A streamlined Speech-to-Text pipeline for Whisper using CTranslate2
Project description
WhisperS2T-Reborn ⚡
An Optimized Speech-to-Text Pipeline for the Whisper Model Using CTranslate2
WhisperS2T-Reborn is a modernized fork of WhisperS2T, an optimized lightning-fast Speech-to-Text (ASR) pipeline. It is tailored for the Whisper model using the CTranslate2 backend to provide faster transcription. It includes several heuristics to enhance transcription accuracy.
Whisper is a general-purpose speech recognition model developed by OpenAI. It is trained on a large dataset of diverse audio and is also a multitasking model that can perform multilingual speech recognition, speech translation, and language identification.
Installation
You must install FFMPEG on your system first.
Install or update to the latest released version of WhisperS2T-Reborn:
pip install -U whisper-s2t-reborn
Quick Start
Transcribe a single file
import whisper_s2t
model = whisper_s2t.load_model(model_identifier="large-v3")
files = ['audio1.wav']
lang_codes = ['en']
tasks = ['transcribe']
initial_prompts = [None]
out = model.transcribe_with_vad(files,
lang_codes=lang_codes,
tasks=tasks,
initial_prompts=initial_prompts,
batch_size=32)
print(out[0][0]) # Print first utterance for first file
"""
[Console Output]
{'text': "Let's bring in Phil Mackie who is there at the palace...",
'avg_logprob': -0.25426941679184695,
'no_speech_prob': 8.147954940795898e-05,
'start_time': 0.0,
'end_time': 24.8}
"""
Batch across multiple files
Passing multiple files allows segments from different files to be batched together, making better use of the GPU:
import whisper_s2t
model = whisper_s2t.load_model(model_identifier="large-v3")
files = ['audio1.wav', 'audio2.wav', 'audio3.wav']
lang_codes = ['en', 'en', 'en']
tasks = ['transcribe', 'transcribe', 'transcribe']
initial_prompts = [None, None, None]
out = model.transcribe_with_vad(files,
lang_codes=lang_codes,
tasks=tasks,
initial_prompts=initial_prompts,
batch_size=32)
# out[0] = results for audio1.wav, out[1] = results for audio2.wav, etc.
for file_idx, transcript in enumerate(out):
print(f"File {files[file_idx]}: {len(transcript)} segments")
Word-level alignment
To enable word-level timestamps, load the model with:
model = whisper_s2t.load_model("large-v3", asr_options={'word_timestamps': True})
Acknowledgements
- Original WhisperS2T: Thanks to shashig for the original WhisperS2T project that this fork is based on.
- OpenAI Whisper Team: Thanks to the OpenAI Whisper Team for open-sourcing the Whisper model.
- CTranslate2 Team: Thanks to the CTranslate2 Team for providing a faster inference engine for Transformers architecture.
- NVIDIA NeMo Team: Thanks to the NVIDIA NeMo Team for their contribution of the open-source VAD model used in this pipeline.
License
This project is licensed under MIT License - see the LICENSE file for details.
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