OpenAI TTS returns 24kHz audio which Android MediaPlayer can't play
via FlutterSound's pcm16WAV codec. Request raw PCM and resample to
16kHz before wrapping in WAV header, matching the local TTS format.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
Local /synthesize and /transcribe endpoints now auto-load Kokoro/Whisper
models on first call instead of returning 503 when not pre-loaded at
startup. This allows switching between Local and OpenAI providers in the
Flutter test page without requiring server restart.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
- Add STT_PROVIDER/TTS_PROVIDER config (local or openai) in settings
- Pipeline uses OpenAI API for STT/TTS when provider is "openai"
- Skip loading local models (Kokoro/faster-whisper) when using OpenAI
- VAD (Silero) always loads for speech detection
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
- Add openai package to voice-service requirements
- Add /api/v1/test/tts/synthesize-openai (tts-1/tts-1-hd/gpt-4o-mini-tts)
- Add /api/v1/test/stt/transcribe-openai (gpt-4o-transcribe/whisper-1)
- Add OPENAI_API_KEY and OPENAI_BASE_URL env vars to voice-service
- Flutter test page: SegmentedButton to toggle Local/OpenAI provider
- All endpoints maintain same response format for easy comparison
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
- STT: record from mic or upload audio file → faster-whisper transcription
- Round-trip: record → STT → TTS → playback (full pipeline test)
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
Browser-accessible page to test text-to-speech synthesis without
going through the full voice pipeline.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
Voice calls now use the same agent task + WS subscription flow as the
chat UI, enabling tool use and command execution during voice sessions.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
Root cause: Pipecat's WebsocketServerTransport creates its own WebSocket
server on (host,port) and expects FrameProcessor subclasses. Our code was
passing a FastAPI WebSocket object as 'host' and using plain STT/TTS/VAD
service classes that aren't FrameProcessors. The pipeline crashed immediately
when receiving audio, causing "disconnects when speaking".
Changes:
- **base_pipeline.py**: Complete rewrite — replaced Pipecat Pipeline with
direct async loop: WebSocket → VAD → STT → Claude LLM → TTS → WebSocket.
Supports barge-in (interrupt TTS when user speaks), audio chunking, and
24kHz→16kHz TTS resampling.
- **session_router.py**: Pass WebSocket directly to pipeline instead of
wrapping in AppTransport.
- **app_transport.py**: Deprecated (no longer needed).
- **kokoro_service.py**: Fix misaki compatibility (MutableToken→MToken
rename), use correct Chinese voice 'zf_xiaoxiao', handle torch tensors.
- **main.py**: Apply misaki monkey-patch before importing kokoro.
- **settings.py**: Change default TTS voice from 'zh_female_1' (non-existent)
to 'zf_xiaoxiao' (valid Kokoro-82M Chinese female voice).
- **requirements.txt**: Remove pipecat-ai dependency, pin kokoro==0.3.5 +
misaki==0.7.17, add Chinese NLP deps (pypinyin, cn2an, jieba, ordered-set).
- **agent_call_page.dart**: Wrap each cleanup step in try/catch to ensure
Navigator.pop() always executes after call ends. Add 3s timeout on session
delete request.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
Addresses reliability gaps in the real-time voice WebSocket connection
between Flutter client and Python voice-service backend.
Backend (voice-service):
- Heartbeat: new _heartbeat_sender coroutine sends JSON ping text frames
every 15s alongside the Pipecat pipeline; failed send = dead connection
- Session preservation: on WebSocket disconnect, sessions are now marked
"disconnected" with a timestamp instead of being deleted, allowing
reconnection within a configurable TTL (default 60s)
- Reconnect endpoint: POST /sessions/{id}/reconnect verifies the session
is alive and in "disconnected" state, returns fresh websocket_url
- Reconnect-aware WS handler: detects "disconnected" sessions, cancels
stale pipeline tasks, creates a new pipeline, sends "session.resumed"
- Background cleanup: asyncio loop every 30s removes sessions that have
been disconnected longer than session_ttl
- Structured event protocol: text frames = JSON control messages
(ping/pong/session.resumed/session.ended/error), binary = PCM audio
- New settings: session_ttl (60s), heartbeat_interval (15s),
heartbeat_timeout (45s)
Flutter (agent_call_page.dart):
- Heartbeat monitoring: tracks last server ping timestamp, triggers
reconnect if no ping received in 45s (3 missed intervals)
- Auto-reconnect: exponential backoff (1s→2s→4s→8s→16s), max 5 attempts;
calls /reconnect endpoint to verify session, rebuilds WebSocket,
resets audio buffer, restarts heartbeat
- Reconnecting UI: yellow warning banner "重新连接中... (N/5)" with
spinner overlay during reconnection attempts
- WebSocket data routing: _onWsData distinguishes String (JSON control)
from binary (audio) frames, handles ping/session.resumed/session.ended
- User-initiated disconnect guard: _userEndedCall flag prevents reconnect
attempts when user intentionally hangs up
- session_id field compatibility: supports session_id/sessionId/id
Flutter (pcm_player.dart):
- Jitter buffer: queues incoming PCM chunks, starts playback only after
accumulating 4800 bytes (150ms at 16kHz 16-bit mono) to smooth out
network timing variance
- reset() method: clears buffer on reconnect to discard stale audio
- Buffer underrun handling: re-enters buffering phase if queue empties
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
Replace traditional on-device speech_to_text with a modern pipeline:
- Record audio via `record` package with hardware noise suppression
- Apply GTCRN neural denoising (sherpa-onnx, ICASSP 2024, 48K params)
- Trim silence, POST to backend /voice/transcribe (faster-whisper)
Changes:
- Add /transcribe endpoint to voice-service for audio file upload
- Add SpeechEnhancer wrapper for sherpa-onnx GTCRN model (523KB)
- Rewrite chat_page.dart voice input: record → denoise → transcribe
- Keep NoiseReducer.trimSilence for silence removal only
- Upgrade record to v6.2.0, add sherpa_onnx, path_provider
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
Model downloads (Whisper, Kokoro, Silero VAD) are synchronous blocking
calls that prevent uvicorn from completing startup and responding to
healthchecks. Move all model loading to a daemon thread so the server
starts immediately.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
- Wrap model loading in try/except so server starts even if models fail
- Fix device env var mapping (unified 'device' field instead of 'whisper_device')
- Default Whisper model to 'base' instead of 'large-v3' (3GB) for CPU deployment
- Increase healthcheck start_period to 120s for model download time
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
hailin