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• Chrome, ffmpeg, and ogg

  13 novembre 2016, par porton

  We need to convert mp4 into ogg by ffmpeg (or maybe some other converter) in such a way that the ogg file to be compatible with Chrome browser.

  The command like

  ffmpeg -i x.mp4 x.ogg

  produces an ogg file which does not work in Chrome.

  ffmpeg -i video_mp4_01_vlublennaya_mama.mp4 x.ogg 2>&1 | tee log.txt

  produces

  ffmpeg version 3.2-2 Copyright (c) 2000-2016 the FFmpeg developers
  
    built with gcc 6.2.0 (Debian 6.2.0-10) 20161027
  
    configuration: --prefix=/usr --extra-version=2 --toolchain=hardened --libdir=/usr/lib/i386-linux-gnu --incdir=/usr/include/i386-linux-gnu --enable-gpl --disable-libtesseract --disable-stripping --disable-decoder=libschroedinger --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libebur128 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmodplug --enable-libmp3lame --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-librubberband --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-libzmq --enable-libzvbi --enable-opengl --enable-sdl2 --enable-x11grab --enable-libdc1394 --enable-libiec61883 --enable-openal --enable-frei0r --enable-libopencv --enable-libx264 --enable-chromaprint --enable-shared
  
    libavutil      55. 34.100 / 55. 34.100
  
    libavcodec     57. 64.100 / 57. 64.100
  
    libavformat    57. 56.100 / 57. 56.100
  
    libavdevice    57.  1.100 / 57.  1.100
  
    libavfilter     6. 65.100 /  6. 65.100
  
    libavresample   3.  1.  0 /  3.  1.  0
  
    libswscale      4.  2.100 /  4.  2.100
  
    libswresample   2.  3.100 /  2.  3.100
  
    libpostproc    54.  1.100 / 54.  1.100
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'video_mp4_01_vlublennaya_mama.mp4':
  
    Metadata:
  
      major_brand     : isom
  
      minor_version   : 512
  
      compatible_brands: isomiso2avc1mp41
  
      encoder         : Lavf54.63.104
  
    Duration: 00:01:23.87, start: 0.000000, bitrate: 840 kb/s
  
      Stream #0:0(und): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 304x224 [SAR 1:1 DAR 19:14], 762 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 tbc (default)
  
      Metadata:
  
        handler_name    : VideoHandler
  
      Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 22050 Hz, stereo, fltp, 74 kb/s (default)
  
      Metadata:
  
        handler_name    : SoundHandler
  
  Output #0, ogg, to 'x.ogg':
  
    Metadata:
  
      major_brand     : isom
  
      minor_version   : 512
  
      compatible_brands: isomiso2avc1mp41
  
      encoder         : Lavf57.56.100
  
      Stream #0:0(und): Video: theora (libtheora), yuv420p, 304x224 [SAR 1:1 DAR 19:14], q=2-31, 200 kb/s, 25 fps, 25 tbn, 25 tbc (default)
  
      Metadata:
  
        handler_name    : VideoHandler
  
        encoder         : Lavc57.64.100 libtheora
  
        major_brand     : isom
  
        minor_version   : 512
  
        compatible_brands: isomiso2avc1mp41
  
      Stream #0:1(und): Audio: vorbis (libvorbis), 22050 Hz, stereo, fltp (default)
  
      Metadata:
  
        handler_name    : SoundHandler
  
        encoder         : Lavc57.64.100 libvorbis
  
        major_brand     : isom
  
        minor_version   : 512
  
        compatible_brands: isomiso2avc1mp41
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (h264 (native) -> theora (libtheora))
  
    Stream #0:1 -> #0:1 (aac (native) -> vorbis (libvorbis))
  
