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• How to make media recorder api individual chunks playable by it self

  10 août 2024, par tedGuy

  I'm trying to send individual chunks to the server instead of sending the whole chunks at once. This way, I have Ffmpeg on my Rails server to convert these chunks to HLS and upload them to S3 to stream the video instantly. However, I've encountered an issue the Media Recorder only provides playable chunks for the first segment after that, they are not playable and need to be concatenated to play.

  


  To avoid this, I've taken a different approach where I start a new Media Recorder every 3 seconds so that I get a playable chunk every time. However, this approach has its issues the video glitches a bit due to the delay when I stop and start a new Media Recorder. Is there a way to achieve this with ease ? This is my current status. please help !

  


  const startVideoRecording = async (
    screenStream: MediaStream,
    audioStream: MediaStream
  ) => {
    setStartingRecording(true);

    try {
      const res = await getVideoId();
      videoId.current = res;
    } catch (error) {
      console.log(error);
      return;
    }

    const outputStream = new MediaStream();
    outputStream.addTrack(screenStream.getVideoTracks()[0]);
    outputStream.addTrack(audioStream.getAudioTracks()[0]); // Add audio track

    const mimeTypes = [
      "video/webm;codecs=h264",
      "video/webm;codecs=vp9",
      "video/webm;codecs=vp8",
      "video/webm",
      "video/mp4",
    ];

    let selectedMimeType = "";
    for (const mimeType of mimeTypes) {
      if (MediaRecorder.isTypeSupported(mimeType)) {
        selectedMimeType = mimeType;
        break;
      }
    }

    if (!selectedMimeType) {
      console.error("No supported mime type found");
      return;
    }

    const videoRecorderOptions = {
      mimeType: selectedMimeType,
    };

    let chunkIndex = 0;

    const startNewRecording = () => {
      // Stop the current recorder if it's running
      if (
        videoRecorderRef.current &&
        videoRecorderRef.current.state === "recording"
      ) {
        videoRecorderRef.current.stop();
      }

      // Create a new MediaRecorder instance
      const newVideoRecorder = new MediaRecorder(
        outputStream,
        videoRecorderOptions
      );
      videoRecorderRef.current = newVideoRecorder;

      newVideoRecorder.ondataavailable = async (event) => {
        if (event.data.size > 0) {
          chunkIndex++;
          totalSegments.current++;
          const segmentIndex = totalSegments.current;
          console.log(event.data);
          handleUpload({ segmentIndex, chunk: event.data });
        }
      };

      // Start recording with a 3-second interval
      newVideoRecorder.start();
    };

    // Start the first recording
    startNewRecording();

    // Set up an interval to restart the recording every 3 seconds
    recordingIntervalIdRef.current = setInterval(() => {
      startNewRecording();
    }, 3000);

    setIsRecording(true);
    setStartingRecording(false);
  };


  


• Capture multiple individual streams ALSA FFMPEG

  3 novembre 2017, par user3170450

  There are similar questions available but I couldn’t understand them properly as per my use case. As I am new to ALSA so I would like to explain my use case first.

  I am opening my application via google chrome and I need to to capture the audio being played in that chrome window. I did it successfully by capturing the speaker sound with following command :

  ffmpeg -f pulse -i alsa_output.pci-0000_00_1b.0.analog-stereo.monitor -ac 1 -ar 16000 test.wav

  But the real problem is I will be opening multiple windows at the same time and I need to capture each audio stream separately. There is something like Virtual Audio device but I don’t know how to configure and use them for my use case.

  Please guide me in right direction.

  Just for a note my system has only one physical sound card available which I am capturing for one window.

• FFMPEG - 2-pass video conversion final file size is larger than expected

  14 juin 2022, par Kevin Korb

  We are converting files and the target filesize is important. For one version of the video it can not be above 750KB.

  


  We are using the 2-pass encoding and specifying the target audio bitrate and target video bitrate, however the longer the duration of the video, the larger the file becomes and exceeds our 750KB target.

  


  


  For example, our file is 53.3 seconds, we know our final target of
750KB. We like our audio at 48kbps (mono).

  


  So we calculate our target (total) bitrate at : 750*8/53.3 = 112.57
total kbps Audio bitrate = 48kbps Video bitrate = 112-48 = 64kbps
video

  


  We would expect the file produced to be 746.2KB. (112*53.3 / 8)

  


  The file produced is 850k(ish).

  


  


  Is there something that takes up space in the file that isn't video or audio that we need to take into consideration ?