 triggers the first fill of the mark window + // and its 40ms playout sleep. Advancing by PLAYOUT_MS fires that sleep + // and the loop blocks at waitForMarkWindow for chunk 4. + await vi.advanceTimersByTimeAsync(PLAYOUT_MS); + expect(sendAudio).toHaveBeenCalledTimes(3); + expect(sendMark).toHaveBeenCalledTimes(3); + expect(p.pendingMarks.length).toBe(3); + + // Simulate a confirmed barge-in: runBargeInCancel calls sendClear + + // cancelSpeaking, and cancelSpeaking drains pendingMarks so the + // sliding-window wait exits on the next tick. + p.runBargeInCancel('the user spoke'); + await sendPromise; + + expect(sendClear).toHaveBeenCalledTimes(1); + expect(p.isSpeaking).toBe(false); + // Chunk 4 must NOT have hit the wire. + expect(sendAudio).toHaveBeenCalledTimes(3); + }); + + it('echoed mark slides the window and the next chunk goes out', async () => { + const sendAudio = vi.fn(); + const sendMark = vi.fn(); + const bridge = makeMockBridge({ sendAudio, sendMark }); + const h = new StreamHandler( + makeDeps({ bridge }), + makeMockWs(), + '+15551111111', + '+15552222222', + ); + const p = primeForFirstMessage(h); + const bytes = Buffer.alloc(CHUNK_BYTES * 4, 0); + const sendPromise = p.sendPacedFirstMessageBytes(bytes); + + // Chunks 1–2 burst (no sleep). Chunk 3 fills the window → 40ms sleep. + await vi.advanceTimersByTimeAsync(PLAYOUT_MS); + // Three chunks in flight, one waiting on the window. + expect(sendAudio).toHaveBeenCalledTimes(3); + expect(sendMark).toHaveBeenCalledTimes(3); + + // Twilio echoes the FIRST chunk's mark — the loop should advance. + await p.onMark('fm_1'); + // Flush microtasks so waitForMarkWindow exits and chunk 4 sends. + await flushMicrotasks(); + // Advance 1 × 40ms for chunk 4's playout sleep. + await vi.advanceTimersByTimeAsync(PLAYOUT_MS); + + expect(sendAudio).toHaveBeenCalledTimes(4); + expect(sendMark).toHaveBeenCalledTimes(4); + // Let the remaining marks "play" so the loop returns. + await p.onMark('fm_2'); + await p.onMark('fm_3'); + await p.onMark('fm_4'); + await sendPromise; + expect(p.pendingMarks.length).toBe(0); + }); + + it('Telnyx (no marks): paces via playout-time and bails on cancelSpeaking', async () => { + const sendAudio = vi.fn(); + const sendMark = vi.fn(); + const sendClear = vi.fn(); + const bridge = makeMockBridge({ + telephonyProvider: 'telnyx', + sendAudio, + sendMark, + sendClear, + }); + const h = new StreamHandler( + makeDeps({ bridge }), + makeMockWs(), + '+15551111111', + '+15552222222', + ); + const p = primeForFirstMessage(h); + // 4 chunks. Each iteration awaits a fake setTimeout (40 ms), so the loop + // emits the first chunk and suspends on the fake clock. + const bytes = Buffer.alloc(CHUNK_BYTES * 4, 0); + const sendPromise = p.sendPacedFirstMessageBytes(bytes); + + // Flush microtasks to let chunk 1 send and hit the fake 40ms timer. + await flushMicrotasks(); + // Telnyx never sends marks — the queue stays empty even mid-loop. + expect(sendMark).not.toHaveBeenCalled(); + expect(p.pendingMarks.length).toBe(0); + // At least the first chunk should have hit the wire by the time + // we trip the cancel. + const sentBeforeCancel = sendAudio.mock.calls.length; + expect(sentBeforeCancel).toBeGreaterThanOrEqual(1); + + p.runBargeInCancel('user spoke'); + // Fire the pending 40ms sleep so the loop can observe isSpeaking=false. + await vi.runAllTimersAsync(); + await sendPromise; + + expect(sendClear).toHaveBeenCalledTimes(1); + expect(p.isSpeaking).toBe(false); + // After cancel no further chunks may go out. + expect(sendAudio).toHaveBeenCalledTimes(sentBeforeCancel); + }); + }); + + describe('cleanup drains pending firstMessage marks', () => { + interface CleanupPriv { + isSpeaking: boolean; + speakingStartedAt: number | null; + firstAudioSentAt: number | null; + aec: unknown; + streamSid: string; + pendingMarks: Array<{ name: string; resolve: () => void; promise: Promise }>; + firstMessageMarkCounter: number; + sendMarkAwaitable: () => Promise | null; + } + + function priv(h: StreamHandler): CleanupPriv { + return h as unknown as CleanupPriv; + } + + function primeForFirstMessage(h: