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• What is the best way to merge .mkv and .mka files using ffmpeg ?

  28 juin 2017, par Robert

  I’m using ffmpeg to merge .mkv and .mka files into .mp4 files. My current command looks like this :

  ffmpeg -i video.mkv -i audio.mka output_path.mp4

  The audio and video files are pre-signed urls from Amazon S3. Even on a server with sufficient resources, this process is going very slowly. I’ve researched situations where you can tell ffmpeg to skip re-encoding each frame, but I think that in my situation it actually does need to re-encode each frame.

  I’ve downloaded 2 sample files to my macbook pro and have installed ffmpeg locally via homebrew. When I run the command

  ffmpeg -i video.mkv -i audio.mka -c copy output.mp4

  I get the following output :

  ffmpeg version 3.3.2 Copyright (c) 2000-2017 the FFmpeg developers
  
    built with Apple LLVM version 8.1.0 (clang-802.0.42)
  
    configuration: --prefix=/usr/local/Cellar/ffmpeg/3.3.2 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-libmp3lame --enable-libx264 --enable-libxvid --enable-opencl --disable-lzma --enable-vda
  
    libavutil      55. 58.100 / 55. 58.100
  
    libavcodec     57. 89.100 / 57. 89.100
  
    libavformat    57. 71.100 / 57. 71.100
  
    libavdevice    57.  6.100 / 57.  6.100
  
    libavfilter     6. 82.100 /  6. 82.100
  
    libavresample   3.  5.  0 /  3.  5.  0
  
    libswscale      4.  6.100 /  4.  6.100
  
    libswresample   2.  7.100 /  2.  7.100
  
    libpostproc    54.  5.100 / 54.  5.100
  
  Input #0, matroska,webm, from '319_audio_1498590673766.mka':
  
    Metadata:
  
      encoder         : GStreamer matroskamux version 1.8.1.1
  
      creation_time   : 2017-06-27T19:10:58.000000Z
  
    Duration: 00:00:03.53, start: 2.831000, bitrate: 50 kb/s
  
      Stream #0:0(eng): Audio: opus, 48000 Hz, stereo, fltp (default)
  
      Metadata:
  
        title           : Audio
  
  Input #1, matroska,webm, from '319_video_1498590673766.mkv':
  
    Metadata:
  
      encoder         : GStreamer matroskamux version 1.8.1.1
  
      creation_time   : 2017-06-27T19:10:58.000000Z
  
    Duration: 00:00:03.97, start: 2.851000, bitrate: 224 kb/s
  
      Stream #1:0(eng): Video: vp8, yuv420p(progressive), 640x480, SAR 1:1 DAR 4:3, 30 tbr, 1k tbn, 1k tbc (default)
  
      Metadata:
  
        title           : Video
  
  [mp4 @ 0x7fa4f0806800] Could not find tag for codec vp8 in stream #0, codec not currently supported in container
  
  Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument
  
  Stream mapping:
  
    Stream #1:0 -> #0:0 (copy)
  
    Stream #0:0 -> #0:1 (copy)
  
      Last message repeated 1 times

  So it appears that the specific encodings I’m working with are vp8 videos and opus audio files, which I believe are incompatible with the .mp4 output container. I would appreciate answers that cover ways of optimally merging vp8 and opus into .mp4 output or answers that point me in the direction of output media formats that are both compatible with vp8 & opus and are playable on web and mobile devices so that I can bypass the re-encoding step altogether.

  EDIT :

  Just wanted to provide a benchmark after following LordNeckbeard’s advice :

  4 min 41 second video transcoded locally on my mac
  
  
  
  LordNeckbeard’s approach : 15 mins 55 seconds (955 seconds)
  
  Current approach : 18 mins 49 seconds (1129 seconds)
  
  
  
  18% speed increase

• Revision 31020 : max = 255, ça bloquait la création de la table sur certaine base

  20 août 2009, par vincent@… — Log

  max = 255, ça bloquait la création de la table sur certaine base

• WebRTC books – a brief review

  1er janvier 2014, par silvia

  I just finished reading Rob Manson’s awesome book “Getting Started with WebRTC” and I can highly recommend it for any Web developer who is interested in WebRTC.

  Rob explains very clearly how to create your first video, audio or data peer-connection using WebRTC in current Google Chrome or Firefox (I think it also now applies to Opera, though that wasn’t the case when his book was published). He makes available example code, so you can replicate it in your own Web application easily, including the setup of a signalling server. He also points out that you need a ICE (STUN/TURN) server to punch through firewalls and gives recommendations for what software is available, but stops short of explaining how to set them up.

  Rob’s focus is very much on the features required in a typical Web application :

  • video calls
  • audio calls
  • text chats
  • file sharing

  In fact, he provides the most in-depth demo of how to set up a good file sharing interface I have come across.

  Rob then also extends his introduction to WebRTC to two key application areas : education and team communication. His recommendations are spot on and required reading for anyone developing applications in these spaces.

  —

  Before Rob’s book, I have also read Alan Johnson and Dan Burnett’s “WebRTC” book on APIs and RTCWEB protocols of the HTML5 Real-Time Web.

  Alan and Dan’s book was written more than a year ago and explains that state of standardisation at that time. It’s probably a little out-dated now, but it still gives you good foundations on why some decisions were made the way they are and what are contentious issues (some of which still remain). If you really want to understand what happens behind the scenes when you call certain functions in the WebRTC APIs of browsers, then this is for you.

  Alan and Dan’s book explains in more details than Rob’s book how IP addresses of communication partners are found, how firewall holepunching works, how sessions get negotiated, and how the standards process works. It’s probably less useful to a Web developer who just wants to implement video call functionality into their Web application, though if something goes wrong you may find yourself digging into the details of SDP, SRTP, DTLS, and other cryptic abbreviations of protocols that all need to work together to get a WebRTC call working.

  —

  Overall, both books are worthwhile and cover different aspects of WebRTC that you will stumble across if you are directly dealing with WebRTC code.