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README.md
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---
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sdk: gradio
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sdk_version: 5.1.0
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app_file: app.py
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pinned: false
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license: mit
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short_description: Stream Audio/Video in real time with WebRTC
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---
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tags: [gradio-custom-component, Video, Audio, streaming, webrtc, realtime]
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title: gradio_webrtc
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short_description: Stream audio/video in realtime with webrtc
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colorFrom: blue
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colorTo: yellow
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sdk: gradio
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pinned: false
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---
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<h1 style='text-align: center; margin-bottom: 1rem'> Gradio WebRTC ⚡️ </h1>
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<div style="display: flex; flex-direction: row; justify-content: center">
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<img style="display: block; padding-right: 5px; height: 20px;" alt="Static Badge" src="https://img.shields.io/badge/version%20-%200.0.5%20-%20orange">
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<a href="https://github.com/freddyaboulton/gradio-webrtc" target="_blank"><img alt="Static Badge" src="https://img.shields.io/badge/github-white?logo=github&logoColor=black"></a>
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</div>
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<h3 style='text-align: center'>
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Stream video and audio in real time with Gradio using WebRTC.
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</h3>
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## Installation
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```bash
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pip install gradio_webrtc
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```
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## Examples:
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1. [Object Detection from Webcam with YOLOv10](https://huggingface.co/spaces/freddyaboulton/webrtc-yolov10n) 📷
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2. [Streaming Object Detection from Video with RT-DETR](https://huggingface.co/spaces/freddyaboulton/rt-detr-object-detection-webrtc) 🎥
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3. [Text-to-Speech](https://huggingface.co/spaces/freddyaboulton/parler-tts-streaming-webrtc) 🗣️
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## Usage
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The WebRTC component supports the following three use cases:
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1. Streaming video from the user webcam to the server and back
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2. Streaming Video from the server to the client
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3. Streaming Audio from the server to the client
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Streaming Audio from client to the server and back (conversational AI) is not supported yet.
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## Streaming Video from the User Webcam to the Server and Back
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```python
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import gradio as gr
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from gradio_webrtc import WebRTC
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def detection(image, conf_threshold=0.3):
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... your detection code here ...
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with gr.Blocks() as demo:
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image = WebRTC(label="Stream", mode="send-receive", modality="video")
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conf_threshold = gr.Slider(
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label="Confidence Threshold",
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minimum=0.0,
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maximum=1.0,
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step=0.05,
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value=0.30,
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)
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image.stream(
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fn=detection,
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inputs=[image, conf_threshold],
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outputs=[image], time_limit=10
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)
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if __name__ == "__main__":
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demo.launch()
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```
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* Set the `mode` parameter to `send-receive` and `modality` to "video".
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* The `stream` event's `fn` parameter is a function that receives the next frame from the webcam
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as a **numpy array** and returns the processed frame also as a **numpy array**.
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* Numpy arrays are in (height, width, 3) format where the color channels are in RGB format.
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* The `inputs` parameter should be a list where the first element is the WebRTC component. The only output allowed is the WebRTC component.
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* The `time_limit` parameter is the maximum time in seconds the video stream will run. If the time limit is reached, the video stream will stop.
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## Streaming Video from the User Webcam to the Server and Back
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```python
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import gradio as gr
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from gradio_webrtc import WebRTC
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import cv2
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def generation():
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url = "https://download.tsi.telecom-paristech.fr/gpac/dataset/dash/uhd/mux_sources/hevcds_720p30_2M.mp4"
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cap = cv2.VideoCapture(url)
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iterating = True
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while iterating:
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iterating, frame = cap.read()
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yield frame
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with gr.Blocks() as demo:
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output_video = WebRTC(label="Video Stream", mode="receive", modality="video")
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button = gr.Button("Start", variant="primary")
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output_video.stream(
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fn=generation, inputs=None, outputs=[output_video],
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trigger=button.click
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)
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if __name__ == "__main__":
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demo.launch()
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```
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* Set the "mode" parameter to "receive" and "modality" to "video".
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* The `stream` event's `fn` parameter is a generator function that yields the next frame from the video as a **numpy array**.
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* The only output allowed is the WebRTC component.
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* The `trigger` parameter the gradio event that will trigger the webrtc connection. In this case, the button click event.
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## Streaming Audio from the Server to the Client
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```python
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import gradio as gr
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from pydub import AudioSegment
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def generation(num_steps):
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for _ in range(num_steps):
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segment = AudioSegment.from_file("/Users/freddy/sources/gradio/demo/audio_debugger/cantina.wav")
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yield (segment.frame_rate, np.array(segment.get_array_of_samples()).reshape(1, -1))
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with gr.Blocks() as demo:
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audio = WebRTC(label="Stream", mode="receive", modality="audio")
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num_steps = gr.Slider(
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label="Number of Steps",
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minimum=1,
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maximum=10,
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step=1,
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value=5,
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)
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button = gr.Button("Generate")
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audio.stream(
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fn=generation, inputs=[num_steps], outputs=[audio],
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trigger=button.click
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)
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```
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* Set the "mode" parameter to "receive" and "modality" to "audio".
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* The `stream` event's `fn` parameter is a generator function that yields the next audio segment as a tuple of (frame_rate, audio_samples).
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* The numpy array should be of shape (1, num_samples).
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* The `outputs` parameter should be a list with the WebRTC component as the only element.
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## Deployment
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When deploying in a cloud environment (like Hugging Face Spaces, EC2, etc), you need to set up a TURN server to relay the WebRTC traffic.
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The easiest way to do this is to use a service like Twilio.
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```python
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from twilio.rest import Client
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import os
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account_sid = os.environ.get("TWILIO_ACCOUNT_SID")
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auth_token = os.environ.get("TWILIO_AUTH_TOKEN")
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client = Client(account_sid, auth_token)
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token = client.tokens.create()
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rtc_configuration = {
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"iceServers": token.ice_servers,
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"iceTransportPolicy": "relay",
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}
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with gr.Blocks() as demo:
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...
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rtc = WebRTC(rtc_configuration=rtc_configuration, ...)
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...
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```
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