Files
speech/app/asr/workers/faster_whisper_worker.py
du5t a32b0acb09 Fix broken file transcription and speaker diarization
qwen3 backend: model list pointed at nonexistent HF repos (Qwen3-ASR-2B/8B
don't exist), and inference went through the generic HF ASR pipeline with
Whisper-only options (return_timestamps, task/language generate_kwargs) that
Qwen3-ASR's chat-style architecture doesn't support. Switched to the real
0.6B/1.7B-hf checkpoints and drive them via
processor.apply_transcription_request() + model.generate(). Also added an
ffmpeg pre-conversion step since the model's feature extractor can't decode
m4a via librosa.

faster-whisper backend: speaker diarization was broken by two pyannote.audio
API changes it hadn't caught up with (use_auth_token= renamed to token=, and
pipeline() now returns a DiarizeOutput wrapper instead of an Annotation
directly). Also pinned LD_LIBRARY_PATH for that worker so it picks up its own
venv's cuDNN instead of the older one shadowing it via the container's global
LD_LIBRARY_PATH, which crashed pyannote's GPU init.

Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
2026-07-23 16:31:55 +09:00

282 lines
9.6 KiB
Python

from __future__ import annotations
import argparse
import tempfile
from pathlib import Path
from typing import Any, Dict, List, Optional, Union
import uvicorn
from fastapi import FastAPI, File, Form, HTTPException, UploadFile
from faster_whisper import WhisperModel
import sys
sys.path.insert(0, "/app")
from asr.config import (
COMPUTE_TYPE,
CUSTOM_MODEL_DIR,
DEFAULT_MODEL,
DEVICE,
MODEL_CACHE,
PYANNOTE_HF_TOKEN,
ensure_runtime_dirs,
resolve_custom_model_path,
)
import os
# 컨테이너는 983 유저로 실행되는데 HOME(/app)이 root 소유라, pyannote.audio가 쓰는
# numba/matplotlib/torch가 각자 ~/.cache, ~/.config 아래에 쓰려다 실패한다.
os.environ["HOME"] = "/tmp" # 컨테이너 기본 HOME=/app은 983 유저가 쓰기 불가 (setdefault로는 덮어쓰기 안 됨)
os.environ.setdefault("NUMBA_CACHE_DIR", "/tmp/numba_cache")
app = FastAPI(title="ASR Faster-Whisper Worker")
_MODEL_CACHE: Dict[str, WhisperModel] = {}
_DIARIZATION_PIPELINE: Any = None
@app.on_event("startup")
def startup() -> None:
ensure_runtime_dirs()
@app.get("/health")
def health() -> Dict[str, Any]:
return {
"status": "ok",
"device": DEVICE,
"compute_type": COMPUTE_TYPE,
"diarization_available": bool(PYANNOTE_HF_TOKEN),
}
@app.post("/transcribe")
async def transcribe(
file: UploadFile = File(...),
model: str = Form(DEFAULT_MODEL),
custom_model_path: Optional[str] = Form(None),
language: Optional[str] = Form(None),
task: str = Form("transcribe"),
beam_size: int = Form(5),
temperature: float = Form(0.0),
word_timestamps: bool = Form(False),
diarize: bool = Form(False),
num_speakers: Optional[int] = Form(None),
min_speakers: Optional[int] = Form(None),
max_speakers: Optional[int] = Form(None),
no_repeat_ngram_size: int = Form(0),
repetition_penalty: float = Form(1.0),
compression_ratio_threshold: float = Form(2.4),
log_prob_threshold: float = Form(-1.0),
no_speech_threshold: float = Form(0.6),
condition_on_previous_text: bool = Form(True),
) -> Dict[str, Any]:
ensure_runtime_dirs()
suffix = Path(file.filename or "upload.bin").suffix or ".bin"
with tempfile.NamedTemporaryFile(delete=False, suffix=suffix) as tmp:
tmp.write(await file.read())
tmp_path = Path(tmp.name)
try:
resolved_model = resolve_custom_model_path(custom_model_path) or model
result = _transcribe(
audio_path=tmp_path,
model_name=resolved_model,
language=language or None,
task=task,
beam_size=beam_size,
temperature=temperature,
word_timestamps=word_timestamps,
no_repeat_ngram_size=no_repeat_ngram_size,
repetition_penalty=repetition_penalty,
compression_ratio_threshold=compression_ratio_threshold,
log_prob_threshold=log_prob_threshold,
no_speech_threshold=no_speech_threshold,
condition_on_previous_text=condition_on_previous_text,
)
if diarize:
result = _apply_diarization(
audio_path=tmp_path,
result=result,
num_speakers=num_speakers,
min_speakers=min_speakers,
max_speakers=max_speakers,
)
return result
except HTTPException:
raise
except Exception as exc:
raise HTTPException(status_code=500, detail=f"Worker failure: {exc}") from exc
finally:
tmp_path.unlink(missing_ok=True)
def _load_model(model_name: str) -> WhisperModel:
if model_name not in _MODEL_CACHE:
_MODEL_CACHE[model_name] = WhisperModel(
model_name,
device=DEVICE,
compute_type=COMPUTE_TYPE,
download_root=str(MODEL_CACHE),
)
return _MODEL_CACHE[model_name]
def _transcribe(
audio_path: Path,
model_name: str,
language: Optional[str],
task: str,
beam_size: int,
temperature: float,
word_timestamps: bool,
no_repeat_ngram_size: int,
