mirror of
https://github.com/HumanAIGC-Engineering/gradio-webrtc.git
synced 2026-02-04 17:39:23 +08:00
322 lines
12 KiB
Python
322 lines
12 KiB
Python
import os
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import gradio as gr
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_docs = {
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"WebRTC": {
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"description": "Stream audio/video with WebRTC",
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"members": {
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"__init__": {
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"rtc_configuration": {
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"type": "dict[str, Any] | None",
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"default": "None",
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"description": "The configration dictionary to pass to the RTCPeerConnection constructor. If None, the default configuration is used.",
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},
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"height": {
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"type": "int | str | None",
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"default": "None",
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"description": "The height of the component, specified in pixels if a number is passed, or in CSS units if a string is passed. This has no effect on the preprocessed video file, but will affect the displayed video.",
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},
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"width": {
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"type": "int | str | None",
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"default": "None",
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"description": "The width of the component, specified in pixels if a number is passed, or in CSS units if a string is passed. This has no effect on the preprocessed video file, but will affect the displayed video.",
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},
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"label": {
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"type": "str | None",
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"default": "None",
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"description": "the label for this component. Appears above the component and is also used as the header if there are a table of examples for this component. If None and used in a `gr.Interface`, the label will be the name of the parameter this component is assigned to.",
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},
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"show_label": {
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"type": "bool | None",
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"default": "None",
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"description": "if True, will display label.",
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},
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"container": {
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"type": "bool",
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"default": "True",
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"description": "if True, will place the component in a container - providing some extra padding around the border.",
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},
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"scale": {
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"type": "int | None",
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"default": "None",
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"description": "relative size compared to adjacent Components. For example if Components A and B are in a Row, and A has scale=2, and B has scale=1, A will be twice as wide as B. Should be an integer. scale applies in Rows, and to top-level Components in Blocks where fill_height=True.",
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},
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"min_width": {
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"type": "int",
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"default": "160",
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"description": "minimum pixel width, will wrap if not sufficient screen space to satisfy this value. If a certain scale value results in this Component being narrower than min_width, the min_width parameter will be respected first.",
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},
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"interactive": {
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"type": "bool | None",
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"default": "None",
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"description": "if True, will allow users to upload a video; if False, can only be used to display videos. If not provided, this is inferred based on whether the component is used as an input or output.",
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},
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"visible": {
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"type": "bool",
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"default": "True",
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"description": "if False, component will be hidden.",
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},
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"elem_id": {
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"type": "str | None",
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"default": "None",
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"description": "an optional string that is assigned as the id of this component in the HTML DOM. Can be used for targeting CSS styles.",
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},
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"elem_classes": {
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"type": "list[str] | str | None",
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"default": "None",
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"description": "an optional list of strings that are assigned as the classes of this component in the HTML DOM. Can be used for targeting CSS styles.",
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},
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"render": {
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"type": "bool",
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"default": "True",
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"description": "if False, component will not render be rendered in the Blocks context. Should be used if the intention is to assign event listeners now but render the component later.",
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},
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"key": {
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"type": "int | str | None",
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"default": "None",
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"description": "if assigned, will be used to assume identity across a re-render. Components that have the same key across a re-render will have their value preserved.",
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},
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"mirror_webcam": {
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"type": "bool",
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"default": "True",
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"description": "if True webcam will be mirrored. Default is True.",
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},
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},
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"events": {"tick": {"type": None, "default": None, "description": ""}},
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},
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"__meta__": {"additional_interfaces": {}, "user_fn_refs": {"WebRTC": []}},
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}
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}
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abs_path = os.path.join(os.path.dirname(__file__), "css.css")
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with gr.Blocks(
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css_paths=abs_path,
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theme=gr.themes.Default(
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font_mono=[
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gr.themes.GoogleFont("Inconsolata"),
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"monospace",
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],
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),
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) as demo:
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gr.Markdown(
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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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""",
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elem_classes=["md-custom"],
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header_links=True,
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)
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gr.Markdown(
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"""
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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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""",
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elem_classes=["md-custom"],
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header_links=True,
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)
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gr.Markdown(
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"""
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##
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""",
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elem_classes=["md-custom"],
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header_links=True,
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)
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gr.ParamViewer(value=_docs["WebRTC"]["members"]["__init__"], linkify=[])
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demo.load(
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None,
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js=r"""function() {
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const refs = {};
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const user_fn_refs = {
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WebRTC: [], };
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requestAnimationFrame(() => {
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Object.entries(user_fn_refs).forEach(([key, refs]) => {
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if (refs.length > 0) {
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const el = document.querySelector(`.${key}-user-fn`);
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if (!el) return;
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refs.forEach(ref => {
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el.innerHTML = el.innerHTML.replace(
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new RegExp("\\b"+ref+"\\b", "g"),
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`<a href="#h-${ref.toLowerCase()}">${ref}</a>`
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);
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})
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}
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})
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Object.entries(refs).forEach(([key, refs]) => {
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if (refs.length > 0) {
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const el = document.querySelector(`.${key}`);
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if (!el) return;
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refs.forEach(ref => {
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el.innerHTML = el.innerHTML.replace(
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new RegExp("\\b"+ref+"\\b", "g"),
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`<a href="#h-${ref.toLowerCase()}">${ref}</a>`
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);
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})
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}
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})
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})
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}
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""",
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)
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demo.launch()
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