런타임 계약과 학습자 흐름 보강

This commit is contained in:
Yun Chan 2026-06-29 08:12:14 +09:00
parent f456b8997a
commit 206018b088
56 changed files with 4306 additions and 1008 deletions

View file

@ -177,6 +177,10 @@ async def voice_ws(websocket: WebSocket) -> None:
receiving = False
audio_started_at: float | None = None
last_audio_end_at: float | None = None
audio_format: str | None = None
audio_sample_rate: int | None = None
audio_channels: int | None = None
audio_sample_width: int | None = None
try:
while True:
@ -217,6 +221,10 @@ async def voice_ws(websocket: WebSocket) -> None:
if ctype == "audio_start":
receiving = True
audio_started_at = time.monotonic()
audio_format = _safe_str(ctrl.get("format"))
audio_sample_rate = _safe_int(ctrl.get("sample_rate"))
audio_channels = _safe_int(ctrl.get("channels"))
audio_sample_width = _safe_int(ctrl.get("sample_width"))
audio_buf.clear()
await _safe_send_json(websocket, {"type": "state", "state": "listening"})
@ -226,13 +234,17 @@ async def voice_ws(websocket: WebSocket) -> None:
silence_ms = _safe_int(ctrl.get("silence_ms"))
if silence_ms is None and last_audio_end_at is not None and audio_started_at is not None:
silence_ms = max(0, int((audio_started_at - last_audio_end_at) * 1000))
end_format = _safe_str(ctrl.get("format")) or audio_format
await _handle_utterance(
websocket,
session_id=session_id,
principal=principal,
voice_preset=voice_preset,
audio=bytes(audio_buf),
fmt=ctrl.get("format"),
fmt=end_format,
sample_rate=_safe_int(ctrl.get("sample_rate")) or audio_sample_rate,
channels=_safe_int(ctrl.get("channels")) or audio_channels,
sample_width=_safe_int(ctrl.get("sample_width")) or audio_sample_width,
audio_started_at=audio_started_at,
audio_ended_at=audio_ended_at,
silence_ms=silence_ms,
@ -241,6 +253,10 @@ async def voice_ws(websocket: WebSocket) -> None:
)
last_audio_end_at = audio_ended_at
audio_started_at = None
audio_format = None
audio_sample_rate = None
audio_channels = None
audio_sample_width = None
audio_buf.clear()
elif ctype == "text_turn":
@ -257,6 +273,27 @@ async def voice_ws(websocket: WebSocket) -> None:
learner_text=learner_text,
)
elif ctype == "stt_result":
receiving = False
audio_buf.clear()
stt_received_at = time.monotonic()
await _handle_stt_result_control(
websocket,
session_id=session_id,
principal=principal,
voice_preset=voice_preset,
ctrl=ctrl,
audio_started_at=audio_started_at,
audio_ended_at=stt_received_at,
last_audio_end_at=last_audio_end_at,
)
last_audio_end_at = stt_received_at
audio_started_at = None
audio_format = None
audio_sample_rate = None
audio_channels = None
audio_sample_width = None
elif ctype == "ping":
await _safe_send_json(websocket, {"type": "pong"})
@ -271,6 +308,60 @@ async def voice_ws(websocket: WebSocket) -> None:
await _safe_close(websocket)
async def _handle_stt_result_control(
websocket: WebSocket,
*,
session_id: str,
principal: Principal,
voice_preset: VoicePreset,
ctrl: dict[str, object],
audio_started_at: float | None = None,
audio_ended_at: float | None = None,
last_audio_end_at: float | None = None,
) -> None:
learner_text = str(ctrl.get("text") or "").strip()
transcript_final = _safe_bool(ctrl.get("final"))
silence_ms = _safe_int(ctrl.get("silence_ms"))
if silence_ms is None and last_audio_end_at is not None and audio_started_at is not None:
silence_ms = max(0, int((audio_started_at - last_audio_end_at) * 1000))
provider_events = _safe_provider_events(ctrl.get("provider_events"))
decision = voice_svc.assess_end_of_turn(
transcript_text=learner_text,
transcript_final=bool(transcript_final),
silence_ms=silence_ms,
)
await _safe_send_json(
websocket,
{
"type": "eot",
"ready": decision.ready,
"reason": decision.reason,
"silence_ms": decision.silence_ms,
"threshold_ms": decision.threshold_ms,
},
)
if not decision.ready:
await _safe_send_json(websocket, {"type": "state", "state": "listening"})
return
await _safe_send_json(websocket, {"type": "state", "state": "thinking"})
await _safe_send_json(
websocket,
{"type": "transcript", "text": learner_text, "final": True, "speaker": "counselor"},
)
await _run_turn_and_speak(
websocket,
session_id=session_id,
principal=principal,
voice_preset=voice_preset,
learner_text=learner_text,
duration_s=_elapsed_seconds(audio_started_at, audio_ended_at),
silence_ms=decision.silence_ms,
barge_in=_safe_bool(ctrl.get("barge_in")),
provider_events=provider_events,
