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• Java : How to merge and cut H.264 mp4 files with ffmpeg

  15 novembre 2016, par Nico

  I wrote an java application to handle my GoPro video data. Recording for a couple of hours the GoPro produces several chapters (mp4-files). Now the program is able to cut out a sequence no matter if it is in just one chapter or it touches more. I execute the outstanding tool ffmpeg.exe with the Processbuilder.
  First the program exports the mp4-files to *.ts-files and then it merges it togehter to one only mp4-file. At least a sequence will be cut out of this merged mp4-file using ffmpeg.exe for the second time. It works quite reliable and quickly.

  But : I do not have one application in its entirely, but also I have two files (jar-Application and ffmpeg.exe). I think there is are java-libraries of ffmpeg but I do not know how to include and how to use them. I am just searching for a one-file-solution.

  Do you know which library I need and where I can download it ? How may I have to use it ? If that is not possible, do you know some other user-friendly libraries doing the same job with less lines of code ? How to use them ?

  I am very grateful about your help and the examples you will give !

• 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

• FFmpeg : concatenation problems

  13 décembre 2016, par CryGuy

  I’d like to concat two mpeg files. Both have an MP2 audio stream that I select, but even though copying is intended, an AC3 stream is written.

  Input files :

  Input #0, mpeg, from 'D:\a.mpg':
  
    Duration: 01:25:54.05, start: 0.500000, bitrate: 4528 kb/s
  
      Stream #0:0[0x1bf]: Data: dvd_nav_packet
  
      Stream #0:1[0x1c0]: Audio: mp2, 48000 Hz, stereo, s16p, 192 kb/s
  
      Stream #0:2[0x80]: Audio: ac3, 48000 Hz, stereo, fltp, 384 kb/s
  
      Stream #0:3[0x1e0]: Video: mpeg2video (Main), yuv420p(tv, top first), 720x576 [SAR 64:45 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc
  
  Input #1, mpeg, from 'D:\b.mpg':
  
    Duration: 00:12:53.11, start: 0.500000, bitrate: 4486 kb/s
  
      Stream #1:0[0x1c0]: Audio: mp2, 48000 Hz, stereo, s16p, 192 kb/s
  
      Stream #1:1[0x80]: Audio: ac3, 48000 Hz, stereo, fltp, 384 kb/s
  
      Stream #1:2[0x1e0]: Video: mpeg2video (Main), yuv420p(tv, top first), 720x576 [SAR 64:45 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 

  ffmpeg args :

  -i "d:\a.mpg" -i "d:\b.mpg" -filter_complex "[0:3][0:1][1:2][1:0]concat=n=2:v=1:a=1[v][a]" -map [v] -map [a] -c copy -target pal-dvd "d:\c.mpg"

  As you can see, I selecting the video and the mp2 stream for both input files.

  Output :

  Stream mapping:
  
    Stream #0:1 (mp2) -> concat:in0:a0
  
    Stream #0:3 (mpeg2video) -> concat:in0:v0
  
    Stream #1:0 (mp2) -> concat:in1:a0
  
    Stream #1:2 (mpeg2video) -> concat:in1:v0
  
    concat:out:v0 -> Stream #0:0 (mpeg2video)
  
    concat:out:a0 -> Stream #0:1 (ac3)

  1. As you can see in the last line, the concat filter outputs an AC3 stream. Why ?
  2. The input is interlaced material. In the output file, the interlaced flag is not set correctly so that it’s played back interlaced.

  I’ve been trying to solve this for a couple of hours now, with and without different mapping options.

  Thanks for reading.