whitphx / streamlit-webrtc-example Goto Github PK

Real time video and audio processing examples with Streamlit and streamlit-webrtc

Home Page: https://webrtc.streamlit.app/

License: MIT License

Python 99.86% Makefile 0.14%

streamlit

streamlit-webrtc-example's Introduction

title	emoji	colorFrom	colorTo	sdk	sdk_version	app_file	pinned	license
Streamlit Webrtc Example	🌍	purple	yellow	streamlit	1.26.0	app.py	false	mit

streamlit-webrtc-example

Hosted on Streamlit Cloud: https://webrtc.streamlit.app/

Deployment notes

Streamlit Cloud automatically triggers the deployment on its CI/CD.

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streamlit-webrtc-example's Issues

Error running app

Error running app in https://webrtc.streamlit.app/

The https://webrtc.streamlit.app/ is not working

The rest of the demo links do not work either.

Error Message:

Oh no.
Error running app. If this keeps happening, please [contact support].

Over resource limits on Streamlit Cloud

unable to view app.py in repo wherever linked

Error running webrtc app

When trying to run the webrtc streamlit demo app, it does not spin up.

https://webrtc.streamlit.app/

How can i change the video stream resolution?

In all examples, it seems the MediaRecorder set the camera to 640x480.

following this thread it seems some walkaround exists but referring to MediaPlater and not MediaRecorder.

aiortc/aiortc#183

How to listen on_audio_ended event

Hi,
Thanks for this wonderful toolkit,I tried to listen the on_audio_ended event, I can not find your example, but my listener is not working,
so could you help me and add it to your examples?

Bug

Hi @whitphx ,The streamlet link that you have provide in about section is not working it is showing error please solve it

Camera is not opening when clicking Start button

Hello,
I was just testing the suggested demo app after the new version release and encountered the following behavior:
First, on the start button it starts loading...

Then, after a while (20sec approx.) it changes to the following screen...

After that the camera does not open and returns to the initial state.

swscaler warning

I am trying to grab a stream from IP camera through RTSP.

for every frame I get [swscaler @ 0x7b4d844f10] deprecated pixel format used, make sure you did set range correctly printed in the console

FFMPEG Details:

ffmpeg version 3.4.11-0ubuntu0.1 Copyright (c) 2000-2022 the FFmpeg developers
built with gcc 7 (Ubuntu/Linaro 7.5.0-3ubuntu1~18.04)
configuration: --prefix=/usr --extra-version=0ubuntu0.1 --toolchain=hardened --libdir=/usr/lib/aarch64-linux-gnu --incdir=/usr/include/aarch64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
libavutil      55. 78.100 / 55. 78.100
libavcodec     57.107.100 / 57.107.100
libavformat    57. 83.100 / 57. 83.100
libavdevice    57. 10.100 / 57. 10.100
libavfilter     6.107.100 /  6.107.100
libavresample   3.  7.  0 /  3.  7.  0
libswscale      4.  8.100 /  4.  8.100
libswresample   2.  9.100 /  2.  9.100
libpostproc    54.  7.100 / 54.  7.100

Other Details

python 3.9.14
streamlit 1.11.0
streamlit_webrtc version: 0.43.4

Examples do not work

Good day, could you please check if everything is fine with the examples page, nothing seem to work right now. Tried on different browsers and devices.

How do i export the recorded audio into a separate file ?

`
complete_audio_chunk = pydub.AudioSegment.empty()
while True:
if webrtc_ctx.audio_receiver:
sound_chunk = pydub.AudioSegment.empty()
try:
audio_frames = webrtc_ctx.audio_receiver.get_frames(timeout=1)
except queue.Empty:
time.sleep(0.1)
status_indicator.write("No frame arrived.")
continue

        status_indicator.write("Running. Say something!")
        for idx, audio_frame in enumerate(audio_frames):
            sound = pydub.AudioSegment(
                data=audio_frame.to_ndarray().tobytes(),
                sample_width=audio_frame.format.bytes,
                frame_rate=audio_frame.sample_rate,
                channels=len(audio_frame.layout.channels),
            )
            op = str(idx) + "omg" + str(audio_frame.to_ndarray().shape)
            text_output.write(op)
            sound_chunk += sound
            complete_audio_chunk += sound
    else:
        status_indicator.write("Audio Receiver is not set. Abort.")
        break

    complete_audio_chunk.export("/home/kp/temp/audio_file.wav", format="wav")
    logger.info("ENEDNEND call")
    st.write("end call")

I tried the above code, (copied from https://github.com/whitphx/streamlit-webrtc-example/blob/main/app.py). Only audio frames are of interest for me. But, audio is not saved in audio_file.wav at all.

