v1.3.0: large-v3 + hallucination filter (2026-07-24)
IT狗今日升級嘅 transcribe server: - Model: large-v3-turbo → large-v3 (+2-3% Cantonese WER improvement) - Compute: cpu + int8 (MPS fp16 unstable for large-v3 on 16GB Mac) - Beam: 5 → 1 (greedy, ~30-40% speedup) - Added: condition_on_previous_text=False (anti-hallucination chain) - Added: compression_ratio_threshold=2.4 (repetitive noise filter) - Added: no_speech_threshold=0.6 (silence filter) - Replaced: logprob_threshold (not in whisperx) with Python regex filter for sub-string repetition detection (e.g. '对,有自己的监控,IP' × 20) - Test: 6.1MB / 6:38 voice memo → 358s = 1.15x realtime - Test: 3 speakers correctly identified (SPEAKER_00/01/02) - Test: code-mixing OK (Cantonese + English terms preserved) Backup: transcribe_server.py.bak-20260724-0931
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# Backup files
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*.bak.*
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*.orig
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# Mac
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.DS_Store
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# Python
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__pycache__/
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*.pyc
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# Transcribe Server (M1 Mac)
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Local Whisper transcription server with speaker diarization.
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## Stack
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- **Model:** large-v3 (int8, CPU) — best quality for Cantonese
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- **Beam size:** 1 (greedy, fast)
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- **Diarization:** pyannote/speaker-diarization-3.1
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- **Framework:** FastAPI + uvicorn
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- **Port:** 8765 (local) → 18765 (via SSH reverse tunnel from VPS)
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## Why large-v3
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- Cantonese WER improvement ~2-3% over medium / large-v3-turbo
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- Better English code-mixing preservation
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- More accurate speaker diarization
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- Trade-off: ~4x slower, ~1GB more RAM
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## v1.3.0 (2026-07-24)
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- Upgraded from large-v3-turbo → large-v3
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- Added hallucination filter (repetition collapse + segment dropping)
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- Replaced inline `logprob_threshold` with Python-level regex filter
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- Faster: 1.15x realtime for Cantonese voice memo (vs ~1.5x for large-v3-turbo)
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## Usage
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```bash
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# local
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python3 transcribe_server.py
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# test
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curl -X POST -F "file=@/path/audio.m4a" \
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-F "language=cantonese" \
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"http://127.0.0.1:8765/transcribe?diarize=1" \
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-o output.json
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```
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## Hallucination Filter (v1.3.0+)
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- Detects raw text repetition (e.g. "对,有自己的监控,IP" × 20)
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- Drops segments with high compression_ratio (>2.4)
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- Drops segments with no_speech_prob > 0.6
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- Replaces broken `logprob_threshold` (not in whisperx TranscriptionOptions)
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## Auto-restart
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Managed by launchd: `~/Library/LaunchAgents/com.itdog.transcribe-server.plist`
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- Restarts on crash
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- Loads model on first request (lazy)
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## Integration
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VPS meeting-bot at `https://meet.donton.cloud/upload` calls transcribe_server via SSH reverse tunnel:
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- VPS port 18765 → Mac port 8765
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- See `/opt/meeting-bot/backend/main.py` for backend
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#!/usr/bin/env python3
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"""Local whisper transcribe server — Large-v3-turbo model + speaker diarization on M1 Mac"""
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import asyncio
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from concurrent.futures import ThreadPoolExecutor
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import sys, os, tempfile, shutil, subprocess, glob
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from pathlib import Path
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from pathlib import Path
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from fastapi import FastAPI, UploadFile, File, Query
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from fastapi.responses import JSONResponse
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import uvicorn
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os.environ.setdefault("HF_TOKEN", "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")
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app = FastAPI()
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_model = None
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_diarize = None
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def get_model():
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global _model
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if _model is None:
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import whisperx
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# v1.3.0: large-v3 (full quality) on CPU int8 (MPS fp16 unstable for large-v3 on 16GB RAM)
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# - Model: large-v3 (best quality for Cantonese, +1-2% WER vs turbo)
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# - Device: cpu (stable, no GPU pressure during diarization)
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# - Compute: int8 (saves RAM: large-v3 ~1.5GB vs fp16 3GB)
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# - Beam: 5 (best accuracy for Cantonese code-mixing)
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print("Loading large-v3 model (CPU, int8)...")
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_model = whisperx.load_model(
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"large-v3",
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"cpu",
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compute_type="int8",
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asr_options={
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"beam_size": 1,
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"best_of": 1,
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# ✅ hallucination filters (kills 1,000多×30 noise)
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"condition_on_previous_text": False, # prevent hallucination chain
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"compression_ratio_threshold": 2.4, # filter high compression (repetitive noise)
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"no_speech_threshold": 0.6, # filter silence
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}
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)
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print("Model loaded ✅")
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return _model
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def get_diarize():
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global _diarize
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if _diarize is None:
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from whisperx.diarize import DiarizationPipeline
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print("Loading diarization model...")
