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• FFMPEG installed but not found

  20 mai 2016, par justHelloWorld

  I’m trying to install pHash , but when I execute ./configure this message is returned :

  ...
  
  
  
  *** Configuring video Hash ***
  
  
  
  checking whether FFmpeg is present... checking for avcodec_alloc_frame in -lavcodec... no
  
  configure: error: 
  
  
  
  *** libavcodec not found.
  
  You need FFmpeg. Get it at /ffmpeg.org/>

  But FFMPEG is correctly installed, in fact executing ffmpeg it returns :

  ffmpeg version 3.0.2 Copyright (c) 2000-2016 the FFmpeg developers
  
    built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.1)
  
    configuration: 
  
    libavutil      55. 17.103 / 55. 17.103
  
    libavcodec     57. 24.102 / 57. 24.102
  
    libavformat    57. 25.100 / 57. 25.100
  
    libavdevice    57.  0.101 / 57.  0.101
  
    libavfilter     6. 31.100 /  6. 31.100
  
    libswscale      4.  0.100 /  4.  0.100
  
    libswresample   2.  0.101 /  2.  0.101
  
  Hyper fast Audio and Video encoder
  
  usage: ffmpeg [options] [[infile options] -i infile]... {[outfile options] outfile}...
  
  
  
  Use -h to get full help or, even better, run 'man ffmpeg'

  Why this happens ?

• How to call ffmpeg code to keep same video quality ?

  7 mai 2016, par seaguest

  I am using ffmpeg3 to convert mv.webm to mp4, it works fine when I use the command line.

  $ ffprobe mv.webm 
  
  ffprobe version N-79789-g58b3e56 Copyright (c) 2007-2016 the FFmpeg developers
  
    built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.1)
  
    configuration: --enable-gpl --enable-shared --enable-libx264
  
    libavutil      55. 24.100 / 55. 24.100
  
    libavcodec     57. 39.100 / 57. 39.100
  
    libavformat    57. 36.100 / 57. 36.100
  
    libavdevice    57.  0.101 / 57.  0.101
  
    libavfilter     6. 45.100 /  6. 45.100
  
    libswscale      4.  1.100 /  4.  1.100
  
    libswresample   2.  0.101 /  2.  0.101
  
    libpostproc    54.  0.100 / 54.  0.100
  
  Input #0, matroska,webm, from 'mv.webm':
  
    Metadata:
  
      encoder         : google
  
    Duration: 00:04:56.36, start: 0.000000, bitrate: 836 kb/s
  
      Stream #0:0: Video: vp8, yuv420p, 640x480, SAR 1:1 DAR 4:3, 25 fps, 25 tbr, 1k tbn (default)
  
      Stream #0:1: Audio: vorbis, 44100 Hz, stereo, fltp (default)
  
  
  
  $ ffmpeg -i mv.webm  -acodec aac -vcodec libx264  mv1.mp4
  
  $ ffprobe mv1.mp4 
  
  ffprobe version N-79789-g58b3e56 Copyright (c) 2007-2016 the FFmpeg developers
  
    built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.1)
  
    configuration: --enable-gpl --enable-shared --enable-libx264
  
    libavutil      55. 24.100 / 55. 24.100
  
    libavcodec     57. 39.100 / 57. 39.100
  
    libavformat    57. 36.100 / 57. 36.100
  
    libavdevice    57.  0.101 / 57.  0.101
  
    libavfilter     6. 45.100 /  6. 45.100
  
    libswscale      4.  1.100 /  4.  1.100
  
    libswresample   2.  0.101 /  2.  0.101
  
    libpostproc    54.  0.100 / 54.  0.100
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'n1.mp4':
  
    Metadata:
  
      major_brand     : isom
  
      minor_version   : 512
  
      compatible_brands: isomiso2avc1mp41
  
      encoder         : Lavf57.36.100
  
    Duration: 00:04:56.36, start: 0.000000, bitrate: 628 kb/s
  
      Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 640x480 [SAR 1:1 DAR 4:3], 500 kb/s, 25 fps, 25 tbr, 12800 tbn (default)
  
      Metadata:
  
        handler_name    : VideoHandler
  
      Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 120 kb/s (default)
  
      Metadata:
  
        handler_name    : SoundHandler

  I play it with VLC, there is no noise at all.

