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• Invalid data found when processing file with ffmpeg API

  18 septembre 2019, par Timmy K

  I have an m4a file containing two AAC-encoded audio streams. I want to decode it on Android with the ffmpeg library. However, avformat_open_input() fails with "Invalid data found when processing input".

  This file was created (i.e. the audio data encoded) with the same library.

  I’ve double-checked the accuracy of the file path I provide to the call.

  The file doesn’t appear to be corrupt. It plays with QuickTime player, and ffprobe reports the output below.

  I’ve looked at the source in ffmpeg’s utils.c but it’s not clear at which of several points (including nested calls) this invalidity of the data in the file is being decided.

  FFPROBE OUTPUT
  
  built with gcc 7.2.0 (crosstool-NG fa8859cb)
  
  configuration: --prefix=/home/ubuntu/miniconda3 --disable-doc --enable- 
  
  shared --extra-cflags='-fPIC -I/home/ubuntu/miniconda3/include' --extra-
  
  cxxflags='=-fPIC' --extra-libs='-L/home/ubuntu/miniconda3/lib -lz' --enable-
  
  pic --disable-static --disable-gpl --disable-nonfree --disable-openssl --
  
  enable-libvpx --cc=/opt/conda/conda- 
  
  bld/ffmpeg_1530807717919/_build_env/bin/x86_64-conda_cos6-linux-gnu-cc --cxx=/opt/conda/conda-bld/ffmpeg_1530807717919/_build_env/bin/x86_64-conda_cos6-linux-gnu-c++ --enable-libopus
  
      libavutil            55. 78.100 / 55. 78.100
  
      libavcodec         57.107.100 / 57.107.100
  
      libavformat        57. 83.100 / 57. 83.100
  
      libavdevice        57. 10.100 / 57. 10.100
  
      libavfilter         6.107.100 /    6.107.100
  
      libswscale            4.    8.100 /    4.    8.100
  
      libswresample     2.    9.100 /    2.    9.100
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'recording-2019-09-03-110626.m4a':
  
      Metadata:
  
          major_brand         : isom
  
          minor_version     : 512
  
          compatible_brands: isomiso2mp41
  
          encoder                 : Lavf58.27.103
  
      Duration: 00:00:05.04, start: 0.000000, bitrate: 637 kb/s
  
          Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 314 kb/s (default)
  
          Metadata:
  
              handler_name        : SoundHandler
  
          Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 315 kb/s (default)
  
          Metadata:
  
              handler_name        : SoundHandler

  The call that fails with "invalid data found" is :

   if ( (ret = avformat_open_input( &inputFormatContext, filePath, 0, 0)) < 0 )
  
    {
  
      __android_log_print(ANDROID_LOG_DEBUG, "MyTag", "Could not open input file '%s' error %s ", filePath, av_err2str(ret) );
  
        return false;
  
    }

  I would expect this to "just work" since I encoded and muxed the two audio tracks with the same build of the ffmpeg libraries.

  When I built the libraries, I configured the build with :

  —enable-decoder=aac,pcm_s16le —enable-encoder=aac,pcm_s16le —enable-demuxer=mp4,wav —enable-muxer=mp4,wav —enable-protocol=file,http

  When I iterate through the codecs using av_codec_iterate() and look for supported sample rates I see :

  Found codec aac
  
  supported sample rate 96000
  
  supported sample rate 88200
  
  supported sample rate 64000
  
  supported sample rate 48000
  
  supported sample rate 44100
  
  supported sample rate 32000
  
  supported sample rate 24000
  
  supported sample rate 22050
  
  supported sample rate 16000
  
  supported sample rate 12000
  
  supported sample rate 11025
  
  supported sample rate 8000
  
  supported sample rate 7350
  
  Found codec pcm_s16le
  
  Found codec aac
  
  Found codec pcm_s16le 

  Each codec appears twice, which I thought was maybe the encoder and decoder. But I note that only one has any supported sample rates. My test files’ sample rate is 44100.

  Any thoughts on what other things I can check for ? I wondered whether I should create and pass an AVInputFormat object explicitly to the call, based upon what I know about the contents of the file. But I don’t know how to construct one.