Over resource limits on Streamlit Cloud

Hey there 👋 Just wanted to let you know that your app on Streamlit Cloud deployed from this repo has gone over its resource limits. Access to the app is temporarily limited. Visit the app to see more details and possible solutions.

Sushil

How to remotely record?

Hi,
Thanks for this wonderful toolkit, I tried to set up this demo on the server but always failed.
If I run the example locally, everything works perfectly (accessed by localhost:port), but if I want to use it remotely, say, deploying it on a server and accessing it using the IP:port, it doesn't work anymore. BTW, the demo page on hosted on the streamlit server you presented also failed for me. I would like to ask, do I need any extra setups? (I have already allowed the related permissions and the pop-up to allow the microphone can successfully be launched)
Thanks a lot!
Shuai

Hi, i'm working on voice translator app, and trying streamlit-webrtc with faster-whisper model
I noticed this link :https://github.com/whitphx/streamlit-webrtc/blob/main/pages/10_sendonly_audio.py
and found there is a while loop to handle audio data like below:

while True:
    if webrtc_ctx.audio_receiver:
        try:
            audio_frames = webrtc_ctx.audio_receiver.get_frames(timeout=1)
        except queue.Empty:
            logger.warning("Queue is empty. Abort.")
            break

        sound_chunk = pydub.AudioSegment.empty()
        for audio_frame in audio_frames:
            sound = pydub.AudioSegment(
                data=audio_frame.to_ndarray().tobytes(),
                sample_width=audio_frame.format.bytes,
                frame_rate=audio_frame.sample_rate,
                channels=len(audio_frame.layout.channels),
...

Doesn't it block UI thread since the while loop is in UI thread in my point view, and once i use below code, the page is pending.
Below is my code, please help to figure out how to fix it

webrtc_ctx = webrtc_streamer(
    key="speech-to-text",
    mode=WebRtcMode.SENDONLY,
    audio_receiver_size=10240,
    rtc_configuration={"iceServers": [{"urls": ["stun:stun.l.google.com:19302"]}]},
    media_stream_constraints={"video": False, "audio": True},
)

status_indicator = st.empty()
text_output = st.empty()
stream = None

while True:
    if webrtc_ctx.audio_receiver:
        sound_chunk = pydub.AudioSegment.empty()
        try:
            audio_frames = webrtc_ctx.audio_receiver.get_frames(timeout=1)
        except queue.Empty:
            time.sleep(0.1)
            status_indicator.write("No frame arrived.")

        status_indicator.write("Running. Say something!")

        for audio_frame in audio_frames:
            sound = pydub.AudioSegment(
                data=audio_frame.to_ndarray().tobytes(),
                sample_width=audio_frame.format.bytes,
                frame_rate=audio_frame.sample_rate,
                channels=len(audio_frame.layout.channels),
            )
            sound_chunk += sound
        print(sound_chunk)
        if len(sound_chunk) > 0:
            sound_chunk = sound_chunk.set_channels(1).set_frame_rate(44100)
            buffer = np.array(sound_chunk.get_array_of_samples())

            segments, _ = WhisperModel(model_size
                                       , device=f"{'cuda' if supports_gpu else 'cpu'}"
                                       , compute_type=compute_type).transcribe(buffer)
            transcript = " ".join(segment.text for segment in segments)
            print(transcript)
            text = " ".join(segment.text for segment in segments)
            text_output.markdown(f"**Text:** {text}")
    else:
        status_indicator.write("AudioReciver is not set. Abort.")