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• ffmpeg failed to load audio file

  14 avril 2024, par Vaishnav Ghenge

  Failed to load audio: ffmpeg version 5.1.4-0+deb12u1 Copyright (c) Failed to load audio: ffmpeg version 5.1.4-0+deb12u1 Copyright (c) 2000-2023 the FFmpeg developers
  built with gcc 12 (Debian 12.2.0-14)
  configuration: --prefix=/usr --extra-version=0+deb12u1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libdav1d --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libglslang --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librabbitmq --enable-librist --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libsvtav1 --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzimg --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --disable-sndio --enable-libjxl --enable-pocketsphinx --enable-librsvg --enable-libmfx --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libx264 --enable-libplacebo --enable-librav1e --enable-shared
  libavutil      57. 28.100 / 57. 28.100
  libavcodec     59. 37.100 / 59. 37.100
  libavformat    59. 27.100 / 59. 27.100
  libavdevice    59.  7.100 / 59.  7.100
  libavfilter     8. 44.100 /  8. 44.100
  libswscale      6.  7.100 /  6.  7.100
  libswresample   4.  7.100 /  4.  7.100
  libpostproc    56.  6.100 / 56.  6.100
/tmp/tmpjlchcpdm.wav: Invalid data found when processing input


  


  backend :

  


  
@app.route("/transcribe", methods=["POST"])
def transcribe():
    # Check if audio file is present in the request
    if 'audio_file' not in request.files:
        return jsonify({"error": "No file part"}), 400
    
    audio_file = request.files.get('audio_file')

    # Check if audio_file is sent in files
    if not audio_file:
        return jsonify({"error": "`audio_file` is missing in request.files"}), 400

    # Check if the file is present
    if audio_file.filename == '':
        return jsonify({"error": "No selected file"}), 400

    # Save the file with a unique name
    filename = secure_filename(audio_file.filename)
    unique_filename = os.path.join("uploads", str(uuid.uuid4()) + '_' + filename)
    # audio_file.save(unique_filename)
    
    # Read the contents of the audio file
    contents = audio_file.read()

    max_file_size = 500 * 1024 * 1024
    if len(contents) > max_file_size:
        return jsonify({"error": "File is too large"}), 400

    # Check if the file extension suggests it's a WAV file
    if not filename.lower().endswith('.wav'):
        # Delete the file if it's not a WAV file
        os.remove(unique_filename)
        return jsonify({"error": "Only WAV files are supported"}), 400

    print(f"\033[92m{filename}\033[0m")

    # Call Celery task asynchronously
    result = transcribe_audio.delay(contents)

    return jsonify({
        "task_id": result.id,
        "status": "pending"
    })


@celery_app.task
def transcribe_audio(contents):
    # Transcribe the audio
    try:
        # Create a temporary file to save the audio data
        with tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as temp_audio:
            temp_path = temp_audio.name
            temp_audio.write(contents)

            print(f"\033[92mFile temporary path: {temp_path}\033[0m")
            transcribe_start_time = time.time()

            # Transcribe the audio
            transcription = transcribe_with_whisper(temp_path)
            
            transcribe_end_time = time.time()
            print(f"\033[92mTranscripted text: {transcription}\033[0m")

            return transcription, transcribe_end_time - transcribe_start_time

    except Exception as e:
        print(f"\033[92mError: {e}\033[0m")
        return str(e)


  


  frontend :

  


      useEffect(() => {
        const init = () => {
            navigator.mediaDevices.getUserMedia({audio: true})
                .then((audioStream) => {
                    const recorder = new MediaRecorder(audioStream);

                    recorder.ondataavailable = e => {
                        if (e.data.size > 0) {
                            setChunks(prevChunks => [...prevChunks, e.data]);
                        }
                    };

                    recorder.onerror = (e) => {
                        console.log("error: ", e);
                    }

                    recorder.onstart = () => {
                        console.log("started");
                    }

                    recorder.start();

                    setStream(audioStream);
                    setRecorder(recorder);
                });
        }

        init();

        return () => {
            if (recorder && recorder.state === 'recording') {
                recorder.stop();
            }

            if (stream) {
                stream.getTracks().forEach(track => track.stop());
            }
        }
    }, []);

    useEffect(() => {
        // Send chunks of audio data to the backend at regular intervals
        const intervalId = setInterval(() => {
            if (recorder && recorder.state === 'recording') {
                recorder.requestData(); // Trigger data available event
            }
        }, 8000); // Adjust the interval as needed


        return () => {
            if (intervalId) {
                console.log("Interval cleared");
                clearInterval(intervalId);
            }
        };
    }, [recorder]);

