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• Deploying QT project with opencv and ffmpeg dylib error

  9 juillet 2019, par Clarky

  When I build my project using the opencv and ffmpeg dylibs my application crashes upon opining with the error :

  Dyld Error Message :
  Library not loaded :
  /usr/local/Cellar/ffmpeg/4.1.3/lib/libswresample.3.dylib

  Referenced from :
  /Applications/myApp.app/Contents/Frameworks/libavcodec.58.dylib
  Reason : image not found

  I have tried using the install_name_tool to try and change the paths using this :

  install_name_tool -change libavcodec.58.dylib 
  
  
  
  @executable_path/../Frameworks/libavcodec.58.dylib
  
  
  
  /Users/me/code/build_output/MyAppQML/myApp.app/Contents/MacOS/myApp

  Although this modified the file I was still getting the same error and crash at run time.

  Is install_name_tool suitable to fix this issue or should I be looking at something else ?

  I’ve used otool on libavcodec.58.dylib which showed that some of the paths where linked to the @executable_path

  @executable_path/../Frameworks/libavcodec.58.dylib (compatibility version 58.0.0, current version 58.35.100)
  
      /usr/local/Cellar/ffmpeg/4.1.3/lib/libswresample.3.dylib (compatibility version 3.0.0, current version 3.3.100)
  
      /usr/local/Cellar/ffmpeg/4.1.3/lib/libavutil.56.dylib (compatibility version 56.0.0, current version 56.22.100)
  
      /usr/lib/libSystem.B.dylib (compatibility version 1.0.0, current version 1238.60.2)
  
      /usr/lib/libiconv.2.dylib (compatibility version 7.0.0, current version 7.0.0)
  
      @executable_path/../Frameworks/liblzma.5.dylib (compatibility version 8.0.0, current version 8.4.0)
  
      @executable_path/../Frameworks/libopencore-amrwb.0.dylib (compatibility version 1.0.0, current version 1.3.0)

• Get lengths of intermediate concatenated files in ffmpeg filter-complex

  17 juillet 2019, par Matthew

  I’m writing a media Electron app that occasionally needs to individually trim => individually normalize => concatenate => convert a varying number of WAV files into MP3.
  I’ve successfully used FFMPEG (via Fluent-ffmpeg) to do so (command wrapped for visibility) :

  ffmpeg -i 3.301_to_8.752_Careful.wav -i 8.752_to_18.751_Careful.wav -y 
  
  -filter_complex 
  
  [0]silenceremove=start_periods=1:start_threshold=-50dB[mid];[mid]loudnorm=I=-16:TP=-1.5:LRA=11[out];
  
  [1]silenceremove=start_periods=1:start_threshold=-50dB[mid];[mid]loudnorm=I=-16:TP=-1.5:LRA=11[b];
  
  [out][b]concat=v=0:a=1[out]
  
   -b:a 128k -ac 1 -acodec libmp3lame -f mp3 -map [out] -y Careful_Merged.mp3

  Here’s the relevant parts of the output :

  Guessed Channel Layout for Input Stream #0.0 : mono
  
  Input #0, wav, from '3.301_to_8.752_Careful.wav':
  
    Duration: 00:00:05.50, bitrate: 705 kb/s
  
      Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono, s16, 705 kb/s
  
  Guessed Channel Layout for Input Stream #1.0 : mono
  
  Input #1, wav, from '8.752_to_18.751_Careful.wav':
  
    Duration: 00:00:10.30, bitrate: 705 kb/s
  
      Stream #1:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono, s16, 705 kb/s
  
  Stream mapping:
  
    Stream #0:0 (pcm_s16le) -> silenceremove
  
    Stream #1:0 (pcm_s16le) -> silenceremove
  
    concat -> Stream #0:0 (libmp3lame)
  
  Press [q] to stop, [?] for help
  
  Output #0, mp3, to 'Careful_Merged.mp3':
  
    Metadata:
  
      TSSE            : Lavf58.28.101
  
      Stream #0:0: Audio: mp3 (libmp3lame), 48000 Hz, mono, fltp, 128 kb/s (default)
  
      Metadata:
  
        encoder         : Lavc58.53.101 libmp3lame
  
  size=     246kB time=00:00:15.69 bitrate= 128.4kbits/s speed=33.8x
  
  video:0kB audio:246kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.170563%

  The benefit is that the process chains nicely, but the downside is that I don’t know the resulting length of the intermediate files after the automagic trim. My app needs a "table of contents" showing the start and end of each segment, but I can’t figure out how to export the duration of each trimmed file before concatenation between
  [0]silenceremove=start_periods=1:start_threshold=-50dB[mid];[mid]loudnorm=I=-16:TP=-1.5:LRA=11[out] and the concatenation [out][b]concat=v=0:a=1[out].

