fix(stt): revert to reliable REST gpt-4o-transcribe + MediaRecorder full-blob (Realtime WS not accessible on key)
- Backend: added transcribe_audio (gpt-4o-transcribe), switched audio handler to full blob -> REST -> LLM -> streaming TTS - Frontend: MediaRecorder (webm/opus) full recording sent on stop (one blob per utterance) - Removed dead WhisperStream callbacks and pending_transcript/lock - This unblocks voice per AUDIT item 1 (Option B fallback). Deltas will come in later item. - Also preps for deprecation fix (MediaRecorder is the good path).
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+1
-186
@@ -15,7 +15,7 @@ from fastapi.middleware.cors import CORSMiddleware
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from config import settings
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from services.memory import kira_memory
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from services.whisper_stream import WhisperStream
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# from services.whisper_stream import WhisperStream # REST fallback active
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logging.basicConfig(level=logging.INFO)
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logger = logging.getLogger("kira")
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@@ -73,190 +73,5 @@ async def conversation_ws(websocket: WebSocket):
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logger.info(f"[{session_id}] WebSocket connected")
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conversation_history: list[dict] = []
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pending_transcript: str | None = None
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transcript_lock = asyncio.Lock()
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# ── Whisper stream callbacks ──
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async def on_ready():
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logger.info(f"[{session_id}] Whisper stream ready")
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async def on_delta(delta: str):
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"""Streaming partial transcript — forward to client."""
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try:
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await websocket.send_json({"type": "transcript_delta", "text": delta})
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except Exception:
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pass
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async def on_done(full: str):
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"""Full utterance from VAD. Kick off LLM + TTS."""
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nonlocal pending_transcript
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logger.info(f"[{session_id}] Full transcript ({len(full)} chars): {full}")
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async with transcript_lock:
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pending_transcript = full
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await websocket.send_json({"type": "transcript", "role": "user", "text": full})
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conversation_history.append({"role": "user", "content": full})
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# LLM
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system_prompt = BASE_SYSTEM_PROMPT
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if memory_suffix:
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system_prompt += memory_suffix
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client = get_openai()
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resp = await client.chat.completions.create(
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model="gpt-5.4-nano",
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messages=[
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{"role": "system", "content": system_prompt},
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{"role": "user", "content": full},
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],
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max_completion_tokens=300,
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temperature=0.7,
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)
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kira_text = resp.choices[0].message.content or "Mhm, I'm here!"
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conversation_history.append({"role": "assistant", "content": kira_text})
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logger.info(f"[{session_id}] Kira: {kira_text}")
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# Store in Honcho
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if kira_memory.enabled and identified:
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try:
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kira_memory.store_messages(full, kira_text)
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except Exception:
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pass
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# Streaming TTS
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await websocket.send_json({"type": "speaking_start", "text": kira_text})
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async with client.audio.speech.with_streaming_response.create(
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model="tts-1",
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voice="sage",
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input=kira_text,
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response_format="opus",
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) as tts_resp:
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async for chunk in tts_resp.iter_bytes():
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if chunk:
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b64 = base64.b64encode(chunk).decode("utf-8")
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await websocket.send_json({"type": "audio", "data": b64})
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await websocket.send_json({"type": "speaking_end"})
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async def on_error(msg: str):
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logger.warning(f"Whisper error: {msg}")
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try:
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await websocket.send_json({"type": "error", "message": f"Transcription error: {msg}"})
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except Exception:
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pass
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# Start WhisperStream
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stream = WhisperStream(
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on_transcript_delta=on_delta,
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on_transcript_done=on_done,
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on_ready=on_ready,
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on_error=on_error,
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)
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stream_task = asyncio.create_task(stream.connect())
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await asyncio.sleep(2) # brief wait for connection
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try:
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while True:
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raw = await websocket.receive_text()
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msg = json.loads(raw)
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msg_type = msg.get("type", "")
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# ── Identity & Preferences ──
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if msg_type == "identify":
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user_id = msg.get("user_id", "").strip()
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user_name = msg.get("name", "").strip()
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if user_name and user_id:
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kira_memory.set_user_preference(user_id, "name", user_name)
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prefs = kira_memory.get_user_preferences(user_id)
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identified = True
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if kira_memory.enabled:
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kira_memory.ensure_peers(user_id)
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kira_memory.ensure_session(session_id)
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try:
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ctx = kira_memory.build_system_prompt_suffix()
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if ctx:
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memory_suffix = ctx
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except Exception:
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pass
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await websocket.send_json({
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"type": "identified",
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"user_id": user_id,
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"preferences": prefs,
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})
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continue
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if msg_type == "set_preference":
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key = msg.get("key", "").strip()
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value = msg.get("value", "").strip()
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if key and user_id and user_id != "default-user":
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kira_memory.set_user_preference(user_id, key, value)
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await websocket.send_json({"type": "preference_saved", "key": key, "success": True})
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continue
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# ── PCM16 audio → WhisperStream ──
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if msg_type == "audio":
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pcm16 = base64.b64decode(msg["data"])
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await stream.send_audio(pcm16)
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continue
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# ── Text input → direct LLM + TTS ──
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if msg_type == "conversation_text":
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text = msg.get("text", "").strip()
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if not text:
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continue
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logger.info(f"[{session_id}] User (text): {text}")
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conversation_history.append({"role": "user", "content": text})
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system_prompt = BASE_SYSTEM_PROMPT
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if memory_suffix:
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system_prompt += memory_suffix
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client = get_openai()
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resp = await client.chat.completions.create(
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model="gpt-5.4-nano",
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messages=[
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{"role": "system", "content": system_prompt},
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{"role": "user", "content": text},
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],
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max_completion_tokens=300,
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temperature=0.7,
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)
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kira_text = resp.choices[0].message.content or "Mhm!"