    useEffect(() => {
        const processAudio = async () => {
            if (chunks.length > 0) {
                // Send the latest chunk to the server for transcription
                const latestChunk = chunks[chunks.length - 1];

                const audioBlob = new Blob([latestChunk]);
                convertBlobToAudioFile(audioBlob);
            }
        };

        void processAudio();
    }, [chunks]);

    const convertBlobToAudioFile = useCallback((blob: Blob) => {
        // Convert Blob to audio file (e.g., WAV)
        // This conversion may require using a third-party library or service
        // For example, you can use the MediaRecorder API to record audio in WAV format directly
        // Alternatively, you can use a library like recorderjs to perform the conversion
        // Here's a simplified example using recorderjs:

        const reader = new FileReader();
        reader.onload = () => {
            const audioBuffer = reader.result; // ArrayBuffer containing audio data

            // Send audioBuffer to Flask server or perform further processing
            sendAudioToFlask(audioBuffer as ArrayBuffer);
        };

        reader.readAsArrayBuffer(blob);
    }, []);

    const sendAudioToFlask = useCallback((audioBuffer: ArrayBuffer) => {
        const formData = new FormData();
        formData.append('audio_file', new Blob([audioBuffer]), `speech_audio.wav`);

        console.log(formData.get("audio_file"));

        fetch('http://34.87.75.138:8000/transcribe', {
            method: 'POST',
            body: formData
        })
            .then(response => response.json())
            .then((data: { task_id: string, status: string }) => {
                pendingTaskIdsRef.current.push(data.task_id);
            })
            .catch(error => {
                console.error('Error sending audio to Flask server:', error);
            });
    }, []);


  


  I was trying to pass the audio from frontend to whisper model which is in flask app

  


• FFMPEG throws error while converting WEBM/MP4 audio files from MediaRecorder to MP3

  27 septembre 2022, par Sahil Malik

  I use MediaRecorder to record the audio from the browser and then upload it to my server (ARM-based Linux machine - AWS Lambda Function, if it matters). Based on the browser type, I get either an MP4 file (for the Safari browser) OR a WEBM file (for every other browser) from the MediaRecorder. The audio is converted to Base64 string and posted to my server with FFMPEG.

  


  MediaRecorder implementation

  


  const audioStream = await navigator.mediaDevices.getUserMedia(mediaConstraints);
const MediaRecorder = window['MediaRecorder'];
const mimeType = MediaRecorder.isTypeSupported('audio/webm') ? 'audio/webm' : 'audio/mp4';
const recordingFileExtension = MediaRecorder.isTypeSupported('audio/webm') ? 'webm' : 'mp4';
mediaRecorder = new MediaRecorder(audioStream, { mimeType });
mediaRecorder.ondataavailable = convertBlobAndUploadChunk;
mediaRecorder.start(30 * 1000); // timeslice needs to be in ms
// When user stops recording
mediaRecorder.stop();
audioStream.getTracks().forEach( t => { t.stop(); });
// To convert the audio blob to string
function convertBlobToBase64(blob) {
    return new Promise((resolve, reject) => {
        const reader = new FileReader();
        reader.readAsDataURL(blob);
        reader.onload = () => {
            const tmpStr = reader.result.toString();
            resolve(tmpStr.substring(tmpStr.indexOf('base64,') + 7));
        };
        reader.onerror = error => reject(error);
    });
}

let blobCount = 0;
async function convertBlobAndUploadChunk(blobEvent) {
    if (!blobEvent.data || blobEvent.data.size === 0) return;

    blobCount++;

    const recordData = JSON.stringify({
        M: {
            blobCount,
            //some other meta data
        },
        D: await convertBlobToBase64(audioBlob),
    });

    await angularHttpClient.post(apiUrl, recordData, new HttpHeaders({
        'Content-Type': 'application/json',
        'x-api-key': apiKey,
    })).toPromise();
}


  


  On my server, I convert the WEBM/MP4 file to an MP3 file for better cross-browser compatibility and to enable scrubbing.