  It’s so elegant as it is, I’d prefer not to save the intermediate files to disk or trim the audio twice to get the info I need. All I need is a duration for each, and I can do the math.
  Is there a filter that I can put inline to export the intermediate duration, or a way to get a log of the concatenation action ?

  Any ideas ?

• FFmpeg from Python has Corrupted Output

  14 juillet 2019, par Mazyod

  In a python script, I have :

  os.system(ff_cmd)
  
  # also tried Popen(ff_cmd, shell=True).wait()

  Then, I ran that same ff_cmd from terminal directly. Here are the results :

  # ff_cmd
  
  ffmpeg -i "114006.mp3" -acodec pcm_s16le -ar 16000 -ac 1 "114-006-a4dec52a.wav"
  
  
  
  # after running from python
  
  % file 114-006-a4dec52a.wav
  
  114-006-a4dec52a.wav: Audio file with ID3 version 2.4.0, contains:MPEG ADTS, layer III, v2,  40 kbps, 16 kHz, Monaural
  
  
  
  # after running from terminal
  
  % file 114-006-a4dec52a.wav
  
  114-006-a4dec52a.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 16000 Hz

  Can anyone please explain why in the world would they result in different outputs ?

  ---

  Output from Python :

  ffmpeg version 4.1.4 Copyright (c) 2000-2019 the FFmpeg developers
  
    built with Apple LLVM version 10.0.1 (clang-1001.0.46.4)
  
    configuration: --prefix=/usr/local/Cellar/ffmpeg/4.1.4 --enable-shared --enable-pthreads --enable-version3 --enable-avresample --cc=clang --host-cflags='-I/Library/Java/JavaVirtualMachines/adoptopenjdk-12.0.1.jdk/Contents/Home/include -I/Library/Java/JavaVirtualMachines/adoptopenjdk-12.0.1.jdk/Contents/Home/include/darwin' --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libmp3lame --enable-libopus --enable-librubberband --enable-libsnappy --enable-libtesseract --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libspeex --enable-videotoolbox --disable-libjack --disable-indev=jack --enable-libaom --enable-libsoxr
  
    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 @ 0x7ffa34001800] Estimating duration from bitrate, this may be inaccurate
  
  Input #0, mp3, from '/.../114006.mp3':
  
    Duration: 00:00:09.53, start: 0.000000, bitrate: 64 kb/s
  
      Stream #0:0: Audio: mp3, 22050 Hz, stereo, fltp, 64 kb/s
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (mp3 (mp3float) -> pcm_s16le (native))
  
  Press [q] to stop, [?] for help
  
  Output #0, wav, to '/.../114-006-a4dec52a.wav':
  
    Metadata:
  
      ISFT            : Lavf58.20.100
  
      Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 16000 Hz, mono, s16, 256 kb/s
  
      Metadata:
  
        encoder         : Lavc58.35.100 pcm_s16le
  
  size=     298kB time=00:00:09.53 bitrate= 256.1kbits/s speed= 166x    
  
  video:0kB audio:298kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.025564%

  Output from terminal :

  ffmpeg version 4.1.4 Copyright (c) 2000-2019 the FFmpeg developers
  
    built with Apple LLVM version 10.0.1 (clang-1001.0.46.4)
  
    configuration: --prefix=/usr/local/Cellar/ffmpeg/4.1.4 --enable-shared --enable-pthreads --enable-version3 --enable-avresample --cc=clang --host-cflags='-I/Library/Java/JavaVirtualMachines/adoptopenjdk-12.0.1.jdk/Contents/Home/include -I/Library/Java/JavaVirtualMachines/adoptopenjdk-12.0.1.jdk/Contents/Home/include/darwin' --host-ldflags= --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libbluray --enable-libmp3lame --enable-libopus --enable-librubberband --enable-libsnappy --enable-libtesseract --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librtmp --enable-libspeex --enable-videotoolbox --disable-libjack --disable-indev=jack --enable-libaom --enable-libsoxr
  
    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 @ 0x7f98ed005800] Estimating duration from bitrate, this may be inaccurate
  
  Input #0, mp3, from '/.../114006.mp3':
  
    Duration: 00:00:09.53, start: 0.000000, bitrate: 64 kb/s
  
      Stream #0:0: Audio: mp3, 22050 Hz, stereo, fltp, 64 kb/s
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (mp3 (mp3float) -> pcm_s16le (native))
  
  Press [q] to stop, [?] for help
  
  Output #0, wav, to '/.../114-006-a4dec52a.wav':
  
    Metadata:
  
      ISFT            : Lavf58.20.100
  
      Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 16000 Hz, mono, s16, 256 kb/s
  
      Metadata:
  
        encoder         : Lavc58.35.100 pcm_s16le
  
  size=     298kB time=00:00:09.53 bitrate= 256.1kbits/s speed= 469x    
  
  video:0kB audio:298kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.025564%