  Press [q] to stop, [?] for help
  
  frame=  117 fps=0.0 q=-0.0 size=     124kB time=00:00:04.96 bitrate= 205.4kbits/s speed=9.91x    
  
  frame=  247 fps=247 q=-0.0 size=     291kB time=00:00:10.12 bitrate= 235.9kbits/s speed=10.1x    
  
  frame=  376 fps=250 q=-0.0 size=     438kB time=00:00:15.28 bitrate= 234.9kbits/s speed=10.2x    
  
  frame=  501 fps=250 q=-0.0 size=     605kB time=00:00:20.30 bitrate= 244.0kbits/s speed=10.1x    
  
  frame=  635 fps=253 q=-0.0 size=     756kB time=00:00:25.58 bitrate= 241.9kbits/s speed=10.2x    
  
  frame=  762 fps=253 q=-0.0 size=     901kB time=00:00:30.74 bitrate= 240.0kbits/s speed=10.2x    
  
  frame=  895 fps=255 q=-0.0 size=    1087kB time=00:00:35.99 bitrate= 247.3kbits/s speed=10.3x    
  
  frame= 1018 fps=254 q=-0.0 size=    1219kB time=00:00:40.92 bitrate= 244.0kbits/s speed=10.2x    
  
  frame= 1149 fps=255 q=-0.0 size=    1367kB time=00:00:46.21 bitrate= 242.3kbits/s speed=10.2x    
  
  frame= 1275 fps=254 q=-0.0 size=    1527kB time=00:00:51.26 bitrate= 244.0kbits/s speed=10.2x    
  
  frame= 1407 fps=255 q=-0.0 size=    1709kB time=00:00:56.47 bitrate= 247.9kbits/s speed=10.2x    
  
  frame= 1533 fps=255 q=-0.0 size=    1865kB time=00:01:01.58 bitrate= 248.0kbits/s speed=10.2x    
  
  frame= 1667 fps=256 q=-0.0 size=    2018kB time=00:01:06.88 bitrate= 247.2kbits/s speed=10.3x    
  
  frame= 1798 fps=256 q=-0.0 size=    2171kB time=00:01:12.12 bitrate= 246.6kbits/s speed=10.3x    
  
  frame= 1923 fps=256 q=-0.0 size=    2334kB time=00:01:17.13 bitrate= 247.9kbits/s speed=10.3x    
  
  [h264 @ 0x81dd77e0] error while decoding MB 8 10, bytestream -13
  
  [h264 @ 0x81dd77e0] concealing 117 DC, 117 AC, 117 MV errors in P frame
  
  [aac @ 0x81dbc020] channel element 2.6 is not allocated
  
  Error while decoding stream #0:1: Invalid data found when processing input
  
  [h264 @ 0x81e11840] Invalid nal size -352448305
  
  [h264 @ 0x81e11840] Error splitting the input into NAL units.
  
  [aac @ 0x81dbc020] Sample rate index in program config element does not match the sample rate index configured by the container.
  
  [aac @ 0x81dbc020] Inconsistent channel configuration.
  
  [aac @ 0x81dbc020] get_buffer() failed
  
  Error while decoding stream #0:1: Invalid argument
  
  [h264 @ 0x81e55860] Invalid nal size 127321414
  
  [h264 @ 0x81e55860] Error splitting the input into NAL units.
  
  Error while decoding stream #0:0: Invalid data found when processing input
  
      Last message repeated 1 times
  
  frame= 2045 fps=255 q=-0.0 size=    2483kB time=00:01:22.05 bitrate= 247.9kbits/s dup=2 drop=0 speed=10.2x    
  
  frame= 2096 fps=255 q=-0.0 Lsize=    2549kB time=00:01:23.86 bitrate= 249.0kbits/s dup=2 drop=0 speed=10.2x    
  
  video:2046kB audio:472kB subtitle:0kB other streams:0kB global headers:7kB muxing overhead: 1.239568%

• Playing Live Audio Stream in HTML5 - MediaSource Errors in Chrome

  18 octobre 2016, par user882807

  I need a way to play a live audio stream using HTML5 (primarily in Google Chrome), so I tried using the following :

  <audio>
  
      <source src="my-live-stream.ogg" type="audio/ogg">
  
  </source></audio>

  While this does work for a live stream, there seems to be a very large, uncontrollable delay/buffer of around 30 seconds or so when this is played.

