FlutterSoundPlayer.feedUint8FromStream() requires interleaved mode.
With interleaved=false, every feed() call threw:
"Cannot feed with UInt8 with non interleaved mode"
- feedUint8FromStream (Uint8List) → requires interleaved: true
- feedFromStream (Float32List) → requires interleaved: false
Since we feed raw PCM bytes (Uint8List), interleaved must be true.
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
Root cause: PcmPlayer called openPlayer() without audio session config,
so Android defaulted to earpiece-only mode. When the mic was actively
recording, playback was silently suppressed — the agent's TTS audio was
sent successfully over WebSocket but never reached the speaker.
Changes:
1. PcmPlayer (pcm_player.dart):
- Added audio_session package for proper audio session management
- Configure AudioSession with playAndRecord category so mic + speaker
work simultaneously
- Set voiceCommunication usage to enable Android hardware AEC (echo
cancellation) — prevents feedback loops when speaker is active
- defaultToSpeaker routes output to loudspeaker instead of earpiece
- Restored setSpeakerOn() method stub (used by UI toggle)
2. AgentCallPage (agent_call_page.dart):
- Fixed fire-and-forget bug: _pcmPlayer.feed() returns Future but was
called without await, causing interleaved feedUint8FromStream calls
- Added _feedChain serializer to guarantee sequential audio feeding
3. Dependencies:
- Added audio_session package to pubspec.yaml
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>
hailin