StreamHandler): CleanupPriv { + const p = priv(h); + p.isSpeaking = true; + p.speakingStartedAt = Date.now() - 5000; + p.firstAudioSentAt = Date.now() - 5000; + p.aec = null; + p.streamSid = 'MZtest'; + return p; + } + + it('handleStop resolves every pending mark', async () => { + const sendMark = vi.fn(); + const bridge = makeMockBridge({ sendMark }); + const h = new StreamHandler( + makeDeps({ bridge }), + makeMockWs(), + '+15551111111', + '+15552222222', + ); + const p = primeForFirstMessage(h); + + // Queue three marks via the public send path then simulate an + // abnormal stop mid firstMessage. Capture each promise so we + // can assert they all resolve after handleStop. + const m1 = p.sendMarkAwaitable(); + const m2 = p.sendMarkAwaitable(); + const m3 = p.sendMarkAwaitable(); + expect(p.pendingMarks.length).toBe(3); + + await h.handleStop(); + + expect(p.pendingMarks.length).toBe(0); + // Every captured promise resolved (await would hang otherwise). + await Promise.all([m1, m2, m3]); + }); + + it('handleWsClose resolves every pending mark', async () => { + const sendMark = vi.fn(); + const bridge = makeMockBridge({ sendMark }); + const h = new StreamHandler( + makeDeps({ bridge }), + makeMockWs(), + '+15551111111', + '+15552222222', + ); + const p = primeForFirstMessage(h); + + const m1 = p.sendMarkAwaitable(); + const m2 = p.sendMarkAwaitable(); + const m3 = p.sendMarkAwaitable(); + expect(p.pendingMarks.length).toBe(3); + + await h.handleWsClose(); + + expect(p.pendingMarks.length).toBe(0); + await Promise.all([m1, m2, m3]); + }); + }); + + describe('firstMessage mark counter resets across sends + on cleanup', () => { + interface CounterPriv { + isSpeaking: boolean; + speakingStartedAt: number | null; + firstAudioSentAt: number | null; + aec: unknown; + streamSid: string; + pendingMarks: Array<{ name: string; resolve: () => void; promise: Promise }>; + firstMessageMarkCounter: number; + sendPacedFirstMessageBytes: (b: Buffer) => Promise; + onMark: (n: string) => Promise; + } + + function priv(h: StreamHandler): CounterPriv { + return h as unknown as CounterPriv; + } + + function primeForFirstMessage(h: StreamHandler): CounterPriv { + const p = priv(h); + p.isSpeaking = true; + p.speakingStartedAt = Date.now() - 5000; + p.firstAudioSentAt = Date.now() - 5000; + p.aec = null; + p.streamSid = 'MZtest'; + return p; + } + + beforeEach(() => { + vi.useFakeTimers(); + }); + + afterEach(() => { + vi.useRealTimers(); + }); + + it('sendPacedFirstMessageBytes resets counter between consecutive sends', async () => { + const sendAudio = vi.fn(); + const sendMark = vi.fn(); + const bridge = makeMockBridge({ sendAudio, sendMark }); + const h = new StreamHandler( + makeDeps({ bridge }), + makeMockWs(), + '+15551111111', + '+15552222222', + ); + const p = primeForFirstMessage(h); + + // CHUNK_BYTES = 1280 matches StreamHandler.PREWARM_CHUNK_BYTES. + // Two chunks stay below FIRST_MESSAGE_MARK_WINDOW (3) so initialFillComplete + // never flips to true and neither chunk triggers a playout sleep on Twilio. + // advanceTimersByTimeAsync flushes microtasks first so both chunks are sent + // synchronously before any time advance occurs. + const CHUNK_BYTES = 1280; + const PLAYOUT_MS = CHUNK_BYTES / 32; // 40ms (not used on Twilio here, kept for Telnyx parity) + const bytes = Buffer.alloc(CHUNK_BYTES * 2, 0); + + const send1 = p.sendPacedFirstMessageBytes(bytes); + // Flush microtasks (both chunks go out without sleep) and advance time + // to drain any stray timers; marks are still pending — echo them. + await vi.advanceTimersByTimeAsync(2 * PLAYOUT_MS); + await p.onMark('fm_1'); + await p.onMark('fm_2'); + await send1; + expect(p.firstMessageMarkCounter).toBe(2); + expect(p.pendingMarks.length).toBe(0); + const markCallsAfterFirst = sendMark.mock.calls.length; + expect( + sendMark.mock.calls.slice(0, markCallsAfterFirst).map((c) => c[1] as string), + ).toEqual(['fm_1', 'fm_2']); + + // Second send: counter must reset to 0 at the top of the loop, + // so the new sequence is fm_1, fm_2 — NOT fm_3, fm_4. + const send2 = p.sendPacedFirstMessageBytes(bytes); + await