Delete pipeline.py
Browse files- pipeline.py +0 -260
pipeline.py
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"""
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Core pipeline: ASR (Whisper) + MT (NLLB-200) functions.
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TTS is handled by tts_engine.py.
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"""
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import torch
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import numpy as np
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import re
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import time
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import os
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import subprocess
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import tempfile
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import logging
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import soundfile as sf
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logger = logging.getLogger(__name__)
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DEVICE = "cuda" if torch.cuda.is_available() else "cpu"
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TORCH_DTYPE = torch.float16 if torch.cuda.is_available() else torch.float32
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# Models (loaded once at startup)
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asr_pipe = None
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mt_tokenizer = None
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mt_model = None
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tts_pipe_local = None # Local TTS for Yoruba/Hausa/Igbo/Zulu
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def load_models():
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"""Load all models at startup."""
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global asr_pipe, mt_tokenizer, mt_model, tts_pipe_local
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from transformers import (
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pipeline as hf_pipeline,
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AutoTokenizer,
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AutoModelForSeq2SeqLM,
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)
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print(f"Device: {DEVICE} | Dtype: {TORCH_DTYPE}")
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print("Loading models...")
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# ASR
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ASR_MODEL_ID = "PlotweaverAI/whisper-small-de-en"
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print(f" Loading ASR: {ASR_MODEL_ID}")
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asr_pipe = hf_pipeline(
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"automatic-speech-recognition",
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model=ASR_MODEL_ID,
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device=DEVICE,
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torch_dtype=TORCH_DTYPE,
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)
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print(" ASR loaded")
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# MT
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MT_MODEL_ID = "PlotweaverAI/nllb-200-distilled-600M-african-6lang"
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print(f" Loading MT: {MT_MODEL_ID}")
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mt_tokenizer = AutoTokenizer.from_pretrained(MT_MODEL_ID)
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mt_model = AutoModelForSeq2SeqLM.from_pretrained(
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MT_MODEL_ID, torch_dtype=TORCH_DTYPE
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).to(DEVICE)
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mt_tokenizer.src_lang = "eng_Latn"
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print(" MT loaded")
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# Local TTS (Yoruba)
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TTS_MODEL_ID = "PlotweaverAI/yoruba-mms-tts-new"
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print(f" Loading local TTS: {TTS_MODEL_ID}")
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tts_pipe_local = hf_pipeline(
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"text-to-speech",
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model=TTS_MODEL_ID,
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device=DEVICE,
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torch_dtype=TORCH_DTYPE,
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)
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print(" Local TTS loaded")
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# Diagnostics
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print(f"\n=== Device diagnostics ===")
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print(f"CUDA available: {torch.cuda.is_available()}")
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if torch.cuda.is_available():
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print(f"CUDA device: {torch.cuda.get_device_name(0)}")
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print(f"ASR on: {next(asr_pipe.model.parameters()).device}")
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print(f"MT on: {next(mt_model.parameters()).device}")
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print(f"TTS on: {next(tts_pipe_local.model.parameters()).device}")
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print(f"YourVoic API key: {'set' if os.environ.get('YOURVOIC_API_KEY') else 'NOT SET'}")
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print(f"==========================\n")
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print("All models loaded!")
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# ---- Text Processing ----
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def split_into_sentences(text):
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"""Split raw ASR text into individual sentences."""
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text = text.strip()
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if not text:
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return []
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text = '. '.join(s.strip().capitalize() for s in text.split('. ') if s.strip())
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if re.search(r'[.!?]', text):
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sentences = re.split(r'(?<=[.!?])\s+', text)
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return [s.strip() for s in sentences if s.strip()]
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words = text.split()
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MAX_WORDS = 12
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sentences = []
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for i in range(0, len(words), MAX_WORDS):
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chunk = ' '.join(words[i:i + MAX_WORDS])
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if not chunk.endswith(('.', '!', '?')):
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chunk += '.'
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chunk = chunk[0].upper() + chunk[1:] if len(chunk) > 1 else chunk.upper()
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sentences.append(chunk)
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return sentences
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# ---- ASR ----
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def transcribe(audio_array, sample_rate=16000):
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"""ASR: English audio to text. Handles both short and long audio."""