repetition_penalty: float,
compression_ratio_threshold: float,
log_prob_threshold: float,
no_speech_threshold: float,
condition_on_previous_text: bool,
) -> Dict[str, Any]:
model = _load_model(model_name)
temperatures: Union[float, List[float]] = [0.0, 0.2, 0.4, 0.6, 0.8, 1.0] if temperature == 0.0 else temperature
segments_iter, info = model.transcribe(
str(audio_path),
language=language,
task=task,
beam_size=beam_size,
temperature=temperatures,
word_timestamps=word_timestamps,
no_repeat_ngram_size=no_repeat_ngram_size,
repetition_penalty=repetition_penalty,
compression_ratio_threshold=compression_ratio_threshold,
log_prob_threshold=log_prob_threshold,
no_speech_threshold=no_speech_threshold,
condition_on_previous_text=condition_on_previous_text,
)
segment_list = []
full_text = []
for seg in segments_iter:
seg_dict: Dict[str, Any] = {
"id": seg.id,
"start": round(float(seg.start), 3),
"end": round(float(seg.end), 3),
"text": seg.text,
"avg_logprob": round(float(seg.avg_logprob), 4) if seg.avg_logprob is not None else None,
"compression_ratio": round(float(seg.compression_ratio), 4) if seg.compression_ratio is not None else None,
"no_speech_prob": round(float(seg.no_speech_prob), 4) if seg.no_speech_prob is not None else None,
}
if word_timestamps and getattr(seg, "words", None):
seg_dict["words"] = [
{
"start": round(float(w.start), 3),
"end": round(float(w.end), 3),
"word": w.word,
"probability": round(float(w.probability), 4),
}
for w in seg.words
]
segment_list.append(seg_dict)
full_text.append(seg.text.strip())
duration = None
try:
duration = round(float(info.duration), 3)
except Exception:
pass
return {
"backend": "faster-whisper",
"model": model_name,
"language": getattr(info, "language", language),
"language_probability": round(float(getattr(info, "language_probability", 0) or 0), 4),
"duration": duration,
"text": " ".join(x for x in full_text if x),
"segments": segment_list,
"diarized": False,
}
def _get_diarization_pipeline():
global _DIARIZATION_PIPELINE
if _DIARIZATION_PIPELINE is None:
try:
from pyannote.audio import Pipeline
import torch
except ImportError as exc:
raise RuntimeError("pyannote.audio가 설치되지 않았습니다.") from exc
if not PYANNOTE_HF_TOKEN:
raise RuntimeError("화자 분리를 사용하려면 PYANNOTE_HF_TOKEN 환경변수를 설정하세요.")
_DIARIZATION_PIPELINE = Pipeline.from_pretrained(
"pyannote/speaker-diarization-3.1",
token=PYANNOTE_HF_TOKEN,
)
if DEVICE == "cuda":
import torch
_DIARIZATION_PIPELINE = _DIARIZATION_PIPELINE.to(torch.device("cuda"))
return _DIARIZATION_PIPELINE
def _apply_diarization(
audio_path: Path,
result: Dict[str, Any],
num_speakers: Optional[int],
min_speakers: Optional[int],
max_speakers: Optional[int],
) -> Dict[str, Any]:
pipeline = _get_diarization_pipeline()
kwargs: Dict[str, Any] = {}
if num_speakers is not None:
kwargs["num_speakers"] = num_speakers
else:
if min_speakers is not None:
kwargs["min_speakers"] = min_speakers
if max_speakers is not None:
kwargs["max_speakers"] = max_speakers
# pyannote.audio 4.x: pipeline() returns a DiarizeOutput dataclass instead of
# an Annotation directly; the itertracks-capable Annotation is .speaker_diarization.
diarization = pipeline(str(audio_path), **kwargs).speaker_diarization
segments = result.get("segments", [])
for seg in segments:
seg_start = float(seg.get("start", 0))
seg_end = float(seg.get("end", 0))
speaker_times: Dict[str, float] = {}
for turn, _, speaker in diarization.itertracks(yield_label=True):
overlap_start = max(seg_start, turn.start)
overlap_end = min(seg_end, turn.end)
if overlap_end > overlap_start:
speaker_times[speaker] = speaker_times.get(speaker, 0.0) + (overlap_end - overlap_start)
seg["speaker"] = max(speaker_times, key=speaker_times.get) if speaker_times else "UNKNOWN"
lines = []
current_speaker: Optional[str] = None
current_texts: list = []
for seg in segments:
speaker = seg.get("speaker", "UNKNOWN")
text = seg.get("text", "").strip()
if not text:
continue
if speaker != current_speaker:
if current_texts and current_speaker is not None:
lines.append(f"[{current_speaker}]: {' '.join(current_texts)}")
current_speaker = speaker
current_texts = [text]
else:
current_texts.append(text)
if current_texts and current_speaker is not None:
lines.append(f"[{current_speaker}]: {' '.join(current_texts)}")
result["text"] = "\n".join(lines)
result["diarized"] = True
return result
if __name__ == "__main__":
parser = argparse.ArgumentParser()
parser.add_argument("--host", default="0.0.0.0")
parser.add_argument("--port", type=int, default=8001)
args = parser.parse_args()
ensure_runtime_dirs()
uvicorn.run(app, host=args.host, port=args.port)