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).
This commit is contained in:
2026-06-04 15:23:57 -04:00
parent 188da1d52a
commit 0e74a16b40
2 changed files with 33 additions and 200 deletions
+1 -186
View File
@@ -15,7 +15,7 @@ from fastapi.middleware.cors import CORSMiddleware
from config import settings from config import settings
from services.memory import kira_memory from services.memory import kira_memory
from services.whisper_stream import WhisperStream # from services.whisper_stream import WhisperStream # REST fallback active
logging.basicConfig(level=logging.INFO) logging.basicConfig(level=logging.INFO)
logger = logging.getLogger("kira") logger = logging.getLogger("kira")
@@ -73,190 +73,5 @@ async def conversation_ws(websocket: WebSocket):
logger.info(f"[{session_id}] WebSocket connected") logger.info(f"[{session_id}] WebSocket connected")
conversation_history: list[dict] = [] conversation_history: list[dict] = []
pending_transcript: str | None = None
transcript_lock = asyncio.Lock()
# ── Whisper stream callbacks ──
async def on_ready():
logger.info(f"[{session_id}] Whisper stream ready")
async def on_delta(delta: str):
"""Streaming partial transcript — forward to client."""
try:
await websocket.send_json({"type": "transcript_delta", "text": delta})
except Exception:
pass
async def on_done(full: str):
"""Full utterance from VAD. Kick off LLM + TTS."""
nonlocal pending_transcript
logger.info(f"[{session_id}] Full transcript ({len(full)} chars): {full}")
async with transcript_lock:
pending_transcript = full
await websocket.send_json({"type": "transcript", "role": "user", "text": full})
conversation_history.append({"role": "user", "content": full})
# LLM
system_prompt = BASE_SYSTEM_PROMPT
if memory_suffix:
system_prompt += memory_suffix
client = get_openai()
resp = await client.chat.completions.create(
model="gpt-5.4-nano",
messages=[
{"role": "system", "content": system_prompt},
{"role": "user", "content": full},
],
max_completion_tokens=300,
temperature=0.7,
)
kira_text = resp.choices[0].message.content or "Mhm, I'm here!"
conversation_history.append({"role": "assistant", "content": kira_text})
logger.info(f"[{session_id}] Kira: {kira_text}")
# Store in Honcho
if kira_memory.enabled and identified:
try:
kira_memory.store_messages(full, kira_text)
except Exception:
pass
# Streaming TTS
await websocket.send_json({"type": "speaking_start", "text": kira_text})
async with client.audio.speech.with_streaming_response.create(
model="tts-1",
voice="sage",
input=kira_text,
response_format="opus",
) as tts_resp:
async for chunk in tts_resp.iter_bytes():
if chunk:
b64 = base64.b64encode(chunk).decode("utf-8")
await websocket.send_json({"type": "audio", "data": b64})
await websocket.send_json({"type": "speaking_end"})
async def on_error(msg: str):
logger.warning(f"Whisper error: {msg}")
try:
await websocket.send_json({"type": "error", "message": f"Transcription error: {msg}"})
except Exception:
pass
# Start WhisperStream
stream = WhisperStream(
on_transcript_delta=on_delta,
on_transcript_done=on_done,
on_ready=on_ready,
on_error=on_error,
)
stream_task = asyncio.create_task(stream.connect())
await asyncio.sleep(2) # brief wait for connection
try:
while True:
raw = await websocket.receive_text()
msg = json.loads(raw)
msg_type = msg.get("type", "")
# ── Identity & Preferences ──
if msg_type == "identify":
user_id = msg.get("user_id", "").strip()
user_name = msg.get("name", "").strip()
if user_name and user_id:
kira_memory.set_user_preference(user_id, "name", user_name)
prefs = kira_memory.get_user_preferences(user_id)
identified = True
if kira_memory.enabled:
kira_memory.ensure_peers(user_id)
kira_memory.ensure_session(session_id)
try:
ctx = kira_memory.build_system_prompt_suffix()
if ctx:
memory_suffix = ctx
except Exception:
pass
await websocket.send_json({
"type": "identified",
"user_id": user_id,
"preferences": prefs,
})
continue
if msg_type == "set_preference":
key = msg.get("key", "").strip()
value = msg.get("value", "").strip()
if key and user_id and user_id != "default-user":
kira_memory.set_user_preference(user_id, key, value)
await websocket.send_json({"type": "preference_saved", "key": key, "success": True})
continue
# ── PCM16 audio → WhisperStream ──
if msg_type == "audio":
pcm16 = base64.b64decode(msg["data"])
await stream.send_audio(pcm16)
continue
# ── Text input → direct LLM + TTS ──
if msg_type == "conversation_text":
text = msg.get("text", "").strip()
if not text:
continue
logger.info(f"[{session_id}] User (text): {text}")
conversation_history.append({"role": "user", "content": text})
system_prompt = BASE_SYSTEM_PROMPT
if memory_suffix:
system_prompt += memory_suffix
client = get_openai()
resp = await client.chat.completions.create(
model="gpt-5.4-nano",
messages=[
{"role": "system", "content": system_prompt},
{"role": "user", "content": text},
],
max_completion_tokens=300,
temperature=0.7,
)
kira_text = resp.choices[0].message.content or "Mhm!"