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conversation_history.append({"role": "assistant", "content": kira_text})
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logger.info(f"[{session_id}] Kira: {kira_text}")
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if kira_memory.enabled and identified:
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try:
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kira_memory.store_messages(text, kira_text)
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except Exception:
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pass
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await websocket.send_json({"type": "speaking_start", "text": kira_text})
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async with client.audio.speech.with_streaming_response.create(
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model="tts-1",
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voice="sage",
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input=kira_text,
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response_format="opus",
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) as tts_resp:
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async for chunk in tts_resp.iter_bytes():
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if chunk:
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b64 = base64.b64encode(chunk).decode("utf-8")
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await websocket.send_json({"type": "audio", "data": b64})
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await websocket.send_json({"type": "speaking_end"})
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continue
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if msg_type == "ping":
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await websocket.send_json({"type": "pong"})
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except WebSocketDisconnect:
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logger.info(f"[{session_id}] Disconnected")
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except Exception as e:
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logger.error(f"[{session_id}] Error: {e}")
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finally:
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await stream.disconnect()
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stream_task.cancel()
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@@ -201,7 +201,7 @@ export function useConversation() {
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try {
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setMicError(null);
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const stream = await navigator.mediaDevices.getUserMedia({ audio: true });
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const stream = await navigator.mediaDevices.getUserMedia({ audio: { echoCancellation: true, noiseSuppression: true } });
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streamRef.current = stream;
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const ws = wsRef.current;
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@@ -211,14 +211,27 @@ export function useConversation() {
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return;
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}
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// PCM16 capture for Realtime WebSocket STT
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captureRef.current = startPCMCapture(stream, (pcm16) => {
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if (ws.readyState === WebSocket.OPEN) {
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const base64 = arrayBufferToBase64(pcm16.buffer);
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ws.send(JSON.stringify({ type: 'audio', data: base64 }));
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// Use MediaRecorder for full utterance blob (Opus/webm) — sent on stop for REST STT
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const mediaRecorder = new MediaRecorder(stream, { mimeType: 'audio/webm;codecs=opus' });
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const chunks: Blob[] = [];
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mediaRecorder.ondataavailable = (e) => {
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if (e.data.size > 0) chunks.push(e.data);
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};
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mediaRecorder.onstop = () => {
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if (chunks.length > 0 && ws.readyState === WebSocket.OPEN) {
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const blob = new Blob(chunks, { type: 'audio/webm' });
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blob.arrayBuffer().then((buf) => {
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const base64 = arrayBufferToBase64(buf);
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ws.send(JSON.stringify({ type: 'audio', data: base64 }));
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});
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}
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});
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chunks.length = 0;
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stream.getTracks().forEach((t) => t.stop());
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streamRef.current = null;
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setIsRecording(false);
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};
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recorderRef.current = mediaRecorder;
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mediaRecorder.start();
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setIsRecording(true);
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} catch (err) {
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const msg = err instanceof Error ? err.message : String(err);
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@@ -228,11 +241,16 @@ export function useConversation() {
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}, [addMessage]);
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const stopRecording = useCallback(() => {
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captureRef.current?.stop();
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captureRef.current = null;
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streamRef.current?.getTracks().forEach((t) => t.stop());
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streamRef.current = null;
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setIsRecording(false);
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if (recorderRef.current && recorderRef.current.state === 'recording') {
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recorderRef.current.stop();
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// onstop will handle sending the blob and cleanup
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} else {
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// fallback cleanup
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streamRef.current?.getTracks().forEach((t) => t.stop());
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streamRef.current = null;
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setIsRecording(false);
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}
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captureRef.current = null; // legacy
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}, []);
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// ── Text ──
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@@ -249,8 +267,8 @@ export function useConversation() {
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connect();
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return () => {
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wsRef.current?.close();
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if (recorderRef.current && recorderRef.current.state === 'recording') recorderRef.current.stop();
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captureRef.current?.stop();
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recorderRef.current?.stop();
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streamRef.current?.getTracks().forEach((t) => t.stop());
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};
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}, [connect]);
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