meetingmind-gpu / app.py
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"""
Slim GPU service for HF Inference Endpoints.
Exposes /diarize, /embed, /transcribe, and /transcribe/stream endpoints.
"""
import io
import json
import logging
import os
import re
import threading
from contextlib import asynccontextmanager
import numpy as np
import soundfile as sf
import librosa
import torch
from fastapi import FastAPI, File, Form, UploadFile
from fastapi.responses import JSONResponse
from pydub import AudioSegment
from sse_starlette.sse import EventSourceResponse
logger = logging.getLogger("gpu_service")
# ---------------------------------------------------------------------------
# Config
# ---------------------------------------------------------------------------
HF_TOKEN = os.environ.get("HF_TOKEN", "")
PYANNOTE_MODEL = "pyannote/speaker-diarization-community-1"
FUNASR_MODEL = "iic/speech_campplus_sv_zh-cn_16k-common"
PYANNOTE_MIN_SPEAKERS = int(os.environ.get("PYANNOTE_MIN_SPEAKERS", "1"))
PYANNOTE_MAX_SPEAKERS = int(os.environ.get("PYANNOTE_MAX_SPEAKERS", "10"))
TARGET_SR = 16000
# ---------------------------------------------------------------------------
# Singletons
# ---------------------------------------------------------------------------
_diarize_pipeline = None
_embed_model = None
_voxtral_model = None
_voxtral_processor = None
VOXTRAL_MODEL_ID = "mistralai/Voxtral-Mini-4B-Realtime-2602"
# Markers to strip from Voxtral output
_MARKER_RE = re.compile(r"\[STREAMING_PAD\]|\[STREAMING_WORD\]")
def _load_diarize_pipeline():
global _diarize_pipeline
if _diarize_pipeline is None:
from pyannote.audio import Pipeline as PyannotePipeline
_diarize_pipeline = PyannotePipeline.from_pretrained(
PYANNOTE_MODEL, token=HF_TOKEN
)
_diarize_pipeline = _diarize_pipeline.to(torch.device("cuda"))
return _diarize_pipeline
def _load_embed_model():
global _embed_model
if _embed_model is None:
from funasr import AutoModel
_embed_model = AutoModel(model=FUNASR_MODEL)
return _embed_model
def _load_voxtral():
"""Lazy-load Voxtral model and processor (first call only)."""
global _voxtral_model, _voxtral_processor
if _voxtral_model is None:
from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor
logger.info("Loading Voxtral model %s ...", VOXTRAL_MODEL_ID)
_voxtral_processor = AutoProcessor.from_pretrained(
VOXTRAL_MODEL_ID, trust_remote_code=True
)
_voxtral_model = AutoModelForSpeechSeq2Seq.from_pretrained(
VOXTRAL_MODEL_ID, torch_dtype=torch.float16, trust_remote_code=True
).to("cuda")
logger.info("Voxtral model loaded.")
return _voxtral_model, _voxtral_processor
def _clean_voxtral_text(text: str) -> str:
"""Strip Voxtral streaming markers and collapse whitespace."""
text = _MARKER_RE.sub("", text)
return " ".join(text.split()).strip()
# ---------------------------------------------------------------------------
# Audio helpers
# ---------------------------------------------------------------------------
def prepare_audio(raw_bytes: bytes) -> np.ndarray:
"""Read any audio format -> float32 mono @ 16 kHz."""
audio, sr = sf.read(io.BytesIO(raw_bytes), dtype="float32")
if audio.ndim > 1:
audio = audio.mean(axis=1)
if sr != TARGET_SR:
audio = librosa.resample(audio, orig_sr=sr, target_sr=TARGET_SR)
return audio
def prepare_audio_slice(raw_bytes: bytes, start_time: float, end_time: float) -> np.ndarray:
"""Read audio, slice by time, return float32 mono @ 16 kHz."""
