#!/usr/bin/env python3 import argparse import math from pathlib import Path from typing import List, Tuple import numpy as np import soundfile as sf import webrtcvad from tqdm import tqdm def read_audio(path: Path) -> Tuple[np.ndarray, int]: data, sr = sf.read(str(path)) if data.ndim == 2: data = data.mean(axis=1) return data.astype(np.float32), sr def write_audio(path: Path, data: np.ndarray, sr: int) -> None: path.parent.mkdir(parents=True, exist_ok=True) sf.write(str(path), data, sr) def frame_generator(audio: np.ndarray, sample_rate: int, frame_ms: int) -> List[np.ndarray]: samples_per_frame = int(sample_rate * frame_ms / 1000) num_frames = int(math.ceil(len(audio) / samples_per_frame)) frames = [] for i in range(num_frames): start = i * samples_per_frame end = min((i + 1) * samples_per_frame, len(audio)) frame = audio[start:end] if len(frame) < samples_per_frame: pad = np.zeros(samples_per_frame - len(frame), dtype=audio.dtype) frame = np.concatenate([frame, pad], axis=0) frames.append(frame) return frames def vad_segments(audio: np.ndarray, sample_rate: int, aggressiveness: int = 2, frame_ms: int = 30, min_speech_ms: int = 300, max_silence_ms: int = 300) -> List[Tuple[int, int]]: assert frame_ms in (10, 20, 30), "WebRTC VAD supports 10/20/30 ms" vad = webrtcvad.Vad(aggressiveness) frames = frame_generator(audio, sample_rate, frame_ms) is_speech = [] for frame in frames: int16 = np.clip(frame * 32768.0, -32768, 32767).astype(np.int16).tobytes() is_speech.append(vad.is_speech(int16, sample_rate)) ms_per_frame = frame_ms segments = [] start = None run_silence = 0 for i, speech in enumerate(is_speech): if speech: if start is None: start = i run_silence = 0 else: if start is not None: run_silence += ms_per_frame if run_silence >= max_silence_ms: end = i duration_ms = (end - start) * ms_per_frame if duration_ms >= min_speech_ms: segments.append((start, end)) start = None run_silence = 0 # tail if start is not None: segments.append((start, len(is_speech))) # convert to sample indices spf = int(sample_rate * frame_ms / 1000) sample_segments = [(s * spf, e * spf) for s, e in segments] return sample_segments def hard_cap_segments(segments: List[Tuple[int, int]], max_seconds: float, sample_rate: int) -> List[Tuple[int, int]]: max_len = int(max_seconds * sample_rate) capped = [] for s, e in segments: length = e - s if length <= max_len: capped.append((s, e)) else: num = math.ceil(length / max_len) for i in range(num): cs = s + i * max_len ce = min(s + (i + 1) * max_len, e) if ce - cs > int(0.2 * sample_rate): # drop ultra-short tails capped.append((cs, ce)) return capped def process_file(in_path: Path, out_dir: Path, max_seconds: float) -> int: audio, sr = read_audio(in_path) if sr != 16000: # Expect normalized inputs; skip others return 0 segments = vad_segments(audio, sr, aggressiveness=2, frame_ms=30, min_speech_ms=250, max_silence_ms=300) segments = hard_cap_segments(segments, max_seconds, sr) stem = in_path.stem for idx, (s, e) in enumerate(segments): seg = audio[s:e] out_path = out_dir / f"{stem}_seg{idx:04d}.wav" write_audio(out_path, seg, sr) return len(segments) def main() -> None: parser = argparse.ArgumentParser(description="VAD-based segmentation to <= max seconds, expects 16kHz mono WAV inputs.") parser.add_argument("--indir", required=True, help="Input audio root") parser.add_argument("--outdir", required=True, help="Output audio root (can be same as input subfolders)") parser.add_argument("--max-seconds", type=float, default=30.0, help="Maximum segment length in seconds") args = parser.parse_args() inroot = Path(args.indir) outroot = Path(args.outdir) wavs = list(inroot.rglob("*.16k.wav")) if not wavs: # fallback: all wavs wavs = list(inroot.rglob("*.wav")) total_segments = 0 for f in tqdm(wavs, desc="Segmenting"): out_dir = f.parent total_segments += process_file(f, out_dir, args.max_seconds) print(f"Total segments written: {total_segments}") if __name__ == "__main__": main()