makaveli10
148 lines
6.5 KiB
Markdown
148 lines
6.5 KiB
Markdown
# whisper-live
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A nearly-live implementation of OpenAI's Whisper.
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This project is a real-time transcription application that uses the OpenAI Whisper model to convert speech input into text output. It can be used to transcribe both live audio input from microphone and pre-recorded audio files.
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Unlike traditional speech recognition systems that rely on continuous audio streaming, we use [voice activity detection (VAD)](https://github.com/snakers4/silero-vad) to detect the presence of speech and only send the audio data to whisper when speech is detected. This helps to reduce the amount of data sent to the whisper model and improves the accuracy of the transcription output.
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## Installation
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- Install PyAudio and ffmpeg
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```bash
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bash scripts/setup.sh
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```
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- Install whisper-live from pip
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```bash
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pip install whisper-live
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```
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### Setting up NVIDIA/TensorRT-LLM for TensorRT backend
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- Please follow [TensorRT_whisper readme](https://github.com/collabora/WhisperLive/blob/main/TensorRT_whisper.md) for setup of [NVIDIA/TensorRT-LLM](https://github.com/NVIDIA/TensorRT-LLM) and for building Whisper-TensorRT engine.
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## Getting Started
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The server supports two backends `faster_whisper` and `tensorrt`. If running `tensorrt` backend follow [TensorRT_whisper readme](https://github.com/collabora/WhisperLive/blob/main/TensorRT_whisper.md)
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### Running the Server
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- [Faster Whisper](https://github.com/SYSTRAN/faster-whisper) backend
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```bash
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python3 run_server.py --port 9090 \
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--backend faster_whisper
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# running with custom model
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python3 run_server.py --port 9090 \
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--backend faster_whisper
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-fw "/path/to/custom/faster/whisper/model"
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```
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- TensorRT backend. Currently, we recommend to only use the docker setup for TensorRT. Follow [TensorRT_whisper readme](https://github.com/collabora/WhisperLive/blob/main/TensorRT_whisper.md) which works as expected. Make sure to build your TensorRT Engines before running the server with TensorRT backend.
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```bash
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# Run English only model
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python3 run_server.py -p 9090 \
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-b tensorrt \
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-trt /home/TensorRT-LLM/examples/whisper/whisper_small_en
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# Run Multilingual model
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python3 run_server.py -p 9090 \
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-b tensorrt \
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-trt /home/TensorRT-LLM/examples/whisper/whisper_small \
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-m
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```
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### Running the Client
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- To transcribe an audio file:
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```python
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from whisper_live.client import TranscriptionClient
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client = TranscriptionClient(
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"localhost",
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9090,
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lang="en",
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translate=False,
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model="small",
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use_vad=False,
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)
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client("tests/jfk.wav")
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```
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This command transcribes the specified audio file (audio.wav) using the Whisper model. It connects to the server running on localhost at port 9090. Using a multilingual model, language for the transcription will be automatically detected. You can also use the language option to specify the target language for the transcription, in this case, English ("en"). The translate option should be set to `True` if we want to translate from the source language to English and `False` if we want to transcribe in the source language.
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- To transcribe from microphone:
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```python
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from whisper_live.client import TranscriptionClient
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client = TranscriptionClient(
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"localhost",
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9090,
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lang="hi",
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translate=True,
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model="small",
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use_vad=True,
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)
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client()
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```
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This command captures audio from the microphone and sends it to the server for transcription. It uses the multilingual model with `hi` as the selected language. We use whisper `small` by default but can be changed to any other option based on the requirements and the hardware running the server. The server also has an option to use `VAD`(voice activity detection) which is set to True by default.
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- To transcribe from a HLS stream:
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```python
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from whisper_live.client import TranscriptionClient
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client = TranscriptionClient(host, port, lang="en", translate=False)
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client(hls_url="http://as-hls-ww-live.akamaized.net/pool_904/live/ww/bbc_1xtra/bbc_1xtra.isml/bbc_1xtra-audio%3d96000.norewind.m3u8")
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```
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This command streams audio into the server from a HLS stream. It uses the same options as the previous command, using the multilingual model and specifying the target language and task.
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## Transcribe audio from browser
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- Run the server with your desired backend as shown [here](https://github.com/collabora/WhisperLive?tab=readme-ov-file#running-the-server)
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### Chrome Extension
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- Refer to [Audio-Transcription-Chrome](https://github.com/collabora/whisper-live/tree/main/Audio-Transcription-Chrome#readme) to use Chrome extension.
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### Firefox Extension
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- Refer to [Audio-Transcription-Firefox](https://github.com/collabora/whisper-live/tree/main/Audio-Transcription-Firefox#readme) to use Mozilla Firefox extension.
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## Whisper Live Server in Docker
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- GPU
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- Faster-Whisper
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```bash
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docker build . -t whisper-live -f docker/Dockerfile.gpu
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docker run -it --gpus all -p 9090:9090 whisper-live:latest
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```
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- TensorRT. Follow [TensorRT_whisper readme](https://github.com/collabora/WhisperLive/blob/main/TensorRT_whisper.md) in order to setup docker and use TensorRT backend. We provide a pre-built docker image which has TensorRT-LLM built and ready to use.
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- CPU
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```bash
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docker build . -t whisper-live -f docker/Dockerfile.cpu
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docker run -it -p 9090:9090 whisper-live:latest
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```
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**Note**: By default we use "small" model size. To build docker image for a different model size, change the size in server.py and then build the docker image.
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## Future Work
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- [ ] Add translation to other languages on top of transcription.
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- [x] TensorRT backend for Whisper.
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## Contact
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We are available to help you with both Open Source and proprietary AI projects. You can reach us via the Collabora website or [vineet.suryan@collabora.com](mailto:vineet.suryan@collabora.com) and [marcus.edel@collabora.com](mailto:marcus.edel@collabora.com).
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## Citations
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```bibtex
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@article{Whisper
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title = {Robust Speech Recognition via Large-Scale Weak Supervision},
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url = {https://arxiv.org/abs/2212.04356},
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author = {Radford, Alec and Kim, Jong Wook and Xu, Tao and Brockman, Greg and McLeavey, Christine and Sutskever, Ilya},
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publisher = {arXiv},
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year = {2022},
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}
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```
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```bibtex
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@misc{Silero VAD,
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author = {Silero Team},
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title = {Silero VAD: pre-trained enterprise-grade Voice Activity Detector (VAD), Number Detector and Language Classifier},
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year = {2021},
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publisher = {GitHub},
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journal = {GitHub repository},
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howpublished = {\url{https://github.com/snakers4/silero-vad}},
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commit = {insert_some_commit_here},
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email = {hello@silero.ai}
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}
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