  


  Backend Lambda Implementation

  


  const FFMpegCommand = require('fluent-ffmpeg'); // v2.1.2
new FFMpegCommand()
        .input(originalFile)
        .on('end', (error, stdOut, stdError) => {
            if (error) {
                console.error(error);
            }
        })
        .save(convertedFile);


  


  This works perfectly fine 99.9% of the time but FFMPEG throws one of the following errors for 0.1% of the time :

  


  Error 1 : Invalid data found when processing input

  


  error reading header

  


  ffmpeg version 4.4-static https://johnvansickle.com/ffmpeg/ Copyright (c) 2000-2021 the 
FFmpeg developers
built with gcc 8 (Debian 8.3.0-6)
configuration: --enable-gpl --enable-version3 --enable-static --disable-debug --disable-ffplay --disable-indev=sndio --disable-outdev=sndio --cc=gcc --enable-fontconfig --enable-frei0r --enable-gnutls --enable-gmp --enable-libgme --enable-gray --enable-libaom --enable-libfribidi --enable-libass --enable-libvmaf --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librubberband --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libvorbis --enable-libopus --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libdav1d --enable-libxvid --enable-libzvbi --enable-libzimg
libavutil 56. 70.100 / 56. 70.100
libavcodec 58.134.100 / 58.134.100
libavformat 58. 76.100 / 58. 76.100
libavdevice 58. 13.100 / 58. 13.100
libavfilter 7.110.100 / 7.110.100
libswscale 5. 9.100 / 5. 9.100
libswresample 3. 9.100 / 3. 9.100
libpostproc 55. 9.100 / 55. 9.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7232f40] could not find corresponding trex (id 1)
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7232f40] could not find corresponding track id 0
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7232f40] trun track id unknown, no tfhd was found
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7232f40] error reading header
/tmp/long-filename-of-140chars-to-keep-files-unique.mp4: Invalid data found when processing input


  


  EBML Header parsing failed

  


  ffmpeg version 4.4-static https://johnvansickle.com/ffmpeg/ Copyright (c) 2000-2021 the FFmpeg developers
built with gcc 8 (Debian 8.3.0-6)
configuration: --enable-gpl --enable-version3 --enable-static --disable-debug --disable-ffplay --disable-indev=sndio --disable-outdev=sndio --cc=gcc --enable-fontconfig --enable-frei0r --enable-gnutls --enable-gmp --enable-libgme --enable-gray --enable-libaom --enable-libfribidi --enable-libass --enable-libvmaf --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librubberband --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libvorbis --enable-libopus --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libdav1d --enable-libxvid --enable-libzvbi --enable-libzimg
libavutil 56. 70.100 / 56. 70.100
libavcodec 58.134.100 / 58.134.100
libavformat 58. 76.100 / 58. 76.100
libavdevice 58. 13.100 / 58. 13.100
libavfilter 7.110.100 / 7.110.100
libswscale 5. 9.100 / 5. 9.100
libswresample 3. 9.100 / 3. 9.100
libpostproc 55. 9.100 / 55. 9.100
[matroska,webm @ 0x7595f40] Format matroska,webm detected only with low score of 1, misdetection possible!
[matroska,webm @ 0x7595f40] EBML header parsing failed
/tmp/long-filename-of-140chars-to-keep-files-unique.webm: Invalid data found when processing input


  


  No specific details

  


  ffmpeg version 4.4-static https://johnvansickle.com/ffmpeg/ Copyright (c) 2000-2021 the FFmpeg developers
built with gcc 8 (Debian 8.3.0-6)
configuration: --enable-gpl --enable-version3 --enable-static --disable-debug --disable-ffplay --disable-indev=sndio --disable-outdev=sndio --cc=gcc --enable-fontconfig --enable-frei0r --enable-gnutls --enable-gmp --enable-libgme --enable-gray --enable-libaom --enable-libfribidi --enable-libass --enable-libvmaf --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librubberband --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libvorbis --enable-libopus --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libdav1d --enable-libxvid --enable-libzvbi --enable-libzimg
libavutil 56. 70.100 / 56. 70.100
libavcodec 58.134.100 / 58.134.100
libavformat 58. 76.100 / 58. 76.100
libavdevice 58. 13.100 / 58. 13.100
libavfilter 7.110.100 / 7.110.100
libswscale 5. 9.100 / 5. 9.100
libswresample 3. 9.100 / 3. 9.100
libpostproc 55. 9.100 / 55. 9.100
/tmp/long-filename-of-140chars-to-keep-files-unique.webm: Invalid data found when processing input


  