• configure.ac : check for sys/auxv.h before defining FLAC__CPU_PPC

  19 septembre 2019, par Peter Seiderer

  configure.ac : check for sys/auxv.h before defining FLAC__CPU_PPC
  Signed-off-by : Peter Seiderer <ps.report@gmx.net>
  
  — -
  
  Note : One of the buildroot autobuilder provided powerpc toolchains
  
  seem to not provide the sys/auxv.h header file, resulting in a
  
  compile failure (see [1] for details).
  [1] http://lists.busybox.net/pipermail/buildroot/2019-September/259732.html
  

  • [DH] configure.ac

• FFMPEG concat audio and videos produces only audio

  19 septembre 2019, par justaguytrynabuildawebsite

  Im trying to combine multiple video files with a single audio file. When using only videos without audio, it works fine but when adding a audio track to the beginning of the concat file i get the return below, and the output.mp4 becomes just audio with no video when previewing in browser.
  FFMPEG command

  ffmpeg -f concat -safe 0 -i /var/www/html/video_tool/edit/out/concat.txt -c copy -flags +global_header /var/www/html/video_tool/edit/out/output.mp4

  concat.txt

  #CONCAT FILE
  
  file '/var/www/html/video_tool/edit/audio/sound.mp3'
  
  file '/var/www/html/video_tool/edit/vids/frame_0.mp4'
  
  file '/var/www/html/video_tool/edit/vids/frame_1.mp4'
  
  file '/var/www/html/video_tool/edit/vids/frame_2.mp4'
  
  file '/var/www/html/video_tool/edit/vids/frame_3.mp4'

  Output

  ffmpeg version 4.1.3-0york1~16.04 Copyright (c) 2000-2019 the FFmpeg developers
  
    built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.11) 20160609
  
    configuration: --prefix=/usr --extra-version='0york1~16.04' --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-avresample --disable-filter=resample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librsvg --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-nonfree --enable-libfdk-aac --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
  
    libavutil      56. 22.100 / 56. 22.100
  
    libavcodec     58. 35.100 / 58. 35.100
  
    libavformat    58. 20.100 / 58. 20.100
  
    libavdevice    58.  5.100 / 58.  5.100
  
    libavfilter     7. 40.101 /  7. 40.101
  
    libavresample   4.  0.  0 /  4.  0.  0
  
    libswscale      5.  3.100 /  5.  3.100
  
    libswresample   3.  3.100 /  3.  3.100
  
    libpostproc    55.  3.100 / 55.  3.100
  
  [mp3 @ 0x5635174e08c0] Estimating duration from bitrate, this may be inaccurate
  
  Input #0, concat, from '/var/www/html/video_tool/edit/out/concat.txt':
  
    Duration: N/A, start: 0.000000, bitrate: 192 kb/s
  
      Stream #0:0: Audio: mp3, 44100 Hz, stereo, fltp, 192 kb/s
  
  Output #0, mp4, to '/var/www/html/video_tool/edit/out/output.mp4':
  
    Metadata:
  
      encoder         : Lavf58.20.100
  
      Stream #0:0: Audio: mp3 (mp4a / 0x6134706D), 44100 Hz, stereo, fltp, 192 kb/s
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)
  
  Press [q] to stop, [?] for help
  
  [mov,mp4,m4a,3gp,3g2,mj2 @ 0x5635174e38c0] Auto-inserting h264_mp4toannexb bitstream filter
  
  [mp4 @ 0x5635174e5080] Non-monotonous DTS in output stream 0:0; previous: 9897984, current: 8976; changing to 9897985. This may result in incorrect timestamps in the output file.
  
  [mp4 @ 0x5635174e5080] Non-monotonous DTS in output stream 0:0; previous: 9897985, current: 8977; changing to 9897986. This may result in incorrect timestamps in the output file.
  
  [mp4 @ 0x5635174e5080] Non-monotonous DTS in output stream 0:0; previous: 9897986, current: 8979; changing to 9897987. This may result in incorrect timestamps in the output file.
  
  [mp4 @ 0x5635174e5080] Non-monotonous DTS in output stream 0:0; previous: 9897987, current: 8980; changing to 9897988. This may result in incorrect timestamps in the output file.
  
  etc, etc...
  
  size=    5347kB time=00:03:44.44 bitrate= 195.1kbits/s speed=2.91e+03x    
  
  video:0kB audio:5311kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.664769%