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• ffmpeg hangs when muxing longer videos

  11 août 2020, par 492357816

  I am using ffmpeg with Janus WebRTC server for muxing video conference recordings. For video conferences under approximaately 5 minutes duration, existing code works well. If the source files are longer than that the process appears to hang without returning either an error or a completed file.
Any advice/suggestions would be most welcome.

  


      var recordingsPath = exports.recordingsPath;
    var videoRecordingPath = `${recordingsPath}/${filename}-video.mjr`;
    var audioRecordingPath = `${recordingsPath}/${filename}-audio.mjr`;
    var videoOutputPath = `${recordingsPath}/${filename}-output1.mp4`;
    var audioOutputPath = `${recordingsPath}/${filename}-output1.wav`;
    var finalOutputPath = `${recordingsPath}/${filename}-final.mp4`;
    var rawConverterPath = "/opt/janus/bin/janus-pp-rec";
    var audioRecordingByteSize = exports.getFileSizeInBytes(audioRecordingPath);
    var audioIsValid = audioRecordingByteSize > 8;

    if (audioIsValid) {
        //turn audio and video mjr files into .wav and mp4 respectively
        exports.runBashSync([ rawConverterPath, videoRecordingPath, videoOutputPath]);
        exports.runBashSync([ rawConverterPath, audioRecordingPath, audioOutputPath]);
        return exports.mergeMp4AndWav(
            audioOutputPath, videoOutputPath, finalOutputPath
        ).then(function(result) {
            // remove source data for audio & video
            exports.runBashSync([ "rm", audioOutputPath]);
            exports.runBashSync([ "rm", videoOutputPath]);
            exports.runBashSync([ "rm", audioRecordingPath]);
            exports.runBashSync([ "rm", videoRecordingPath]);
        });
    } else {
        // handle cases where audio is not available
        feedback("no audio");
        return new Promise(function(resolve) {
            exports.runBashSync([ rawConverterPath, videoRecordingPath, finalOutputPath], true);
            if (Number.isInteger(audioRecordingByteSize)) {
                exports.runBashSync([ "rm", audioRecordingPath]);
            }
            exports.runBashSync([ "rm", videoRecordingPath]);
            resolve(true);
        });
    }
};

exports.joinMp4s = function(mp4Filenames, outputPath) {
    feedback("joining mp4s");
    if (mp4Filenames.length === 1) {
        feedback("single-stream case");
        exports.runBashSync(["mv", mp4Filenames[0], outputPath]);
        return new Promise(function(resolve) { resolve(true); });
    }
    feedback("multi-stream case");
    var joinCmd = ["ffmpeg"];
    mp4Filenames.forEach(function(filename) {
        joinCmd.push(`-i ${filename}`);
    });
    joinCmd.push("-strict -2");
    var totalHeight = 960;
    var totalWidth = 1280;

    joinCmd.push(`-filter_complex "color=size=${totalWidth}x${totalHeight}:c=Black [base];`);

    // var numStrms = mp4Filenames.length;
    var streamsPattern = exports.setStreamsPattern(mp4Filenames);
    var strmHeight = streamsPattern.height;
    var strmWidth = streamsPattern.width;
    var strmDmns = `${strmWidth}x${strmHeight}`;
    feedback("streamDimensions: " + strmDmns);

    var i;
    for (i = 0; i < mp4Filenames.length; i++) {
        joinCmd.push(`[${i}:v] setpts=PTS-STARTPTS, scale=${strmDmns} [temp${i}];`);
    }
    for (i = 0; i < mp4Filenames.length; i++) {
        var xCoord;
        var yCoord;
        if (i === 0) {
            xCoord = streamsPattern.coords[i].xCoord;
            yCoord = streamsPattern.coords[i].yCoord;
            joinCmd.push(`[base][temp${i}] overlay=shortest=1:x=${xCoord}:y=${yCoord} [tmp${i + 1}];`);
            // joinCmd.push(`[base][temp${i}] overlay=shortest=1 [tmp${i + 1}];`);
        } else {
            var cmd = `[tmp${i}][temp${i}] overlay=shortest=1`;
            xCoord = streamsPattern.coords[i].xCoord;
            yCoord = streamsPattern.coords[i].yCoord;

            if (xCoord) { cmd += ":x=" + xCoord; }
            if (yCoord) { cmd += ":y=" + yCoord; }

            if (i + 1 !== mp4Filenames.length) { cmd += ` [tmp${i + 1}];`; }
            joinCmd.push(cmd);
        }
    }

    joinCmd.push(`" ${outputPath}`);
    feedback("join command: " + joinCmd);
    return exports.runBashAsync(joinCmd).then(function(result) {

        mp4Filenames.forEach(function(filename) {
            feedback("removing: " + filename);
            exports.runBashSync(`rm ${filename}`);
        });
    });
};



  


• FFMPEG OpenTok archive audio drift

  25 août 2020, par Chris Kooken

  I am using OpenTok to build a live video platform. It generates webm files from each users stream.

