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• 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
});

  


  


  



• ffmpeg concat doesn't keep video speed/framerate ?

  28 août 2016, par Nick

  I have a bunch of small webm clips in directory vidchunks. These clips were generated using javascript MediaRecorder API.

  MediaRecorder code :

  var mRecorder;
  
  var chunks = [];
  
  navigator.mediaDevices.getUserMedia({
  
    audio: false,
  
    video: {
  
    width: 1280,
  
    height: 760,
  
    mozMediaSource: "screen",
  
    mediaSource: "screen"
  
  }
  
  }).then(function(stream) {
  
    mRecorder = new MediaRecorder(stream,{mimeType:"video/webm"});
  
    mRecorder.ondataavailable = function(event) {
  
      var blob = event.data;
  
      chunks.push(blob);
  
      var vidchunk = new Blob(chunks);
  
      // save to directory vid chunk.
  
    };
  
    mRecorder.start(5000);
  
  }).catch(function(error) {
  
    console.log(error.message);
  
  });

  Then, once I have a bunch of these 5 second clips, I merge them :

  ffmpeg -safe 0 \
  
  -f concat -i <(for f in `ls vidchunks/* | sort -V`; do echo file $f; done) \
  
  -c:v copy -r 30 -vsync drop test.webm

  When I play back these chunks individually, they play correctly. All 5 seconds of video are rendered smoothly.

  However, once I merge, some 5 second chunks are condensed to like 1 second. So the total video, which should be 50 seconds for 10 chunks, turns out to be like 38 seconds because three of the chunks got condensed to 1 second.

  Any ideas on how to fix this ?

  EDIT : I’ve tried :

  • without "-r 30 -vsync drop"
  • without "-c:v copy"

  No changes...

  EDIT2 (some ffprobe outputs) :

  ffprobe 1472343170-1.webm

  ffprobe version 2.8.6-1ubuntu2 Copyright (c) 2007-2016 the FFmpeg developers
  
    built with gcc 5.3.1 (Ubuntu 5.3.1-11ubuntu1) 20160311
  
    configuration: --prefix=/usr --extra-version=1ubuntu2 --build-suffix=-ffmpeg --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --cc=cc --cxx=g++ --enable-gpl --enable-shared --disable-stripping --disable-decoder=libopenjpeg --disable-decoder=libschroedinger --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-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librtmp --enable-libschroedinger --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-libxvid --enable-libzvbi --enable-openal --enable-opengl --enable-x11grab --enable-libdc1394 --enable-libiec61883 --enable-libzmq --enable-frei0r --enable-libx264 --enable-libopencv
  
    libavutil      54. 31.100 / 54. 31.100
  
    libavcodec     56. 60.100 / 56. 60.100
  
    libavformat    56. 40.101 / 56. 40.101
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 40.101 /  5. 40.101
  
    libavresample   2.  1.  0 /  2.  1.  0
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  2.101 /  1.  2.101
  
    libpostproc    53.  3.100 / 53.  3.100
  
  Input #0, matroska,webm, from '1472343170-1.webm':
  
    Metadata:
  
      encoder         : Chrome
  
    Duration: N/A, start: 0.000000, bitrate: N/A
  
      Stream #0:0(eng): Video: vp8, yuv420p, 1280x760, SAR 1:1 DAR 32:19, 30 fps, 30 tbr, 1k tbn, 1k tbc (default)

  ffprobe 1472343245-16.webm

  ffprobe version 2.8.6-1ubuntu2 Copyright (c) 2007-2016 the FFmpeg developers
  
    built with gcc 5.3.1 (Ubuntu 5.3.1-11ubuntu1) 20160311
  
    configuration: --prefix=/usr --extra-version=1ubuntu2 --build-suffix=-ffmpeg --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --cc=cc --cxx=g++ --enable-gpl --enable-shared --disable-stripping --disable-decoder=libopenjpeg --disable-decoder=libschroedinger --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-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librtmp --enable-libschroedinger --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-libxvid --enable-libzvbi --enable-openal --enable-opengl --enable-x11grab --enable-libdc1394 --enable-libiec61883 --enable-libzmq --enable-frei0r --enable-libx264 --enable-libopencv
  
    libavutil      54. 31.100 / 54. 31.100
  
    libavcodec     56. 60.100 / 56. 60.100
  
    libavformat    56. 40.101 / 56. 40.101
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 40.101 /  5. 40.101
  
    libavresample   2.  1.  0 /  2.  1.  0
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  2.101 /  1.  2.101
  
    libpostproc    53.  3.100 / 53.  3.100
  
  Input #0, matroska,webm, from '1472343245-16.webm':
  