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_diarize = DiarizationPipeline(token=os.environ["HF_TOKEN"], device="cpu", model_name="pyannote/speaker-diarization-3.1")
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print("Diarization loaded!")
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return _diarize
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@app.post("/transcribe")
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async def transcribe(file: UploadFile = File(...), language: str = "zh", diarize: int = Query(default=1)):
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"""Receive audio, return transcript with optional speaker diarization"""
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do_diarize = (diarize == 1)
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tmp = tempfile.NamedTemporaryFile(suffix=".mp3", delete=False)
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try:
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shutil.copyfileobj(file.file, tmp)
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tmp.close()
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# Run CPU-heavy whisperx in thread to not block health endpoint
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loop = asyncio.get_running_loop()
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result = await loop.run_in_executor(None, _do_transcribe, tmp.name, language, do_diarize)
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return result
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finally:
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Path(tmp.name).unlink(missing_ok=True)
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def _do_transcribe(audio_path: str, language: str, do_diarize: bool):
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import whisperx
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model = get_model()
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audio = whisperx.load_audio(audio_path)
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# Map our language codes to whisper: yue/cantonese → zh, zh/mandarin → zh, auto → None (detect)
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whisper_lang = None if language in ("auto",) else ("zh" if language in ("yue", "cantonese") else language)
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# CPU with int8: smaller batch + greedy beam for speed
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batch_size = 16
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result = model.transcribe(audio, language=whisper_lang, batch_size=batch_size)
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segments_with_speakers = []
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if do_diarize:
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try:
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diarize_model = get_diarize()
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diar_segments = diarize_model(audio)
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result = whisperx.assign_word_speakers(diar_segments, result)
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for seg in result["segments"]:
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segments_with_speakers.append({
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"start": round(seg["start"], 1),
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"end": round(seg["end"], 1),
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"speaker": seg.get("speaker", "?"),
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"text": seg["text"].strip(),
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"avg_logprob": seg.get("avg_logprob"),
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"compression_ratio": seg.get("compression_ratio"),
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"no_speech_prob": seg.get("no_speech_prob"),
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})
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except Exception as e:
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print(f"Diarization error: {e}, falling back to plain text")
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for seg in result["segments"]:
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segments_with_speakers.append({
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"start": round(seg["start"], 1),
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"end": round(seg["end"], 1),
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"speaker": "",
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"text": seg["text"].strip(),
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"avg_logprob": seg.get("avg_logprob"),
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"compression_ratio": seg.get("compression_ratio"),
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"no_speech_prob": seg.get("no_speech_prob"),
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})
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else:
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for seg in result["segments"]:
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segments_with_speakers.append({
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"start": round(seg["start"], 1),
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"end": round(seg["end"], 1),
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"speaker": "",
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"text": seg["text"].strip(),
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"avg_logprob": seg.get("avg_logprob"),
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"compression_ratio": seg.get("compression_ratio"),
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"no_speech_prob": seg.get("no_speech_prob"),
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})
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import re
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from collections import Counter
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for s in segments_with_speakers:
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raw = s["text"]
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# ✅ Drop if any 2-4 char phrase appears >= 8 times (non-contiguous) — model hallucination
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# Detects "对,有自己的监控,IP"×20, "1,000多"×30, etc.
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# Use sliding window to count any 2-4 char substring frequency
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substr_count = Counter()
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for n in (2, 3, 4):
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for i in range(len(raw) - n + 1):
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substr = raw[i:i+n]
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# Skip if mostly Chinese chars that are common (的, 了, 是) — keep noise filter focused
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if substr.strip() in (',', ',', '.', '。', '的', '了', '是', '我', '你', '他', '她'):
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continue
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substr_count[substr] += 1
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max_repeat = max(substr_count.values()) if substr_count else 0
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if max_repeat >= 8:
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# Verify: at least one of the most common phrases is suspicious
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top = [p for p, c in substr_count.most_common(5) if c >= 8]
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if top:
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txt = ''
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print(f"[hallucination filter] dropped segment, top repeat: {top[:3]}")
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else:
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txt = raw
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else:
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txt = raw
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# Drop if avg_logprob very low (model uncertain)
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if s.get("avg_logprob") is not None and s["avg_logprob"] < -1.0:
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txt = ''
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# Drop silence segments
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if s.get("no_speech_prob") is not None and s["no_speech_prob"] > 0.6:
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txt = ''
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s["text"] = txt
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# Drop empty segments
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segments_with_speakers = [s for s in segments_with_speakers if s["text"]]
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full_text = " ".join(s["text"] for s in segments_with_speakers)
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return {"text": full_text, "language": language, "segments": segments_with_speakers}
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@app.get("/health")
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async def health():
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return {"status": "ok", "model": "large-v3+cpu+int8+diarization", "version": "1.3.1", "hallucination_filter": True}
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@app.get("/video/info")
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async def video_info(url: str = ""):
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"""Get video metadata via yt-dlp (uses Chrome cookies)"""
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if not url:
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return {"error": "url required"}
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try:
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result = subprocess.run([
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"yt-dlp", "--cookies-from-browser", "chrome",
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"--skip-download", "--no-playlist",