  I need the delay to be a couple of seconds or less so this method doesn’t work.

  As an alternative I have tried sending the audio over a WebSocket connection as 1 second individual audio files, which are then appended to a SourceBuffer and played in an audio element using Media Source Extensions.

  After experimenting with a number of formats (MediaSource.isTypeSupported seems to be rather limited in audio support), I got this working using a Vorbis audio stream in a WebM container, which sounds perfect with no audible gaps. Other formats worked less well as they need to be gapless - e.g. MP3 and AAC end up with tiny audible gaps between each 1 second segment.

  While this seems to work at first, when looking at chrome://media-internals, the following errors repeatedly appear :

  00:00:09 544    info    Estimating WebM block duration to be 3ms for the last (Simple)Block in the Cluster for this Track. Use BlockGroups with BlockDurations at the end of each Track in a Cluster to avoid estimation.
  
  00:00:09 585    error   Large timestamp gap detected; may cause AV sync to drift. time:8994999us expected:9231000us delta:-236001us
  
  00:01:05 239    debug   Skipping splice frame generation: not enough samples for splicing new buffer at 65077997us. Have 1us, but need 1000us.

  Eventually the playback stops as though the pause button has been pressed on the audio element. It still shows the pause rather than play button, but the current time stops advancing :

  

  Pressing the pause button and then the play button that replaces it doesn’t make it start playing again, but manually dragging the position slider further ahead makes it continue playing.

  I have tried setting sourceBuffer.mode = 'sequence'; but this doesn’t seem to help.

  Is there anything that needs to be changed in how the audio files are being encoded, or how they are played back in JavaScript to fix this ?

  ---

  Additional details :

  1. The audio stream is encoded into 1 second WebM/Vorbis files using FFmpeg on Windows.
  2. A background worker is used in the browser to receive the audio segments and pass them to the main page thread, which appends them to the audio stream. Otherwise the playback freezes.

  Source code :

  • Web player : https://github.com/SamuelFisher/WebSocketAudio
  • WebSocket server and encoder : https://github.com/SamuelFisher/WebSocketAudioServer

• Audio recorded with MediaRecorder on Chrome missing duration

  27 octobre 2016, par suppp111

  I am recording audio (oga/vorbis) files with MediaRecorder. When I record these file through Chrome I get problems : I cannot edit the files on ffmpeg and when I try to play them on Firefox it says they are corrupt (they do play fine on Chrome though).

  Looking at their metadata on ffmpeg I get this :

  Input #0, matroska,webm, from '91.oga':
  
    Metadata:
  
      encoder         : Chrome
  
    Duration: N/A, start: 0.000000, bitrate: N/A
  
      Stream #0:0(eng): Audio: opus, 48000 Hz, mono, fltp (default)
  
  [STREAM]
  
  index=0
  
  codec_name=opus
  
  codec_long_name=Opus (Opus Interactive Audio Codec)
  
  profile=unknown
  
  codec_type=audio
  
  codec_time_base=1/48000
  
  codec_tag_string=[0][0][0][0]
  
  codec_tag=0x0000
  
  sample_fmt=fltp
  
  sample_rate=48000
  
  channels=1
  
  channel_layout=mono
  
  bits_per_sample=0
  
  id=N/A
  
  r_frame_rate=0/0
  
  avg_frame_rate=0/0
  
  time_base=1/1000
  
  start_pts=0
  
  start_time=0.000000
  
  duration_ts=N/A
  
  duration=N/A
  
  bit_rate=N/A
  
  max_bit_rate=N/A
  
  bits_per_raw_sample=N/A
  
  nb_frames=N/A
  
  nb_read_frames=N/A
  
  nb_read_packets=N/A
  
  DISPOSITION:default=1
  
  DISPOSITION:dub=0
  
  DISPOSITION:original=0
  
  DISPOSITION:comment=0
  
  DISPOSITION:lyrics=0
  
  DISPOSITION:karaoke=0
  
  DISPOSITION:forced=0
  
  DISPOSITION:hearing_impaired=0
  
  DISPOSITION:visual_impaired=0
  
  DISPOSITION:clean_effects=0
  
  DISPOSITION:attached_pic=0
  
  TAG:language=eng
  
  [/STREAM]
  
