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• FFmpeg audio stream extraction on non-interleaved AVI - slow compared to AviSynth

  8 mai 2019, par LLL

  I want to extract the audio stream of an avi file as a wav file, it works but it is really slow ( 4-5fps) although I just want to copy the stream.

  Here is the type of stream I want to extract (ffprobe info) :
  Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s

  Going through AviSynth does it about 100 times faster, but I would prefer a pure FFmpeg solution. Why such a speed difference ? It looks like FFmpeg is reading and processing through the whole file whereas AviSynth can just extract the data without reading it.

  Example :
  ffmpeg -i file.avi -vn -ac 2 -c:a copy audio.wav
  or
  ffmpeg -i file.avi -map 0:a -ac 2 -c:a copy audio.wav
  both work fine but take time.

  Using an AviSynth script as input :
  ffmpeg -i script.avs -map 0:a -ac 2 -c:a copy audio.wav
  with script.avs containing just :
  AviSource("file.avi")
  does the same but almost instantaneously !

  Any idea why AviSynth is so much faster and if there is a way to get the same speed in FFmpeg ?

  Edit : adding logs
  Using FFmpeg directly :

  E:\>ffmpeg -i "file.avi" -map 0:a -c:a copy -y -benchmark "output.wav"
  
  ffmpeg version N-92936-ged3b64402e Copyright (c) 2000-2019 the FFmpeg developers
  
    built with gcc 8.2.1 (GCC) 20181201
  
    configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt
  
    libavutil      56. 25.100 / 56. 25.100
  
    libavcodec     58. 43.100 / 58. 43.100
  
    libavformat    58. 25.100 / 58. 25.100
  
    libavdevice    58.  6.101 / 58.  6.101
  
    libavfilter     7. 47.100 /  7. 47.100
  
    libswscale      5.  4.100 /  5.  4.100
  
    libswresample   3.  4.100 /  3.  4.100
  
    libpostproc    55.  4.100 / 55.  4.100
  
  [avi @ 0000018d3c38a680] non-interleaved AVI
  
  Guessed Channel Layout for Input Stream #0.1 : stereo
  
  Input #0, avi, from 'file.avi':
  
    Duration: 00:18:37.49, start: 0.000000, bitrate: 534682 kb/s
  
      Stream #0:0: Video: rawvideo, bgr24, 1280x720, 533183 kb/s, 24.11 fps, 24.11 tbr, 24.10 tbn, 24.10 tbc
  
      Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
  
  Output #0, wav, to 'output.wav':
  
    Metadata:
  
      ISFT            : Lavf58.25.100
  
      Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
  
  Stream mapping:
  
    Stream #0:1 -> #0:0 (copy)
  
  Press [q] to stop, [?] for help
  
  size=  192445kB time=00:18:37.12 bitrate=1411.2kbits/s speed=4.77x
  
  video:0kB audio:192445kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000040%
  
  bench: utime=1.188s stime=50.766s rtime=234.254s
  
  bench: maxrss=17468kB

  Using AviSynth :

  E:\>ffmpeg -i "soundout.avs" -map 0:a -c:a copy -y -benchmark "output.wav"
  
  ffmpeg version N-92936-ged3b64402e Copyright (c) 2000-2019 the FFmpeg developers
  
    built with gcc 8.2.1 (GCC) 20181201
  
    configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt
  
    libavutil      56. 25.100 / 56. 25.100
  
    libavcodec     58. 43.100 / 58. 43.100
  
    libavformat    58. 25.100 / 58. 25.100
  
    libavdevice    58.  6.101 / 58.  6.101
  
    libavfilter     7. 47.100 /  7. 47.100
  
    libswscale      5.  4.100 /  5.  4.100
  
    libswresample   3.  4.100 /  3.  4.100
  
    libpostproc    55.  4.100 / 55.  4.100
  
  Guessed Channel Layout for Input Stream #0.1 : stereo
  
  Input #0, avisynth, from 'soundout.avs':
  
    Duration: 00:18:37.49, start: 0.000000, bitrate: N/A
  
      Stream #0:0: Video: rawvideo (BGR[24] / 0x18524742), bgr24, 1280x720, 24.11 fps, 24.11 tbr, 24.10 tbn, 24.10 tbc
  
      Stream #0:1: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
  
  Output #0, wav, to 'output.wav':
  
    Metadata:
  
      ISFT            : Lavf58.25.100
  
      Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
  
  Stream mapping:
  
    Stream #0:1 -> #0:0 (copy)
  
  Press [q] to stop, [?] for help
  
  size=  192445kB time=00:18:37.11 bitrate=1411.2kbits/s speed= 155x
  
  video:0kB audio:192445kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000040%
  
  bench: utime=0.234s stime=1.047s rtime=7.236s
  
  bench: maxrss=23792kB

  Edit : tests after "reencoding" AVI file :
  Onto something...
  Say my original file is f.avi. Here is ffprobe’s results :