vi.advanceTimersByTimeAsync(2 * PLAYOUT_MS); + const newMarks = sendMark.mock.calls + .slice(markCallsAfterFirst) + .map((c) => c[1] as string); + expect(newMarks).toEqual(['fm_1', 'fm_2']); + expect(p.firstMessageMarkCounter).toBe(2); + + await p.onMark('fm_1'); + await p.onMark('fm_2'); + await send2; + }); + + it('handleStop resets firstMessageMarkCounter', async () => { + const h = new StreamHandler( + makeDeps(), + makeMockWs(), + '+15551111111', + '+15552222222', + ); + const p = priv(h); + // Pretend a prior turn left the counter at 7. + p.firstMessageMarkCounter = 7; + + await h.handleStop(); + + expect(p.firstMessageMarkCounter).toBe(0); + }); + + it('handleWsClose resets firstMessageMarkCounter', async () => { + const h = new StreamHandler( + makeDeps(), + makeMockWs(), + '+15551111111', + '+15552222222', + ); + const p = priv(h); + p.firstMessageMarkCounter = 7; + + await h.handleWsClose(); + + expect(p.firstMessageMarkCounter).toBe(0); + }); + }); + + describe('onMark only updates lastConfirmedMark on a matched mark', () => { + interface OnMarkPriv { + pendingMarks: Array<{ name: string; resolve: () => void; promise: Promise }>; + lastConfirmedMark: string; + } + + it('does not overwrite lastConfirmedMark for an unknown mark name', async () => { + const h = new StreamHandler( + makeDeps(), + makeMockWs(), + '+15551111111', + '+15552222222', + ); + const p = h as unknown as OnMarkPriv; + + // Seed a real matched mark so lastConfirmedMark has a known + // baseline that the unmatched echo must not overwrite. + let resolveSeed!: () => void; + const seedPromise = new Promise((r) => { + resolveSeed = r; + }); + p.pendingMarks.push({ name: 'fm_seed', resolve: resolveSeed, promise: seedPromise }); + await h.onMark('fm_seed'); + expect(p.lastConfirmedMark).toBe('fm_seed'); + + // Emit a mark name that is NOT in pendingMarks — e.g. echo + // arrived after drain, or for an unknown identifier. The + // handler's lastConfirmedMark must NOT be clobbered. + await h.onMark('unknown_xyz'); + expect(p.lastConfirmedMark).toBe('fm_seed'); + }); + + it('updates lastConfirmedMark only after the queue match succeeds', async () => { + const h = new StreamHandler( + makeDeps(), + makeMockWs(), + '+15551111111', + '+15552222222', + ); + const p = h as unknown as OnMarkPriv; + expect(p.lastConfirmedMark).toBe(''); + + let resolveA!: () => void; + const promiseA = new Promise((r) => { + resolveA = r; + }); + p.pendingMarks.push({ name: 'fm_1', resolve: resolveA, promise: promiseA }); + + await h.onMark('fm_1'); + expect(p.lastConfirmedMark).toBe('fm_1'); + expect(p.pendingMarks.length).toBe(0); + }); + }); }); diff --git a/libraries/typescript/tests/unit/transcoding-stateful.test.ts b/libraries/typescript/tests/unit/transcoding-stateful.test.ts index 529c06ab..1b6011d7 100644 --- a/libraries/typescript/tests/unit/transcoding-stateful.test.ts +++ b/libraries/typescript/tests/unit/transcoding-stateful.test.ts @@ -184,13 +184,20 @@ describe('StatefulResampler 16k→8k', () => { } }); - it('DC signal passes through unchanged (FIR unity gain on DC)', () => { + it('DC signal passes through unchanged after startup transient (FIR unity gain on DC)', () => { const r = createResampler16kTo8k(); const input = i16buf(Array(20).fill(5000)); const out = r.process(input); const samples = readI16(out); - // Allow ±1 LSB for integer rounding. - for (const s of samples) expect(Math.abs(s - 5000)).toBeLessThanOrEqual(1); + // The FIR history is zero-initialized (correct initial condition for + // no prior audio). The very first output sample blends zero history + // with the DC signal and produces a lower value — this is the + // expected startup transient. From sample[1] onward the filter is + // in steady state and unity gain at DC gives exactly 5000 (±1 LSB). + expect(samples.length).toBeGreaterThanOrEqual(2); + for (let i = 1; i < samples.length; i++) { + expect(Math.abs(samples[i] - 5000)).toBeLessThanOrEqual(1); + } }); it('handles odd-byte input via carry (no throw, output still even)', () => {
+ + Status From → To Carrier
+ No calls match "{search}"