  [FORMAT]
  
  filename=91.oga
  
  nb_streams=1
  
  nb_programs=0
  
  format_name=matroska,webm
  
  format_long_name=Matroska / WebM
  
  start_time=0.000000
  
  duration=N/A
  
  size=7195
  
  bit_rate=N/A
  
  probe_score=100
  
  TAG:encoder=Chrome

  As you can see there are problems with the duration. I have looked at posts like this :
  How can I add predefined length to audio recorded from MediaRecorder in Chrome ?

  But even trying that, I got errors when trying to chop and merge files.For example when running :

  ffmpeg -f concat  -i 89_inputs.txt -c copy final.oga

  I get a lot of this :

  [oga @ 00000000006789c0] Non-monotonous DTS in output stream 0:0; previous: 57612, current: 1980; changing to 57613. This may result in incorrect timestamps in the output file.
  
  [oga @ 00000000006789c0] Non-monotonous DTS in output stream 0:0; previous: 57613, current: 2041; changing to 57614. This may result in incorrect timestamps in the output file.
  
  DTS -442721849179034176, next:42521 st:0 invalid dropping
  
  PTS -442721849179034176, next:42521 invalid dropping st:0
  
  [oga @ 00000000006789c0] Non-monotonous DTS in output stream 0:0; previous: 57614, current: 2041; changing to 57615. This may result in incorrect timestamps in the output file.
  
  [oga @ 00000000006789c0] Timestamps are unset in a packet for stream 0. This is deprecated and will stop working in the future. Fix your code to set the timestamps properly
  
  DTS -442721849179031296, next:42521 st:0 invalid dropping
  
  PTS -442721849179031296, next:42521 invalid dropping st:0

  Does anyone know what we need to do to audio files recorded from Chrome for them to be useful ? Or is there a problem with my setup ?

  Recorder js :

  if (navigator.getUserMedia) {
  
    console.log('getUserMedia supported.');
  
  
  
    var constraints = { audio: true };
  
    var chunks = [];
  
  
  
    var onSuccess = function(stream) {
  
      var mediaRecorder = new MediaRecorder(stream);
  
  
  
      record.onclick = function() {
  
        mediaRecorder.start();
  
        console.log(mediaRecorder.state);
  
        console.log("recorder started");
  
        record.style.background = "red";
  
  
  
        stop.disabled = false;
  
        record.disabled = true;
  
  
  
        var aud = document.getElementById("audioClip");
  
        start = aud.currentTime;
  
      }
  
  
  
      stop.onclick = function() {
  
        console.log(mediaRecorder.state);
  
        console.log("Recording request sent.");
  
        mediaRecorder.stop();
  
      }
  
  
  
      mediaRecorder.onstop = function(e) {
  
        console.log("data available after MediaRecorder.stop() called.");
  
  
  
        var audio = document.createElement('audio');
  
        audio.setAttribute('controls', '');
  
        audio.setAttribute('id', 'audioClip');
  
  
  
        audio.controls = true;
  
        var blob = new Blob(chunks, { 'type' : 'audio/ogg; codecs="vorbis"' });
  
        chunks = [];
  
        var audioURL = window.URL.createObjectURL(blob);
  
        audio.src = audioURL;
  
  
  
        sendRecToPost(blob);   // this just send the audio blob to the server by post
  
        console.log("recorder stopped");
  
  
  
      }