seg = AudioSegment.from_file(io.BytesIO(raw_bytes))
seg = seg[int(start_time * 1000):int(end_time * 1000)]
seg = seg.set_frame_rate(TARGET_SR).set_channels(1).set_sample_width(2)
return np.array(seg.get_array_of_samples(), dtype=np.float32) / 32768.0
# ---------------------------------------------------------------------------
# App
# ---------------------------------------------------------------------------
@asynccontextmanager
async def lifespan(app: FastAPI):
# Warm up diarization pipeline at startup (embedding model lazy-loads)
_load_diarize_pipeline()
yield
app = FastAPI(title="GPU Service (HF Endpoint)", lifespan=lifespan)
@app.get("/health")
async def health():
return {"status": "ok", "gpu_available": torch.cuda.is_available()}
@app.post("/diarize")
async def diarize(
audio: UploadFile = File(...),
min_speakers: int | None = Form(None),
max_speakers: int | None = Form(None),
):
try:
raw = await audio.read()
audio_16k = prepare_audio(raw)
pipeline = _load_diarize_pipeline()
waveform = torch.from_numpy(audio_16k).unsqueeze(0).float()
input_data = {"waveform": waveform, "sample_rate": TARGET_SR}
result = pipeline(
input_data,
min_speakers=min_speakers or PYANNOTE_MIN_SPEAKERS,
max_speakers=max_speakers or PYANNOTE_MAX_SPEAKERS,
)
# pyannote v4 compat
diarization = getattr(result, "speaker_diarization", result)
segments = []
for turn, _, speaker in diarization.itertracks(yield_label=True):
segments.append(
{
"speaker": speaker,
"start": round(turn.start, 3),
"end": round(turn.end, 3),
"duration": round(turn.end - turn.start, 3),
}
)
segments.sort(key=lambda s: s["start"])
return {"segments": segments}
except Exception as e:
return JSONResponse(status_code=500, content={"error": str(e)})
@app.post("/embed")
async def embed(
audio: UploadFile = File(...),
start_time: float | None = Form(None),
end_time: float | None = Form(None),
):
try:
raw = await audio.read()
if start_time is not None and end_time is not None:
audio_16k = prepare_audio_slice(raw, start_time, end_time)
else:
audio_16k = prepare_audio(raw)
model = _load_embed_model()
result = model.generate(input=audio_16k, output_dir=None)
raw_emb = result[0]["spk_embedding"]
if hasattr(raw_emb, "cpu"):
raw_emb = raw_emb.cpu().numpy()
emb = np.array(raw_emb).flatten()
# L2-normalize
norm = np.linalg.norm(emb)
if norm > 0:
emb = emb / norm
return {"embedding": emb.tolist(), "dim": len(emb)}
except Exception as e:
return JSONResponse(status_code=500, content={"error": str(e)})
@app.post("/transcribe")
async def transcribe(
audio: UploadFile = File(...),
prompt: str = Form("Transcribe this audio."),
):
try:
raw = await audio.read()
audio_16k = prepare_audio(raw)
model, processor = _load_voxtral()
inputs = processor(
audios=audio_16k,
sampling_rate=TARGET_SR,
text=prompt,
return_tensors="pt",
).to("cuda")
output_ids = model.generate(**inputs, max_new_tokens=1024)
text = processor.batch_decode(output_ids, skip_special_tokens=True)[0]
text = _clean_voxtral_text(text)
return {"text": text}
except Exception as e:
logger.exception("Transcription failed")
return JSONResponse(status_code=500, content={"error": str(e)})
@app.post("/transcribe/stream")
async def transcribe_stream(
audio: UploadFile = File(...),
prompt: str = Form("Transcribe this audio."),
):
try:
raw = await audio.read()
audio_16k = prepare_audio(raw)
except Exception as e:
logger.exception("Audio preparation failed")
return JSONResponse(status_code=500, content={"error": str(e)})
async def event_generator():
try:
from transformers import TextIteratorStreamer
model, processor = _load_voxtral()
inputs = processor(
audios=audio_16k,
sampling_rate=TARGET_SR,
text=prompt,
return_tensors="pt",
).to("cuda")
streamer = TextIteratorStreamer(
processor.tokenizer, skip_prompt=True, skip_special_tokens=True
)
gen_kwargs = {**inputs, "max_new_tokens": 1024, "streamer": streamer}
thread = threading.Thread(target=model.generate, kwargs=gen_kwargs)
thread.start()
full_text = ""
for chunk in streamer:
chunk = _MARKER_RE.sub("", chunk)
if chunk:
full_text += chunk
yield {"event": "token", "data": json.dumps({"token": chunk})}
thread.join()
full_text = " ".join(full_text.split()).strip()
yield {"event": "done", "data": json.dumps({"text": full_text})}
except Exception as e:
logger.exception("Streaming transcription failed")
yield {"event": "error", "data": json.dumps({"error": str(e)})}
return EventSourceResponse(event_generator())