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if len(audio_array) < 1600:
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return ""
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duration_s = len(audio_array) / sample_rate
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if sample_rate != 16000:
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import torchaudio.functional as F_audio
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audio_tensor = torch.from_numpy(audio_array).float()
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audio_tensor = F_audio.resample(audio_tensor, sample_rate, 16000)
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audio_array = audio_tensor.numpy()
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sample_rate = 16000
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if duration_s <= 28:
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result = asr_pipe(
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{"raw": audio_array, "sampling_rate": sample_rate},
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return_timestamps=False,
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)
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return result["text"].strip()
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# Long-form: native Whisper generate
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model = asr_pipe.model
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processor = asr_pipe.feature_extractor
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tokenizer = asr_pipe.tokenizer
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inputs = processor(
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audio_array, sampling_rate=16000, return_tensors="pt",
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truncation=False, padding="longest", return_attention_mask=True,
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)
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input_features = inputs.input_features.to(DEVICE, dtype=TORCH_DTYPE)
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attention_mask = inputs.attention_mask.to(DEVICE) if "attention_mask" in inputs else None
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generate_kwargs = {"return_timestamps": True, "language": "en", "task": "transcribe"}
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if attention_mask is not None:
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generate_kwargs["attention_mask"] = attention_mask
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with torch.no_grad():
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predicted_ids = model.generate(input_features, **generate_kwargs)
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transcription = tokenizer.batch_decode(predicted_ids, skip_special_tokens=True)[0]
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return transcription.strip()
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# ---- MT ----
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def translate_sentence(text, target_nllb_code, fast=True, max_length=256):
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"""Translate a single sentence from English to target language."""
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inputs = mt_tokenizer(text, return_tensors="pt", truncation=True).to(DEVICE)
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tgt_lang_id = mt_tokenizer.convert_tokens_to_ids(target_nllb_code)
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generate_kwargs = {
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"forced_bos_token_id": tgt_lang_id,
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"repetition_penalty": 1.5,
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"no_repeat_ngram_size": 3,
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}
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if fast:
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generate_kwargs.update({"max_length": 128, "num_beams": 1, "do_sample": False})
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else:
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generate_kwargs.update({"max_length": max_length, "num_beams": 4, "early_stopping": True})
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with torch.no_grad():
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output_ids = mt_model.generate(**inputs, **generate_kwargs)
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return mt_tokenizer.decode(output_ids[0], skip_special_tokens=True)
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def translate_text(text, target_nllb_code, fast=True):
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"""Split and translate full text sentence-by-sentence."""
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sentences = split_into_sentences(text)
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if not sentences:
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return "", [], []
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translations = []
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for s in sentences:
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yo = translate_sentence(s, target_nllb_code, fast=fast)
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translations.append(yo)
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return ' '.join(translations), sentences, translations
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# ---- Video Processing ----
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def extract_audio_from_video(video_path, output_path, target_sr=16000):
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"""Extract audio track from video as 16kHz mono WAV."""
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cmd = [
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"ffmpeg", "-y", "-i", video_path,
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"-vn", "-acodec", "pcm_s16le", "-ar", str(target_sr), "-ac", "1",
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output_path,
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]
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result = subprocess.run(cmd, capture_output=True, text=True)
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if result.returncode != 0:
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raise RuntimeError(f"ffmpeg extraction failed: {result.stderr[:200]}")
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return output_path
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def get_media_duration(path):
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"""Get duration in seconds."""
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cmd = [
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"ffprobe", "-v", "error",
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"-show_entries", "format=duration",
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"-of", "default=noprint_wrappers=1:nokey=1", path,
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]
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result = subprocess.run(cmd, capture_output=True, text=True)
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if result.returncode != 0:
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raise RuntimeError(f"ffprobe failed: {result.stderr[:200]}")
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return float(result.stdout.strip())
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def stretch_audio_to_duration(input_path, output_path, target_duration_s):
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"""Stretch/compress audio to match target duration."""
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current_duration = get_media_duration(input_path)
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if current_duration <= 0:
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raise RuntimeError("Invalid audio duration")
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ratio = current_duration / target_duration_s
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filters = []
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remaining = ratio
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while remaining > 2.0:
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filters.append("atempo=2.0")
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remaining /= 2.0
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while remaining < 0.5:
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filters.append("atempo=0.5")
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remaining /= 0.5
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filters.append(f"atempo={remaining:.4f}")
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cmd = ["ffmpeg", "-y", "-i", input_path, "-filter:a", ",".join(filters), output_path]
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result = subprocess.run(cmd, capture_output=True, text=True)
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if result.returncode != 0:
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raise RuntimeError(f"ffmpeg tempo failed: {result.stderr[:200]}")
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return output_path
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def mux_video_audio(video_path, audio_path, output_path, extend_video=False, target_duration=None):
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"""Combine video with new audio. Optionally extend video by freezing last frame."""
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if extend_video and target_duration:
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cmd = [
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"ffmpeg", "-y", "-i", video_path, "-i", audio_path,
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"-filter_complex", f"[0:v]tpad=stop_mode=clone:stop_duration={target_duration}[v]",
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"-map", "[v]", "-map", "1:a:0",
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"-c:v", "libx264", "-preset", "fast", "-c:a", "aac",
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"-t", str(target_duration), output_path,
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]
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else:
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cmd = [
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"ffmpeg", "-y", "-i", video_path, "-i", audio_path,
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"-c:v", "copy", "-c:a", "aac",
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"-map", "0:v:0", "-map", "1:a:0", "-shortest", output_path,
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]
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result = subprocess.run(cmd, capture_output=True, text=True)
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if result.returncode != 0:
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raise RuntimeError(f"ffmpeg mux failed: {result.stderr[:200]}")
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return output_path
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