mirror of
https://github.com/HumanAIGC-Engineering/gradio-webrtc.git
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853d6a06b5
* Add code * add code * add code * Rename messages * rename * add code * Add demo * docs + demos + bug fixes * add code * styles * user guide * Styles * Add code * misc docs updates * print nit * whisper + pr * url for images * whsiper update * Fix bugs * remove demo files * version number * Fix pypi readme * Fix * demos * Add llama code editor * Update llama code editor and object detection cookbook * Add more cookbook demos * add code * Fix links for PR deploys * add code * Fix the install * add tts * TTS docs * Typo * Pending bubbles for reply on pause * Stream redesign (#63) * better error handling * Websocket error handling * add code --------- Co-authored-by: Freddy Boulton <freddyboulton@hf-freddy.local> * remove docs from dist * Some docs typos * more typos * upload changes + docs * docs * better phone * update docs * add code * Make demos better * fix docs + websocket start_up * remove mention of FastAPI app * fastphone tweaks * add code * ReplyOnStopWord fixes * Fix cookbook * Fix pypi readme * add code * bump versions * sambanova cookbook * Fix tags * Llm voice chat * kyutai tag * Add error message to all index.html * STT module uses Moonshine * Not required from typing extensions * fix llm voice chat * Add vpn warning * demo fixes * demos * Add more ui args and gemini audio-video * update cookbook * version 9 --------- Co-authored-by: Freddy Boulton <freddyboulton@hf-freddy.local>
70 lines
2.1 KiB
Markdown
70 lines
2.1 KiB
Markdown
## Demo does not work when deploying to the cloud
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Make sure you are using a TURN server. See [deployment](../deployment).
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## Recorded input audio sounds muffled during output audio playback
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By default, the microphone is [configured](https://github.com/freddyaboulton/gradio-webrtc/blob/903f1f70bd586f638ad3b5a3940c7a8ec70ad1f5/backend/gradio_webrtc/webrtc.py#L575) to do echo cancellation.
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This is what's causing the recorded audio to sound muffled when the streamed audio starts playing.
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You can disable this via the `track_constraints` (see [Advanced Configuration](../advanced-configuration)) with the following code:
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```python
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stream = Stream(
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track_constraints={
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"echoCancellation": False,
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"noiseSuppression": {"exact": True},
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"autoGainControl": {"exact": True},
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"sampleRate": {"ideal": 24000},
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"sampleSize": {"ideal": 16},
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"channelCount": {"exact": 1},
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},
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rtc_configuration=None,
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mode="send-receive",
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modality="audio",
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)
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```
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## How to raise errors in the UI
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You can raise `WebRTCError` in order for an error message to show up in the user's screen. This is similar to how `gr.Error` works.
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!!! warning
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The `WebRTCError` class is only supported in the `WebRTC` component.
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Here is a simple example:
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```python
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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(
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"/Users/freddy/sources/gradio/demo/audio_debugger/cantina.wav"
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)
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yield (
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segment.frame_rate,
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np.array(segment.get_array_of_samples()).reshape(1, -1),
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)
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time.sleep(3.5)
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raise WebRTCError("This is a test error")
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with gr.Blocks() as demo:
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audio = WebRTC(
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label="Stream",
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mode="receive",
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modality="audio",
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)
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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], trigger=button.click
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)
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demo.launch()
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``` |