add back word .srt, update readme

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Max Bain
2022-12-19 19:12:50 +00:00
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<h1 align="center">WhisperX</h1>
<h6 align="center">Made by Max Bain • :globe_with_meridians: <a href="https://www.maxbain.com/">https://www.maxbain.com/</a></h6>
<p align="center">
<a href="https://github.com/m-bain/whisperX/stargazers">
<img src="https://img.shields.io/github/stars/m-bain/whisperX.svg?colorA=orange&colorB=orange&logo=github"
alt="GitHub stars">
</a>
<a href="https://github.com/hanxiao/bert-as-service/issues">
<img src="https://img.shields.io/github/issues/hanxiao/bert-as-service.svg"
alt="GitHub issues">
</a>
<a href="https://github.com/m-bain/whisperX/blob/master/LICENSE">
<img src="https://img.shields.io/github/license/m-bain/whisperX.svg"
alt="GitHub license">
</a>
<a href="https://twitter.com/intent/tweet?text=&url=https%3A%2F%2Fgithub.com%2Fm-bain%2FwhisperX">
<img src="https://img.shields.io/twitter/url/https/github.com/m-bain/whisperX.svg?style=social" alt="Twitter">
</a>
</p>
<h6 align="center">Made by Max Bain • :globe_with_meridians: <a href="https://www.maxbain.com">https://www.maxbain.com</a></h6>
<p align="left">Whisper-Based Automatic Speech Recognition (ASR) with improved timestamp accuracy using forced alignment.
@ -28,22 +47,22 @@ Install this package using
You may also need to install ffmpeg, rust etc. Follow openAI instructions here https://github.com/openai/whisper#setup.
<h2 align="left">Examples💬</h2>
### English
Run whisper on example segment (using default params)
`whisperx examples/sample01.wav --model medium.en --output examples/whisperx --align_model WAV2VEC2_ASR_LARGE_LV60K_960H --align_extend 2`
`whisperx examples/sample01.wav --model medium.en --output examples/whisperx --align_model WAV2VEC2_ASR_BASE_960H --align_extend 2`
If low gpu memory is required, use a smaller align model e.g. `WAV2VEC2_ASR_BASE_LV60K_960H`
For increased timestamp accuracy, at the cost of higher gpu mem, use a bigger alignment model e.g.
Using normal whisper out of the box, many transcriptions are out of sync:
`WAV2VEC2_ASR_LARGE_LV60K_960H` or `HUBERT_ASR_XLARGE`
https://user-images.githubusercontent.com/36994049/207743923-b4f0d537-29ae-4be2-b404-bb941db73652.mov
Now, using *WhisperX* with forced alignment to wav2vec2.0:
Result using *WhisperX* with forced alignment to wav2vec2.0 large:
https://user-images.githubusercontent.com/36994049/208253969-7e35fe2a-7541-434a-ae91-8e919540555d.mp4
Compare this to original whisper out the box, where many transcriptions are out of sync:
https://user-images.githubusercontent.com/36994049/207743923-b4f0d537-29ae-4be2-b404-bb941db73652.mov
## Other Languages
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<h2 align="left">Limitations ⚠️</h2>
- Not thoroughly tested, especially for non-english, results may vary -- please post issue to let me know its results on your data
- Not thoroughly tested, especially for non-english, results may vary -- please post issue to let me know the results on your data
- Whisper normalises spoken numbers e.g. "fifty seven" to arabic numerals "57". Need to perform this normalization after alignment, so the phonemes can be aligned. Currently just ignores numbers.
- Assumes the initial whisper timestamps are accurate to some degree (within margin of 2 seconds, adjust if needed -- bigger margins more prone to alignment errors)
- Hacked this up quite quickly, there might be some errors, please raise an issue if you encounter any.
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[ ] Automatic align model selection based on language detection
[ ] Reduce GPU (clear cache etc.)
[ ] Option to minimise gpu load (chunk wav2vec)
[ ] Incorporating word-level speaker diarization
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<h2 align="left">Contact</h2>
Contact maxbain[at]robots[dot]ox[dot]ac[dot]uk if using this for commerical purposes.
Contact maxbain[at]robots[dot]ox[dot]ac[dot]uk if using this commerically.
<h2 align="left">Acknowledgements 🙏</h2>