)
async def _handle_utterance(
websocket: WebSocket,
*,
@ -279,6 +370,9 @@ async def _handle_utterance(
voice_preset: VoicePreset,
audio: bytes,
fmt: Optional[str],
sample_rate: int | None = None,
channels: int | None = None,
sample_width: int | None = None,
audio_started_at: float | None = None,
audio_ended_at: float | None = None,
silence_ms: int | None = None,
@ -293,10 +387,17 @@ async def _handle_utterance(
# STT begins after the learner stops speaking.
await _safe_send_json(websocket, {"type": "state", "state": "thinking"})
filename, content_type = _audio_meta(fmt)
upload_audio, upload_fmt = _normalize_audio_upload(
audio,
fmt=fmt,
sample_rate=sample_rate,
channels=channels,
sample_width=sample_width,
)
filename, content_type = _audio_meta(upload_fmt)
try:
stt = await voice_service.transcribe(
audio, filename=filename, content_type=content_type
upload_audio, filename=filename, content_type=content_type
)
except VoiceUnavailable as e:
await _safe_send_json(websocket, {"type": "degraded", "reason": str(e)})
@ -308,7 +409,7 @@ async def _handle_utterance(
return
learner_text = stt.text
audio_ref = _voice_audio_ref(audio, fmt)
audio_ref = _voice_audio_ref(upload_audio, upload_fmt)
duration_s = stt.duration or _elapsed_seconds(audio_started_at, audio_ended_at)
speech_rate = _estimate_speech_rate(learner_text, duration_s)
provider_events = _merge_provider_events(provider_events, getattr(stt, "provider_events", []))
@ -342,12 +443,15 @@ async def _run_turn_and_speak(
voice_preset: VoicePreset,
learner_text: str,
audio_ref: str | None = None,
duration_s: float | None = None,
silence_ms: int | None = None,
speech_rate: float | None = None,
barge_in: bool | None = None,
provider_events: list[dict[str, object]] | None = None,
) -> None:
"""Run one counseling turn and stream synthesized client speech."""
if speech_rate is None:
speech_rate = _estimate_speech_rate(learner_text, duration_s)
sess, err = await _load_voice_session(session_id, principal)
if sess is None:
await _safe_send_json(websocket, {"type": "error", "detail": err or "session not found or ended"})
@ -364,10 +468,12 @@ async def _run_turn_and_speak(
card=sess.persona,
state=sess.state,
learner_text=learner_text,
recall_summary=recall.recall_summary,
pinned_facts=recall.pinned_facts,
recent_turns=sess.recent_turns(visible_to="client"),
kb_behavior_cues=kb_cues,
memory=orchestrator.TurnMemory(
recall_summary=recall.recall_summary,
pinned_facts=recall.pinned_facts,
recent_turns=sess.recent_turns(visible_to="client"),
kb_behavior_cues=kb_cues,
),
theory_mode=sess.theory_mode,
)
assert ctx.state_after is not None
@ -656,6 +762,56 @@ def _audio_meta(fmt: Optional[str]) -> tuple[str, str]:
return table.get(f, ("audio.webm", "audio/webm"))
def _normalize_audio_upload(
audio: bytes,
*,
fmt: Optional[str],
sample_rate: int | None = None,
channels: int | None = None,
sample_width: int | None = None,
) -> tuple[bytes, str]:
f = (fmt or "webm").lower().lstrip(".") or "webm"
if f != "pcm":
return audio, f
if sample_width not in (None, 2):
raise ValueError("pcm sample_width must be 2 bytes")
return _wav_from_pcm16(
audio,
sample_rate=_bounded_int(sample_rate, default=48000, minimum=8000, maximum=96000),
channels=_bounded_int(channels, default=1, minimum=1, maximum=2),
), "wav"
def _bounded_int(value: int | None, *, default: int, minimum: int, maximum: int) -> int:
if value is None:
return default
return min(maximum, max(minimum, value))
def _wav_from_pcm16(pcm: bytes, *, sample_rate: int, channels: int) -> bytes:
byte_rate = sample_rate * channels * 2
block_align = channels * 2
data_size = len(pcm)
header = b"".join(
[
b"RIFF",
(36 + data_size).to_bytes(4, "little"),
b"WAVE",
b"fmt ",
(16).to_bytes(4, "little"),
(1).to_bytes(2, "little"),
channels.to_bytes(2, "little"),
sample_rate.to_bytes(4, "little"),
byte_rate.to_bytes(4, "little"),
block_align.to_bytes(2, "little"),
(16).to_bytes(2, "little"),
b"data",
data_size.to_bytes(4, "little"),
]
)
return header + pcm
def _voice_audio_ref(audio: bytes, fmt: Optional[str]) -> str | None:
if not audio:
return None
@ -688,6 +844,13 @@ def _safe_int(value: object) -> int | None:
return None
def _safe_str(value: object) -> str | None:
if isinstance(value, str):
text = value.strip()
return text or None
return None
def _safe_bool(value: object) -> bool | None:
if value is None:
return None