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WhisperLive/README.md
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# WhisperLive
<h2 align="center">
<a href="https://www.youtube.com/watch?v=0PHWCApIcCI"><img
src="https://img.youtube.com/vi/0PHWCApIcCI/0.jpg" style="background-color:rgba(0,0,0,0);" height=300 alt="WhisperLive"></a>
<br><br>A nearly-live implementation of OpenAI's Whisper.
<br><br>
</h2>
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.
## Installation
- Install PyAudio and ffmpeg
```bash
bash scripts/setup.sh
```
- Install whisper-live from pip
```bash
pip install whisper-live
```
### Setting up NVIDIA/TensorRT-LLM for TensorRT backend
- 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.
## Getting Started
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)
### Running the Server
- [Faster Whisper](https://github.com/SYSTRAN/faster-whisper) backend
```bash
python3 run_server.py --port 9090 \
--backend faster_whisper
# running with custom model
python3 run_server.py --port 9090 \
--backend faster_whisper
-fw "/path/to/custom/faster/whisper/model"
```
- 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.
```bash
# Run English only model
python3 run_server.py -p 9090 \
-b tensorrt \
-trt /home/TensorRT-LLM/examples/whisper/whisper_small_en
# Run Multilingual model
python3 run_server.py -p 9090 \
-b tensorrt \
-trt /home/TensorRT-LLM/examples/whisper/whisper_small \
-m
```
#### Controlling OpenMP Threads
To control the number of threads used by OpenMP, you can set the `OMP_NUM_THREADS` environment variable. This is useful for managing CPU resources and ensuring consistent performance. If not specified, `OMP_NUM_THREADS` is set to `1` by default. You can change this by using the `--omp_num_threads` argument:
```bash
python3 run_server.py --port 9090 \
--backend faster_whisper \
--omp_num_threads 4
```
### Running the Client
- Initializing the client with below parameters:
- `lang`: Language of the input audio, applicable only if using a multilingual model.
- `translate`: If set to `True` then translate from any language to `en`.
- `model`: Whisper model size.
- `use_vad`: Whether to use `Voice Activity Detecion` on the server.
- `save_output_recording`: Set to True to save the microphone input as a `.wav` file during live transcription. This option is helpful for recording sessions for later playback or analysis. Defaults to `False`.
- `output_recording_filename`: Specifies the `.wav` file path where the microphone input will be saved if `save_output_recording` is set to `True`.
```python
from whisper_live.client import TranscriptionClient
client = TranscriptionClient(
"localhost",
9090,
lang="en",
translate=False,
model="small",
use_vad=False,
save_output_recording=True, # Only used for microphone input, False by Default
output_recording_filename="./output_recording.wav" # Only used for microphone input
)
```
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.
- Trancribe an audio file:
```python
client("tests/jfk.wav")
```
- To transcribe from microphone:
```python
client()
```
- TO transcribe from a RTSP stream:
```python
client(rtsp_url="rtsp://admin:admin@192.168.0.1/rtsp")
```
- To transcribe from a HLS stream:
```python
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")
```
## Browser Extensions
- Run the server with your desired backend as shown [here](https://github.com/collabora/WhisperLive?tab=readme-ov-file#running-the-server).
- Transcribe audio directly from your browser using our Chrome or Firefox extensions. Refer to [Audio-Transcription-Chrome](https://github.com/collabora/whisper-live/tree/main/Audio-Transcription-Chrome#readme) and [Audio-Transcription-Firefox](https://github.com/collabora/whisper-live/tree/main/Audio-Transcription-Firefox#readme) for setup instructions.
## Whisper Live Server in Docker
- GPU
- Faster-Whisper
```bash
docker run -it --gpus all -p 9090:9090 ghcr.io/collabora/whisperlive-gpu:latest
```
- 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.
- CPU
```bash
docker run -it -p 9090:9090 ghcr.io/collabora/whisperlive-cpu:latest
```
**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.
## Future Work
- [ ] Add translation to other languages on top of transcription.
- [x] TensorRT backend for Whisper.
## Contact
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).
## Citations
```bibtex
@article{Whisper
title = {Robust Speech Recognition via Large-Scale Weak Supervision},
url = {https://arxiv.org/abs/2212.04356},
author = {Radford, Alec and Kim, Jong Wook and Xu, Tao and Brockman, Greg and McLeavey, Christine and Sutskever, Ilya},
publisher = {arXiv},
year = {2022},
}
```
```bibtex
@misc{Silero VAD,
author = {Silero Team},
title = {Silero VAD: pre-trained enterprise-grade Voice Activity Detector (VAD), Number Detector and Language Classifier},
year = {2021},
publisher = {GitHub},
journal = {GitHub repository},
howpublished = {\url{https://github.com/snakers4/silero-vad}},
email = {hello@silero.ai}
}