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refactor code using black formatter and add dockerhub push cmd, and update base image to version containing cudnn8 #13
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Original file line number | Diff line number | Diff line change |
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@@ -13,18 +13,15 @@ VoiceStreamAI is a Python 3 -based server and JavaScript client solution that en | |
- Customizable audio chunk processing strategies. | ||
- Support for multilingual transcription. | ||
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## Demo Video | ||
## Demo | ||
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[View Demo Video](https://raw.githubusercontent.com/TyreseDev/VoiceStreamAI/main/img/voicestreamai_test.mp4) | ||
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https://github.com/alesaccoia/VoiceStreamAI/assets/1385023/9b5f2602-fe0b-4c9d-af9e-4662e42e23df | ||
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## Demo Client | ||
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![Client Demo](/img/client.png "Client Demo") | ||
![Demo Image](https://raw.githubusercontent.com/TyreseDev/VoiceStreamAI/main/img/client.png) | ||
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## Running with Docker | ||
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This will not guide you in detail on how to use CUDA in docker, see for example [here](https://medium.com/@kevinsjy997/configure-docker-to-use-local-gpu-for-training-ml-models-70980168ec9b). | ||
This will not guide you in detail on how to use CUDA in docker, see for example [here](https://medium.com/@kevinsjy997/configure-docker-to-use-local-gpu-for-training-ml-models-70980168ec9b). | ||
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Still, these are the commands for Linux: | ||
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@@ -52,13 +49,13 @@ After getting your VAD token (see next sections) run: | |
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sudo docker volume create huggingface_models | ||
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sudo docker run --gpus all -p 8765:8765 -v huggingface_models:/root/.cache/huggingface -e PYANNOTE_AUTH_TOKEN='VAD_TOKEN_HERE' voicestreamai | ||
sudo docker run --gpus all -p 80:80 -v huggingface_models:/root/.cache/huggingface -e PYANNOTE_AUTH_TOKEN='VAD_TOKEN_HERE' voicestreamai | ||
``` | ||
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The "volume" stuff will allow you not to re-download the huggingface models each time you re-run the container. If you don't need this, just use: | ||
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```bash | ||
sudo docker run --gpus all -p 8765:8765 -e PYANNOTE_AUTH_TOKEN='VAD_TOKEN_HERE' voicestreamai | ||
sudo docker run --gpus all -p 80:80 -e PYANNOTE_AUTH_TOKEN='VAD_TOKEN_HERE' voicestreamai | ||
``` | ||
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## Normal, Manual Installation | ||
|
@@ -92,7 +89,7 @@ The VoiceStreamAI server can be customized through command line arguments, allow | |
- `--asr-type`: Specifies the type of Automatic Speech Recognition (ASR) pipeline to use (default: `faster_whisper`). | ||
- `--asr-args`: A JSON string containing additional arguments for the ASR pipeline (one can for example change `model_name` for whisper) | ||
- `--host`: Sets the host address for the WebSocket server (default: `127.0.0.1`). | ||
- `--port`: Sets the port on which the server listens (default: `8765`). | ||
- `--port`: Sets the port on which the server listens (default: `80`). | ||
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For running the server with the standard configuration: | ||
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@@ -103,7 +100,7 @@ For running the server with the standard configuration: | |
python3 -m src.main --vad-args '{"auth_token": "vad token here"}' | ||
``` | ||
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You can see all the command line options with the command: | ||
You can see all the command line options with the command: | ||
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```bash | ||
python3 -m src.main --help | ||
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@@ -112,13 +109,12 @@ python3 -m src.main --help | |
## Client Usage | ||
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1. Open the `client/VoiceStreamAI_Client.html` file in a web browser. | ||
2. Enter the WebSocket address (default is `ws://localhost:8765`). | ||
2. Enter the WebSocket address (default is `ws://localhost/ws`). | ||
3. Configure the audio chunk length and offset. See below. | ||
4. Select the language for transcription. | ||
5. Click 'Connect' to establish a WebSocket connection. | ||
6. Use 'Start Streaming' and 'Stop Streaming' to control audio capture. | ||
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## Technology Overview | ||
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- **Python Server**: Manages WebSocket connections, processes audio streams, and handles voice activity detection and transcription. | ||
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@@ -207,10 +203,8 @@ Please make sure that the end variables are in place for example for the VAD aut | |
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### Dependence on Audio Files | ||
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Currently, VoiceStreamAI processes audio by saving chunks to files and then running these files through the models. | ||
Currently, VoiceStreamAI processes audio by saving chunks to files and then running these files through the models. | ||
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## Contributors | ||
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- Alessandro Saccoia - [[email protected]](mailto:[email protected]) | ||
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This project is open for contributions. Feel free to fork the repository and submit pull requests. |
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The Dockerfile in the current repository will lead to the following issues:
The current commit can fix this issue.