  


  I am using FFMPEG to convert webm (WEBRTC) videos to MP4s to edit in my NLE. The problem I am having is my audio is drifting. I THINK it is because the user pauses the audio during the stream. This is the command i'm running

  


  ffmpeg -acodec libopus -i 65520df3-1033-480e-adde-1856d18e2352.webm  -max_muxing_queue_size 99999999 65520df3-1033-480e-adde-1856d18e2352.webm.new.mp4


  


  The problem is I think, whenever the user muted themselves, there are no frames. But the PTS is in tact.

  


  This is from the OpenTok documentation (my WebRTC platform)

  


  


  Audio and video frames may not arrive with monotonic timestamps ; frame rates are not always consistent. This is especially relevant if either the video or audio track is disabled for a time, using one of publishVideo or publishAudio publisher properties.

  


  


  


  Frame presentation timestamps (PTS) are written based on NTP
timestamps taken at the time of capture, offset by the timestamp of
the first received frame. Even if a track is muted and later unmuted,
the timestamp offset should remain consistent throughout the duration
of the entire stream. When decoding in post-processing, a gap in PTS
between consecutive frames will exist for the duration of the track
mute : there are no "silent" frames in the container.

  


  


  How can I convert these files and have them play in sync ? Note, when I play in QuickTime or VLC, the files are synced correctly.

  


  EDIT
I've gotten pretty close with this command :

  


   ffmpeg -acodec libopus -i $f -max_muxing_queue_size 99999999 -vsync 1 -af aresample=async=1 -r 30 $f.mp4


  


  But every once in a while, I get a video where the audio starts right away, and they wont actually be talking in the video until half-way thought the video. My guess is they muted themselves during the video conference... so in some cases audio is 5-10 mins ahead. Again, plays fine in quicktime, but pulled into my NLE, its way off.

  


• FFMPEG audio not lining up

  31 août 2020, par Chris Kooken

  I am using OpenTok to build a live video platform. It generates webm files from each users stream.

  


  I am using FFMPEG to convert webm (WEBRTC) videos to MP4s to edit in my NLE. The problem I am having is my audio is drifting. I THINK it is because the user pauses the audio during the stream. This is the command i'm running

  


  ffmpeg -acodec libopus -i 65520df3-1033-480e-adde-1856d18e2352.webm  -max_muxing_queue_size 99999999 65520df3-1033-480e-adde-1856d18e2352.webm.new.mp4


  


  The problem is I think, whenever the user muted themselves, there are no frames. But the PTS is in tact.

  


  This is from the OpenTok documentation (my WebRTC platform)

  


  


  Audio and video frames may not arrive with monotonic timestamps ; frame rates are not always consistent. This is especially relevant if either the video or audio track is disabled for a time, using one of publishVideo or publishAudio publisher properties.

  


  


  


  Frame presentation timestamps (PTS) are written based on NTP
timestamps taken at the time of capture, offset by the timestamp of
the first received frame. Even if a track is muted and later unmuted,
the timestamp offset should remain consistent throughout the duration
of the entire stream. When decoding in post-processing, a gap in PTS
between consecutive frames will exist for the duration of the track
mute : there are no "silent" frames in the container.

  


  


  How can I convert these files and have them play in sync ? Note, when I play in QuickTime or VLC, the files are synced correctly.

  


  EDIT
I've gotten pretty close with this command :

  


   ffmpeg -acodec libopus -i $f -max_muxing_queue_size 99999999 -vsync 1 -af aresample=async=1 -r 30 $f.mp4


  


  But every once in a while, I get a video where the audio starts right away, and they wont actually be talking in the video until half-way thought the video. My guess is they muted themselves during the video conference... so in some cases audio is 5-10 mins ahead. Again, plays fine in quicktime, but pulled into my NLE, its way off.