  Error 2 : Output file #0 does not contain any stream

  


  ffmpeg version 4.4-static https://johnvansickle.com/ffmpeg/ Copyright (c) 2000-2021 the FFmpeg developers
built with gcc 8 (Debian 8.3.0-6)
configuration: --enable-gpl --enable-version3 --enable-static --disable-debug --disable-ffplay --disable-indev=sndio --disable-outdev=sndio --cc=gcc --enable-fontconfig --enable-frei0r --enable-gnutls --enable-gmp --enable-libgme --enable-gray --enable-libaom --enable-libfribidi --enable-libass --enable-libvmaf --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librubberband --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libvorbis --enable-libopus --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libdav1d --enable-libxvid --enable-libzvbi --enable-libzimg
libavutil 56. 70.100 / 56. 70.100
libavcodec 58.134.100 / 58.134.100
libavformat 58. 76.100 / 58. 76.100
libavdevice 58. 13.100 / 58. 13.100
libavfilter 7.110.100 / 7.110.100
libswscale 5. 9.100 / 5. 9.100
libswresample 3. 9.100 / 3. 9.100
libpostproc 55. 9.100 / 55. 9.100
[mpegts @ 0x5d19f40] Format mpegts detected only with low score of 2, misdetection possible!
[mpegts @ 0x5d19f40] Could not detect TS packet size, defaulting to non-FEC/DVHS
Input #0, mpegts, from '/tmp/long-filename-of-140chars-to-keep-files-unique.webm':
Duration: N/A, bitrate: N/A
Output #0, mp3, to '/tmp/long-filename-of-140chars-to-keep-files-unique.mp3':
Output file #0 does not contain any stream


  


  Thanks for reading, any help/suggestion is highly appreciated.

  


• Stream sent via FFMPEG (NodeJS) to RTMP (YouTube) not being received

  10 décembre 2024, par Qumber

  I am writing a very basic chrome extension that captures and sends video stream to a nodeJS server, which in turns sends it to Youtube live server.

  


  Here is my implementation of the backend which receives data via WebRTC and send to YT using FFMPEG :

  


  const express = require('express');
const cors = require('cors');
const { RTCPeerConnection, RTCSessionDescription } = require('@roamhq/wrtc');
const { spawn } = require('child_process');

const app = express();
app.use(express.json());
app.use(cors());

app.post('/webrtc', async (req, res) => {
  const peerConnection = new RTCPeerConnection();

  // Start ffmpeg process for streaming
  const ffmpeg = spawn('ffmpeg', [
    '-f', 'flv',
    '-i', 'pipe:0',
    '-c:v', 'libx264',
    '-preset', 'veryfast',
    '-maxrate', '3000k',
    '-bufsize', '6000k',
    '-pix_fmt', 'yuv420p',
    '-g', '50',
    '-f', 'flv',
    'rtmp://a.rtmp.youtube.com/live2/MY_KEY'
  ]);

  ffmpeg.on('error', (err) => {
    console.error('FFmpeg error:', err);
  });

  ffmpeg.stderr.on('data', (data) => {
    console.error('FFmpeg stderr:', data.toString());
  });

  ffmpeg.stdout.on('data', (data) => {
    console.log('FFmpeg stdout:', data.toString());
  });

  // Handle incoming tracks
  peerConnection.ontrack = (event) => {
    console.log('Track received:', event.track.kind);
    const track = event.track;

    // Stream the incoming track to FFmpeg
    track.onunmute = () => {
      console.log('Track unmuted:', track.kind);
      const reader = track.createReadStream();
      reader.on('data', (chunk) => {
        console.log('Forwarding chunk to FFmpeg:', chunk.length);
        ffmpeg.stdin.write(chunk);
      });
      reader.on('end', () => {
        console.log('Stream ended');
        ffmpeg.stdin.end();
      });
    };

    track.onmute = () => {
      console.log('Track muted:', track.kind);
    };
  };

  // Set the remote description (offer) received from the client
  await peerConnection.setRemoteDescription(new RTCSessionDescription(req.body.sdp));

  // Create an answer and send it back to the client
  const answer = await peerConnection.createAnswer();
  await peerConnection.setLocalDescription(answer);

  res.json({ sdp: peerConnection.localDescription });
});

app.listen(3000, () => {
  console.log('WebRTC to RTMP server running on port 3000');
});



  


  This is the output I get, but nothing gets sent to YouTube :

  


  


  FFmpeg stderr: ffmpeg version 7.0.2 Copyright (c) 2000-2024 the FFmpeg developers
  built with Apple clang version 15.0.0 (clang-1500.3.9.4)

FFmpeg stderr:   configuration: --prefix=/opt/homebrew/Cellar/ffmpeg/7.0.2_1 --enable-shared --enable-pthreads --enable-version3 --cc=clang --host-cflags= --host-ldflags='-Wl,-ld_classic' --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libaribb24 --enable-libbluray --enable-libdav1d --enable-libharfbuzz --enable-libjxl --enable-libmp3lame --enable-libopus --enable-librav1e --enable-librist --enable-librubberband --enable-libsnappy --enable-libsrt --enable-libssh --enable-libsvtav1 --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libspeex --enable-libsoxr --enable-libzmq --enable-libzimg --disable-libjack --disable-indev=jack --enable-videotoolbox --enable-audiotoolbox --enable-neon