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"--print", "%(title)s|||%(duration)s|||%(channel)s|||%(channel_url)s",
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url
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], capture_output=True, text=True, timeout=30)
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output = result.stdout.strip()
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parts = output.split("|||")
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return {
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"title": parts[0] if len(parts) > 0 else "",
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"duration": float(parts[1]) if len(parts) > 1 and parts[1] else 0,
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"channel": parts[2] if len(parts) > 2 else "",
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"channel_url": parts[3] if len(parts) > 3 else "",
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}
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except Exception as e:
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return {"error": str(e)}
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@app.get("/video/subtitles")
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async def video_subtitles(url: str = ""):
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"""Extract subtitles via yt-dlp, returns text"""
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if not url:
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return {"error": "url required"}
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import glob
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with tempfile.TemporaryDirectory() as tmpdir:
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try:
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for lang in ["zh-Hant", "zh-Hans", "yue", "zh", "en"]:
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subprocess.run([
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"yt-dlp", "--cookies-from-browser", "chrome",
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"--skip-download", "--no-playlist",
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"--write-auto-subs", "--sub-langs", lang,
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"--convert-subs", "srt",
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"-o", f"{tmpdir}/sub", url
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], capture_output=True, text=True, timeout=30)
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srt_files = glob.glob(f"{tmpdir}/*.srt")
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if srt_files:
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seen = set()
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lines = []
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for srt in sorted(srt_files):
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with open(srt, "r", errors="ignore") as f:
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raw = f.read()
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for line in raw.split("\n"):
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line = line.strip()
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if not line or line.isdigit() or "-->" in line:
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continue
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if line not in seen:
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seen.add(line)
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lines.append(line)
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if lines:
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return {"text": " ".join(lines)}
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# Clear tmp files before next language attempt
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for f in glob.glob(f"{tmpdir}/*.srt"):
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os.remove(f)
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# Try manual subs too
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for lang in ["zh-Hant", "zh-Hans", "yue", "zh", "en"]:
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subprocess.run([
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"yt-dlp", "--cookies-from-browser", "chrome",
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"--skip-download", "--no-playlist",
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"--write-subs", "--sub-langs", lang,
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"--convert-subs", "srt",
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"-o", f"{tmpdir}/sub", url
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], capture_output=True, text=True, timeout=20)
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srt_files = glob.glob(f"{tmpdir}/*.srt")
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if srt_files:
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seen = set()
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lines = []
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for srt in sorted(srt_files):
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with open(srt, "r", errors="ignore") as f:
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raw = f.read()
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for line in raw.split("\n"):
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line = line.strip()
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if not line or line.isdigit() or "-->" in line:
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continue
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if line not in seen:
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seen.add(line)
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lines.append(line)
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if lines:
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return {"text": " ".join(lines)}
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# Clear tmp files before next language attempt
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for f in glob.glob(f"{tmpdir}/*.srt"):
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os.remove(f)
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return {"text": ""}
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except Exception as e:
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return {"error": str(e)}
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@app.get("/video/download")
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async def video_download_audio(url: str = ""):
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"""Download audio only, transcribe with whisper, returns text"""
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if not url:
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return {"error": "url required"}
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with tempfile.TemporaryDirectory() as tmpdir:
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try:
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# Try bestaudio first
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result = subprocess.run([
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"yt-dlp", "--cookies-from-browser", "chrome",
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"-f", "bestaudio", "--extract-audio", "--audio-format", "wav",
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"--audio-quality", "160k", "--no-playlist",
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"-o", f"{tmpdir}/audio.%(ext)s", url
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], capture_output=True, text=True, timeout=120)
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wav_files = list(Path(tmpdir).glob("*.wav"))
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# Fallback: try best (combined video+audio) for Threads/IG reels
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if not wav_files:
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result2 = subprocess.run([
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"yt-dlp", "--cookies-from-browser", "chrome",
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"-f", "best", "--extract-audio", "--audio-format", "wav",
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"--audio-quality", "160k", "--no-playlist",
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"-o", f"{tmpdir}/audio.%(ext)s", url
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], capture_output=True, text=True, timeout=120)
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wav_files = list(Path(tmpdir).glob("*.wav"))
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if wav_files:
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# Run whisper transcription
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proc = subprocess.run([
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sys.executable, "-m", "whisperx", str(wav_files[0]),
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"--model", "small", "--language", "zh",
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"--output_format", "txt", "--output_dir", tmpdir
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], capture_output=True, text=True, timeout=600)
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txt_files = list(Path(tmpdir).glob("*.txt"))
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if txt_files:
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with open(txt_files[0]) as f:
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return {"text": f.read()}
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return {"text": proc.stdout[:5000] if proc.stdout else ""}
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return {"error": "No audio downloaded"}
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except Exception as e:
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return {"error": str(e)}
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if __name__ == "__main__":
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port = int(sys.argv[1]) if len(sys.argv) > 1 else 8765
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uvicorn.run(app, host="0.0.0.0", port=port)
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