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video-vec2wav2-tokenizer-3
Version 3 - continuation shard of the video-to-AI-dataset tokenizer project.
Version 2 - continuation shard of the video-to-AI-dataset tokenizer project.
Production-ready pipeline (Python package video_vec2wav2_tokenizer, CLI command
video2dataset) that turns a folder of videos into clean AI training datasets
for speech recognition (ASR) and text-to-speech (TTS).
videos ──► audio (16 kHz mono PCM) ──► whisper transcript ──► clips ──► metadata.csv / dataset.jsonl / tts_metadata.csv / report.json
- Video processing — recursive scan of
mp4 / mkv / avi / mov / webm, FFmpeg audio extraction to mono · 16 kHz · 16-bit PCM WAV. - Speech recognition — faster-whisper, CPU & CUDA, automatic language detection, word-level timestamps.
- Segmentation — cut audio by transcript timestamps into
dataset/audio/000001.wav …. - Dataset generation —
metadata.csv,dataset.jsonl,tts_metadata.csv. - Feature extraction (optional) — streaming
features/train.bin+train.datwith float32 samples, mel spectrograms, duration and sample rate. - Statistics —
report.jsonwith totals, durations and language distribution. - Training —
train_wav2vec2.py: HuggingFace Wav2Vec2 CTC with resume, multi-GPU, mixed precision and checkpointing. - Performance — multiprocessing, batch processing, tqdm progress bars and memory-efficient streaming that scales past 1 TB of source media.
Installation
Requires Python 3.11+ and the FFmpeg binary on your PATH.
# 1. FFmpeg (one of):
# macOS: brew install ffmpeg
# Ubuntu: sudo apt install ffmpeg
# Windows: winget install Gyan.FFmpeg (or choco install ffmpeg)
# 2. The package
python -m venv .venv && source .venv/bin/activate # Windows: .venv\Scripts\activate
pip install -r requirements.txt
pip install -e . # exposes the `video2dataset` command
# Optional: training extras (torch + transformers)
pip install -e ".[train]"
Verify:
video2dataset --version
ffmpeg -version
Quick start
# Drop videos into input/videos/ (any nesting), then run the full pipeline:
video2dataset process ./input/videos --extract-features
Outputs land under dataset/:
dataset/
├── audio/
│ ├── 000001.wav
│ ├── 000002.wav
│ └── ...
├── metadata.csv # 000001.wav|Hello world
├── dataset.jsonl # {"audio":"audio/000001.wav","text":"Hello world","duration":2.5}
├── tts_metadata.csv # 000001.wav|speaker_001|Hello world
└── report.json
features/
├── train.bin
└── train.dat
CLI
video2dataset process ./videos # video -> full dataset
video2dataset transcribe ./audio # wav -> transcripts/<name>.jsonl
video2dataset segment ./audio # wav + sidecar transcript -> clips
video2dataset features ./dataset # clips -> features/train.{bin,dat}
video2dataset train ./dataset # fine-tune Wav2Vec2 CTC
Global flags (override config.yaml): --config, --output-dir, --device {auto,cpu,cuda}, --whisper-model, --language, -v/--verbose.
# Examples
video2dataset process ./input/videos --whisper-model large-v3 --device cuda
video2dataset process ./input/videos --language en --output-dir build/en
video2dataset transcribe ./dataset/_work --device cpu
video2dataset features ./dataset
video2dataset train ./dataset --epochs 30 --batch-size 8 --resume
Multi-GPU training
torchrun --nproc_per_node=4 -m video_vec2wav2_tokenizer.training.train_wav2vec2 \
--dataset-dir dataset --output-dir dataset/checkpoints
Configuration (config.yaml)
model_name: facebook/wav2vec2-base-960h # training model
whisper_model: large-v3 # ASR model
language: null # null = auto-detect
device: auto # auto | cpu | cuda
sample_rate: 16000
channels: 1
min_segment_length: 1.0
max_segment_length: 20.0
default_speaker: speaker_001
speaker_mode: fixed # fixed | per_video | per_file
output_dir: dataset
features_dir: features
num_workers: 4
batch_size: 16
speaker_mode controls TTS speaker identification: a single fixed speaker, one
per source video, or one per clip.
Architecture
video-vec2wav2-tokenizer-3/ # repo root
├── config.yaml
├── requirements.txt
├── pyproject.toml
├── README.md
├── input/videos/ # drop source videos here
└── video_vec2wav2_tokenizer/ # importable package
├── cli/ # argparse command surface
├── audio/ # FFmpeg extraction + WAV loading/resampling
├── transcription/ # faster-whisper wrapper (lazy import)
├── segmentation/ # timestamp normalisation + clip cutting
├── features/ # streaming bin/dat feature store + mel spectrograms
├── training/ # Wav2Vec2 CTC trainer
├── utils/ # config, logging, data models, IO
├── configs/ # packaged default.yaml
├── tests/ # pytest suite (no heavy deps required)
├── pipeline.py # end-to-end orchestration
├── dataset_builder.py / statistics.py
└── main.py # `python main.py` entry point
Design notes:
- Lazy heavy imports.
faster-whisper,torchandtransformersare imported only when their command runs, so the core library and tests stay light and fast. - Streaming everywhere. Transcription consumes whisper's generator; clips and manifest rows are produced per source video; the feature store appends one clip at a time. Memory use is independent of corpus size.
- Graceful degradation. WAV IO and mel spectrograms fall back to stdlib +
NumPy when
soundfile/librosaaren't installed. - Fault tolerant. A failing video is logged and skipped; the run continues.
Output formats
| File | Format |
|---|---|
metadata.csv |
`000001.wav |
dataset.jsonl |
{"audio":"audio/000001.wav","text":"Hello world","duration":2.5} |
tts_metadata.csv |
`000001.wav |
report.json |
totals, total/average duration, language distribution |
features/train.bin |
flat little-endian float32: audio samples + mel per clip |
features/train.dat |
JSONL index (offsets, shapes, duration, sample rate) |
Testing
pip install -e ".[dev]"
pytest # runs video_vec2wav2_tokenizer/tests
pytest --cov=video_vec2wav2_tokenizer # with coverage
The default test suite needs only numpy, pyyaml and pytest — no FFmpeg,
whisper or torch required.
License
MIT.
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