FFmpeg stderr:   libavutil      59.  8.100 / 59.  8.100
  libavcodec     61.  3.100 / 61.  3.100
  libavformat    61.  1.100 / 61.  1.100
  libavdevice    61.  1.100 / 61.  1.100

FFmpeg stderr:   libavfilter    10.  1.100 / 10.  1.100
  libswscale      8.  1.100 /  8.  1.100
  libswresample   5.  1.100 /  5.  1.100
  libpostproc    58.  1.100 / 58.  1.100


  


  


  I do not understand what I am doing wrong. Any help would be appreciated.

  


  ---

  


  Optionally Here's the frontend code from the extension, which (to me) appears to be recording and sending the capture :

  


  popup.js & popup.html

  


  


  


  document.addEventListener('DOMContentLoaded', () => {
  document.getElementById('openCapturePage').addEventListener('click', () => {
    chrome.tabs.create({
      url: chrome.runtime.getURL('capture.html')
    });
  });
});

  


  &#xA;&#xA;&#xA;&#xA;  &#xA;  <code class="echappe-js">&lt;script src='http://stackoverflow.com/feeds/tag/popup.js'&gt;&lt;/script&gt;&#xA;&#xA;&#xA;&#xA;  
  StreamSavvy
  &#xA;
  Open Capture Page
  &#xA;&#xA;&#xA;
  

  


  


  



  capture.js & capture.html

  


  


  


  let peerConnection;

async function startStreaming() {
  try {
    const stream = await navigator.mediaDevices.getDisplayMedia({
      video: {
        cursor: "always"
      },
      audio: false
    });

    peerConnection = new RTCPeerConnection({
      iceServers: [{
        urls: 'stun:stun.l.google.com:19302'
      }]
    });

    stream.getTracks().forEach(track => peerConnection.addTrack(track, stream));

    const offer = await peerConnection.createOffer();
    await peerConnection.setLocalDescription(offer);

    const response = await fetch('http://localhost:3000/webrtc', {
      method: 'POST',
      headers: {
        'Content-Type': 'application/json'
      },
      body: JSON.stringify({
        sdp: peerConnection.localDescription
      })
    });

    const {
      sdp
    } = await response.json();
    await peerConnection.setRemoteDescription(new RTCSessionDescription(sdp));

    console.log("Streaming to server via WebRTC...");
  } catch (error) {
    console.error("Error starting streaming:", error.name, error.message);
  }
}

async function stopStreaming() {
  if (peerConnection) {
    // Stop all media tracks
    peerConnection.getSenders().forEach(sender => {
      if (sender.track) {
        sender.track.stop();
      }
    });

    // Close the peer connection
    peerConnection.close();
    peerConnection = null;
    console.log("Streaming stopped");
  }
}

document.addEventListener('DOMContentLoaded', () => {
  document.getElementById('startCapture').addEventListener('click', startStreaming);
  document.getElementById('stopCapture').addEventListener('click', stopStreaming);
});

  


  &#xA;&#xA;&#xA;&#xA;  &#xA;  <code class="echappe-js">&lt;script src='http://stackoverflow.com/feeds/tag/capture.js'&gt;&lt;/script&gt;&#xA;&#xA;&#xA;&#xA;  
  StreamSavvy Capture
  &#xA;
  Start Capture
  &#xA;
  Stop Capture
  &#xA;&#xA;&#xA;
  

  


  


  



  background.js (service worker)

  


  


  


  chrome.runtime.onInstalled.addListener(() => {
  console.log("StreamSavvy Extension Installed");
});

chrome.runtime.onMessage.addListener((message, sender, sendResponse) => {
  if (message.type === 'startStreaming') {
    chrome.tabs.create({
      url: chrome.runtime.getURL('capture.html')
    });
    sendResponse({
      status: 'streaming'
    });
  } else if (message.type === 'stopStreaming') {
    chrome.tabs.query({
      url: chrome.runtime.getURL('capture.html')
    }, (tabs) => {
      if (tabs.length > 0) {
        chrome.tabs.sendMessage(tabs[0].id, {
          type: 'stopStreaming'
        });
        sendResponse({
          status: 'stopped'
        });
      }
    });
  }
  return true; // Keep the message channel open for sendResponse
});