conversation_history.append({"role": "assistant", "content": kira_text})
logger.info(f"[{session_id}] Kira: {kira_text}")
if kira_memory.enabled and identified:
try:
kira_memory.store_messages(text, kira_text)
except Exception:
pass
await websocket.send_json({"type": "speaking_start", "text": kira_text})
async with client.audio.speech.with_streaming_response.create(
model="tts-1",
voice="sage",
input=kira_text,
response_format="opus",
) as tts_resp:
async for chunk in tts_resp.iter_bytes():
if chunk:
b64 = base64.b64encode(chunk).decode("utf-8")
await websocket.send_json({"type": "audio", "data": b64})
await websocket.send_json({"type": "speaking_end"})
continue
if msg_type == "ping":
await websocket.send_json({"type": "pong"})
except WebSocketDisconnect:
logger.info(f"[{session_id}] Disconnected")
except Exception as e:
logger.error(f"[{session_id}] Error: {e}")
finally:
await stream.disconnect()
stream_task.cancel()
+28 -10
View File
@@ -201,7 +201,7 @@ export function useConversation() {
try { try {
setMicError(null); setMicError(null);
const stream = await navigator.mediaDevices.getUserMedia({ audio: true }); const stream = await navigator.mediaDevices.getUserMedia({ audio: { echoCancellation: true, noiseSuppression: true } });
streamRef.current = stream; streamRef.current = stream;
const ws = wsRef.current; const ws = wsRef.current;
@@ -211,14 +211,27 @@ export function useConversation() {
return; return;
} }
// PCM16 capture for Realtime WebSocket STT // Use MediaRecorder for full utterance blob (Opus/webm) — sent on stop for REST STT
captureRef.current = startPCMCapture(stream, (pcm16) => { const mediaRecorder = new MediaRecorder(stream, { mimeType: 'audio/webm;codecs=opus' });
if (ws.readyState === WebSocket.OPEN) { const chunks: Blob[] = [];
const base64 = arrayBufferToBase64(pcm16.buffer); mediaRecorder.ondataavailable = (e) => {
if (e.data.size > 0) chunks.push(e.data);
};
mediaRecorder.onstop = () => {
if (chunks.length > 0 && ws.readyState === WebSocket.OPEN) {
const blob = new Blob(chunks, { type: 'audio/webm' });
blob.arrayBuffer().then((buf) => {
const base64 = arrayBufferToBase64(buf);
ws.send(JSON.stringify({ type: 'audio', data: base64 })); ws.send(JSON.stringify({ type: 'audio', data: base64 }));
}
}); });
}
chunks.length = 0;
stream.getTracks().forEach((t) => t.stop());
streamRef.current = null;
setIsRecording(false);
};
recorderRef.current = mediaRecorder;
mediaRecorder.start();
setIsRecording(true); setIsRecording(true);
} catch (err) { } catch (err) {
const msg = err instanceof Error ? err.message : String(err); const msg = err instanceof Error ? err.message : String(err);
@@ -228,11 +241,16 @@ export function useConversation() {
}, [addMessage]); }, [addMessage]);
const stopRecording = useCallback(() => { const stopRecording = useCallback(() => {
captureRef.current?.stop(); if (recorderRef.current && recorderRef.current.state === 'recording') {
captureRef.current = null; recorderRef.current.stop();
// onstop will handle sending the blob and cleanup
} else {
// fallback cleanup
streamRef.current?.getTracks().forEach((t) => t.stop()); streamRef.current?.getTracks().forEach((t) => t.stop());
streamRef.current = null; streamRef.current = null;
setIsRecording(false); setIsRecording(false);
}
captureRef.current = null; // legacy
}, []); }, []);
// ── Text ── // ── Text ──
@@ -249,8 +267,8 @@ export function useConversation() {
connect(); connect();
return () => { return () => {
wsRef.current?.close(); wsRef.current?.close();
if (recorderRef.current && recorderRef.current.state === 'recording') recorderRef.current.stop();
captureRef.current?.stop(); captureRef.current?.stop();
recorderRef.current?.stop();
streamRef.current?.getTracks().forEach((t) => t.stop()); streamRef.current?.getTracks().forEach((t) => t.stop());
}; };
}, [connect]); }, [connect]);