    Metadata:
  
      encoder         : Chrome
  
    Duration: N/A, start: 0.000000, bitrate: N/A
  
      Stream #0:0(eng): Video: vp8, yuv420p, 1280x760, SAR 1:1 DAR 32:19, 30 fps, 30 tbr, 1k tbn, 1k tbc (default)

  ffprobe 1472343350-37.webm

  ffprobe version 2.8.6-1ubuntu2 Copyright (c) 2007-2016 the FFmpeg developers
  
    built with gcc 5.3.1 (Ubuntu 5.3.1-11ubuntu1) 20160311
  
    configuration: --prefix=/usr --extra-version=1ubuntu2 --build-suffix=-ffmpeg --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --cc=cc --cxx=g++ --enable-gpl --enable-shared --disable-stripping --disable-decoder=libopenjpeg --disable-decoder=libschroedinger --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-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librtmp --enable-libschroedinger --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-libxvid --enable-libzvbi --enable-openal --enable-opengl --enable-x11grab --enable-libdc1394 --enable-libiec61883 --enable-libzmq --enable-frei0r --enable-libx264 --enable-libopencv
  
    libavutil      54. 31.100 / 54. 31.100
  
    libavcodec     56. 60.100 / 56. 60.100
  
    libavformat    56. 40.101 / 56. 40.101
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 40.101 /  5. 40.101
  
    libavresample   2.  1.  0 /  2.  1.  0
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  2.101 /  1.  2.101
  
    libpostproc    53.  3.100 / 53.  3.100
  
  Input #0, matroska,webm, from '1472343350-37.webm':
  
    Metadata:
  
      encoder         : Chrome
  
    Duration: N/A, start: 0.000000, bitrate: N/A
  
      Stream #0:0(eng): Video: vp8, yuv420p, 1280x760, SAR 1:1 DAR 32:19, 30 fps, 30 tbr, 1k tbn, 1k tbc (default)

  I also noticed that the merged video plays normal when the mouse is moving around on the screen, but when the mouse is not moving, it moves 5x faster.

  I know this because I have a timer on the screen that I’m recording, and I can see that the timer speeds up significantly when mouse isn’t moving but when mouse is moving, the timer moves at normal speed.

  EDIT3 :
  The DTS/PTS looks like resets after each file is merged :

  DTS -9223363446915256, next:6866667 st:0 invalid dropping
  
  PTS -9223363446915256, next:6866667 invalid dropping st:0
  
  DTS -9223363446915222, next:6900000 st:0 invalid dropping
  
  PTS -9223363446915222, next:6900000 invalid dropping st:0
  
  DTS -9223363446915188, next:6933333 st:0 invalid dropping
  
  PTS -9223363446915188, next:6933333 invalid dropping st:0
  
  [concat @ 0x1cde3e0] DTS -9223363446920184 < 5006 out of order
  
  DTS -9223363446920184, next:5033333 st:0 invalid dropping
  
  PTS -9223363446920184, next:5033333 invalid dropping st:0
  
  DTS -9223363446920056, next:5066667 st:0 invalid dropping
  
  PTS -9223363446920056, next:5066667 invalid dropping st:0
  
  DTS -9223363446919956, next:5100000 st:0 invalid dropping
  
  PTS -9223363446919956, next:5100000 invalid dropping st:0

  EDIT4 : Tried ffmpeg -i "$f" -y -c copy -fflags +genpts "$f" and then merge again.

  Works a lot better, but now other files are getting skipped.

  Here’s ffprobe from a file that’s skipped :

  Metadata:
  
      encoder         : Lavf56.40.101
  
    Duration: 00:00:01.17, start: 0.000000, bitrate: 428 kb/s
  
      Stream #0:0(eng): Video: vp8, yuv420p, 1280x760, SAR 1:1 DAR 32:19, 30 fps, 30 tbr, 1k tbn, 1k tbc (default)
  
  [webm @ 0x1c01940] Codec for stream 0 does not use global headers but container format requires global headers
  
  Output #0, webm, to '1472343225-12.webm':
  
    Metadata:
  
      encoder         : Lavf56.40.101
  
      Stream #0:0(eng): Video: vp8, yuv420p, 1280x760 [SAR 1:1 DAR 32:19], q=2-31, 30 fps, 30 tbr, 1k tbn, 1k tbc (default)
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)

  Here’s ffprobe from a file not skipped :

  Input #0, matroska,webm, from '1472343215-10.webm':
  
    Metadata:
  
      encoder         : Lavf56.40.101
  
    Duration: 00:00:05.03, start: 0.000000, bitrate: 80 kb/s
  
      Stream #0:0(eng): Video: vp8, yuv420p, 1280x760, SAR 1:1 DAR 32:19, 30 fps, 30 tbr, 1k tbn, 1k tbc (default)
  
  [webm @ 0x1349940] Codec for stream 0 does not use global headers but container format requires global headers
  
  Output #0, webm, to '1472343215-10.webm':
  
    Metadata:
  
      encoder         : Lavf56.40.101
  
      Stream #0:0(eng): Video: vp8, yuv420p, 1280x760 [SAR 1:1 DAR 32:19], q=2-31, 30 fps, 30 tbr, 1k tbn, 1k tbc (default)
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)