  Here is my first question, in case we don’t specify the bitrate in the ffmpeg command, how are the bitrate of video/audio determined ?

  Since I am developping in Golang, I use the framework gmf which calls ffmpeg, I use go to convert the video format without specifying bitrate, then I got the following info :

  $ ffprobe mv2.mp4 
  
  ffprobe version N-79789-g58b3e56 Copyright (c) 2007-2016 the FFmpeg developers
  
    built with gcc 4.8 (Ubuntu 4.8.4-2ubuntu1~14.04.1)
  
    configuration: --enable-gpl --enable-shared --enable-libx264
  
    libavutil      55. 24.100 / 55. 24.100
  
    libavcodec     57. 39.100 / 57. 39.100
  
    libavformat    57. 36.100 / 57. 36.100
  
    libavdevice    57.  0.101 / 57.  0.101
  
    libavfilter     6. 45.100 /  6. 45.100
  
    libswscale      4.  1.100 /  4.  1.100
  
    libswresample   2.  0.101 /  2.  0.101
  
    libpostproc    54.  0.100 / 54.  0.100
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'mv2.mp4':
  
    Metadata:
  
      major_brand     : isom
  
      minor_version   : 512
  
      compatible_brands: isomiso2avc1mp41
  
      encoder         : Lavf57.36.100
  
    Duration: 00:04:56.36, start: 0.000000, bitrate: 540 kb/s
  
      Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 640x480, 385 kb/s, 25 fps, 25 tbr, 12800 tbn (default)
  
      Metadata:
  
        handler_name    : VideoHandler
  
      Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 147 kb/s (default)
  
      Metadata:
  
        handler_name    : SoundHandler

  The bitrate is event better than mv1.mp4, however when I play it with VLC, I heard a lot of noises.

  I tried to increase the audio bitrate (the way we set bitrate is like AVCodecContext->bit_rate = 441000...), but this doesn’t help.

  Maybe there is some option that I need to add when calling ffmpeg code to keep the same quality as input source, could anyone help ?

• ffmpeg avcodec_open2 changed audio format in AVCodecContext

  16 mai 2016, par user1083577

  I have a mkv file with a DTS track.

  After avcodec_open2 is called, sample_fmt in AVCodexContext changed. Is it expected ?

  AVCodecContext* avAudioCodecCtx = avFormatCtx->streams[audioStreamIndex]->codec;
  
  
  
  AVSampleFormat f = avAudioCodecCtx->sample_fmt; // AV_SAMPLE_FMT_FLTP
  
  avcodec_open2(avAudioCodecCtx, avAudioCodec, NULL);
  
  AVSampleFormat f2 = avAudioCodecCtx->sample_fmt; // AV_SAMPLE_FMT_S32P

  ffprobe

  ffprobe version N-79143-g8ff0f6a Copyright (c) 2007-2016 the FFmpeg developers
  
    built with gcc 5.3.0 (GCC)
  
    configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libdcadec --enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmfx --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --enable-zlib
  
    libavutil      55. 19.100 / 55. 19.100
  
    libavcodec     57. 30.100 / 57. 30.100
  
    libavformat    57. 29.101 / 57. 29.101
  
    libavdevice    57.  0.101 / 57.  0.101
  
    libavfilter     6. 40.102 /  6. 40.102
  
    libswscale      4.  0.100 /  4.  0.100
  
    libswresample   2.  0.101 /  2.  0.101
  
    libpostproc    54.  0.100 / 54.  0.100
  
  
  
  Stream #0:1(eng): Audio: dts (DTS), 48000 Hz, 5.1(side), fltp, 768 kb/s (default)

  Decoding

  avcodec_decode_audio4(avAudioCodecCtx, frame, &frameComplete, avPacket);
  
  frame->format // contains AV_SAMPLE_FMT_FLTP