  [avi @ 0x55a9c4b1e740] non-interleaved AVI
  
  Input #0, avi, from 'f.avi':
  
    Duration: 00:00:38.18, start: 0.000000, bitrate: 1104582 kb/s
  
      Stream #0:0: Video: rawvideo, bgr24, 1632x1200, 1104265 kb/s, 23.47 fps, 23.47 tbr, 23.47 tbn, 23.47 tbc
  
      Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 2 channels, s16, 1411 kb/s

  Extracting audio takes a long time.
  Now if I "reencode" the file in another AVI :

  ffmpeg -i f.avi -c copy f2.avi

  I can extract the audio from f2.avi in milliseconds !
  FFprobe on f2.avi :

  Input #0, avi, from 'f2.avi':
  
    Metadata:
  
      encoder         : Lavf57.56.101
  
    Duration: 00:00:38.18, start: 0.000000, bitrate: 1104456 kb/s
  
      Stream #0:0: Video: rawvideo, bgr24, 1632x1200, 1104265 kb/s, 23.47 fps, 23.47 tbr, 23.47 tbn, 23.47 tbc
  
      Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 2 channels, s16, 1411 kb/s

  It’s the same apart from the Metadata, which shouldn’t make a difference, but with this comparison I see the problem must have to do with the fact that the original is non-interleaved !
  I would assume it was easier to read and extract the audio from a non-interleaved file but maybe this is not conforming to AVI standards, hence the extra work needed ?

• Subtitles in ffmpeg/libavfilter

  15 juin 2021, par Captain Jack

  I have a C program to read video/audio with libav/ffmpeg libraries and decode it.

  



  I am playing with some filters and most work just fine. I can draw text, overlay logos, flip and invert video colours. However, I am having big issues overlaying subtitles.

  



  My filter is very simple.

  



  const char *vfilter_descr = "[in]subtitles=subs.srt[out]";


  



  On the console I get this :

  



  [Parsed_subtitles_0 @ 0x7fe76c703240] Shaper: FriBidi 0.19.7 (SIMPLE) HarfBuzz-ng 2.4.0 (COMPLEX)
[Parsed_subtitles_0 @ 0x7fe76c703240] Using font provider coretext
[Parsed_subtitles_0 @ 0x7fe76c703240] fontselect: (Arial, 400, 0) -> /Library/Fonts/Microsoft/Arial.ttf, -1, ArialMT
[Parsed_subtitles_0 @ 0x7fe76c703240] fontselect: (Arial, 400, 100) -> /Library/Fonts/Microsoft/Arial Italic.ttf, -1, Arial-ItalicMT


  



  ...which somewhat confirms that subtitles are loading, though I am not sure why there are two fonts being loaded ?

  



  However, they are not showing at all - almost as if they never loaded. I tried several different files, including ASS ones but no luck.

  



  ffmpeg version is the latest one.

  



  $ ffmpeg -v
ffmpeg version 4.1.3 Copyright (c) 2000-2019 the FFmpeg developers
  built with Apple LLVM version 9.0.0 (clang-900.0.39.2)
  configuration: --prefix=/usr/local/Cellar/ffmpeg/4.1.3_1 --enable-shared --enable-pthreads --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags='-I/Library/Java/JavaVirtualMachines/adoptopenjdk-11.0.2.jdk/Contents/Home/include -I/Library/Java/JavaVirtualMachines/adoptopenjdk-11.0.2.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


  



  Any ideas ?

  


• How to transcode raw uncompressed RTP to an H264 RTSP stream

  10 mai 2019, par Gino

  I am new to streaming and am trying to figure out how to transcode streams via ffmpeg.

  I have a few raw rtp uncompressed streams where some are on address 239.x.x.x and others are on 169.x.x.x.

  I want to setup an RTSP server to grab those streams and transcode them into H264 and stream them out to a new address and port.

  I have tried some ffmpeg commands but I keep getting errors about having to compile ffmpeg with pthreads.

  I have no idea how to do that so does anyone know what commands I can use that will work with the current windows version of ffmpeg ?

  For now, I am just trying to save the stream to a file to see if that works. Command I am using is :

  ffmpeg -i rtp://224.1.1.10:6972 transcoded test.mp4

  and the return I get in the command line is

  ffmpeg version 4.1.3 Copyright (c) 2000-2019 the FFmpeg developers
  
    built with gcc 8.3.1 (GCC) 20190414
  
    configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth
  
    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
  
    libswscale      5.  3.100 /  5.  3.100
  
    libswresample   3.  3.100 /  3.  3.100
  
    libpostproc    55.  3.100 / 55.  3.100
  
  
  
  [udp @ 000002cb292abf40] 'circular_buffer_size' option was set but it is not supported on this build (pthread support is required)  
  
  [udp @ 000002cb292bc200] 'circular_buffer_size' option was set but it is not supported on this build (pthread support is required)  
  
  rtp://224.1.1.10:6972: Immediate exit requested  
  
  Exiting normally, received signal 2.