This script modifies OpenAI's Whisper to produce more reliable timestamps.
jfk.mp4
- updated to use Whisper's more reliable word-level timestamps method.
- the more reliable word timestamps allows regrouping all words into segments with more natural boundaries.
- can now suppress silence with Silero VAD (requires PyTorch 1.2.0+)
- non-VAD silence suppression is also more robust
- see Quick 1.X → 2.X Guide
a.mp4
- more control over the timestamps than default Whisper
- supports direct preprocessing with Demucs to isolate voice
- support dynamic quantization to decrease memory usage for inference on CPU
- lower memory usage than default Whisper when transcribing very long input audio tracks
pip install -U stable-ts
To install the lastest commit:
pip install -U git+https://github.com/SinanAkkoyun/whisper-stable-ts-prob.git
Transcribe audio then save result as JSON file which contains the original inference results.
This allows results to be reprocessed different without having to redo inference.
Change audio.json
to audio.srt
to process it directly into SRT.
stable-ts audio.mp3 -o audio.json
Processing JSON file of the results into SRT.
stable-ts audio.json -o audio.srt
Transcribe multiple audio files then process the results directly into SRT files.
stable-ts audio1.mp3 audio2.mp3 audio3.mp3 -o audio1.srt audio2.srt audio3.srt
import stable_whisper
model = stable_whisper.load_model('base')
# modified model should run just like the regular model but accepts additional parameters
result = model.transcribe('audio.mp3')
# srt/vtt
result.to_srt_vtt('audio.srt')
# ass
result.to_ass('audio.ass')
# json
result.save_as_json('audio.json')
- for reliable segment timestamps, do not disable word timestamps with
word_timestamps=False
because word timestamps is also used to correct segment timestamps - use
demucs=True
andvad=True
for music - if audio is not transcribing properly compared to whisper, try
mel_first=True
at cost of more memory usuage for long audio tracks
results_to_sentence_srt(result, 'audio.srt')
→result.to_srt_vtt('audio.srt', word_level=False)
results_to_word_srt(result, 'audio.srt')
→result.to_srt_vtt('output.srt', segment_level=False)
results_to_sentence_word_ass(result, 'audio.srt')
→result.to_ass('output.ass')
- there's no need to stabilize segment after inference because they're already stabilized during inference
transcribe()
returns aWhisperResult
object which can be converted todict
with.to_dict()
. e.gresult.to_dict()
Stable-ts has a preset for regrouping words into different into segments with more natural boundaries.
This preset is enabled by regroup=True
. But there are other built-in regrouping methods that allow you to customize the regrouping logic.
This preset is just a predefined a combination of those methods.
xata.mp4
result0 = model.transcribe('audio.mp3', regroup=True) # regroup is True by default
# regroup=True is same as below
result1 = model.transcribe('audio.mp3', regroup=False)
(
result1
.split_by_punctuation([('.', ' '), '。', '?', '?', ',', ','])
.split_by_gap(.5)
.merge_by_gap(.15, max_words=3)
.split_by_punctuation([('.', ' '), '。', '?', '?'])
)
# result0 == result1
- Requirement: Pillow or opencv-python
import stable_whisper
# regions on the waveform colored red is where it will be likely be suppressed and marked to as silent
# [q_levels=20] and [k_size=5] are defaults for non-VAD.
stable_whisper.visualize_suppression('audio.mp3', 'image.png', q_levels=20, k_size = 5)
# [vad_threshold=0.35] is defaults for VAD.
stable_whisper.visualize_suppression('audio.mp3', 'image.png', vad=True, vad_threshold=0.35)
import stable_whisper
stable_whisper.encode_video_comparison(
'audio.mp3',
['audio_sub1.srt', 'audio_sub2.srt'],
output_videopath='audio.mp4',
labels=['Example 1', 'Example 2']
)
This project is licensed under the MIT License - see the LICENSE file for details
Includes slight modification of the original work: Whisper