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• How to detect audio sampling rate with avprobe / ffprobe ?

  8 août 2013, par Devy

  I am using libav 9.6, installed via Homebrew.

  $ avprobe -version
  
  avprobe version 9.6, Copyright (c) 2007-2013 the Libav developers
  
    built on Jun  8 2013 02:44:19 with Apple LLVM version 4.2 (clang-425.0.24) (based on LLVM 3.2svn)
  
  avprobe 9.6
  
  libavutil     52.  3. 0 / 52.  3. 0
  
  libavcodec    54. 35. 0 / 54. 35. 0
  
  libavformat   54. 20. 3 / 54. 20. 3
  
  libavdevice   53.  2. 0 / 53.  2. 0
  
  libavfilter    3.  3. 0 /  3.  3. 0
  
  libavresample  1.  0. 1 /  1.  0. 1
  
  libswscale     2.  1. 1 /  2.  1. 1

  Even though the sampling rate is displayed in the stdout in the command line output, the -show_format option doesn't surface the sampling rate information for the audio file at all.

  Here is the BASH terminal output :

  $ avprobe  -v verbose -show_format -of json  sample.gsm
  
  avprobe version 9.6, Copyright (c) 2007-2013 the Libav developers
  
    built on Jun  8 2013 02:44:19 with Apple LLVM version 4.2 (clang-425.0.24)
  
  (based on LLVM 3.2svn)
  
    configuration: --prefix=/usr/local/Cellar/libav/9.6 --enable-shared
  
  --enable-pthreads --enable-gpl --enable-version3 --enable-nonfree
  
  --enable-hardcoded-tables --enable-avresample --enable-vda --enable-gnutls
  
  --enable-runtime-cpudetect --disable-indev=jack --cc=cc --host-cflags=
  
  --host-ldflags= --enable-libx264 --enable-libfaac --enable-libmp3lame
  
  --enable-libxvid --enable-avplay
  
    libavutil     52.  3. 0 / 52.  3. 0
  
    libavcodec    54. 35. 0 / 54. 35. 0
  
    libavformat   54. 20. 3 / 54. 20. 3
  
    libavdevice   53.  2. 0 / 53.  2. 0
  
    libavfilter    3.  3. 0 /  3.  3. 0
  
    libavresample  1.  0. 1 /  1.  0. 1
  
    libswscale     2.  1. 1 /  2.  1. 1
  
  [gsm @ 0x7f8012806600] Estimating duration from bitrate, this may be inaccurate
  
  Input #0, gsm, from 'sample.gsm':
  
    Duration: 00:03:52.32, start: 0.000000, bitrate: 13 kb/s
  
      Stream #0.0: Audio: gsm, 8000 Hz, mono, s16, 13 kb/s
  
  {  "format" : {
  
      "filename" : "sample.gsm",
  
      "nb_streams" : 1,
  
      "format_name" : "gsm",
  
      "format_long_name" : "raw GSM",
  
      "start_time" : "0.000000",
  
      "duration" : "232.320000",
  
      "size" : "383328.000000",
  
      "bit_rate" : "13200.000000"
  
    }}

  And the python code example :

  >>> filename = 'sample.gsm'
  
  >>> result = subprocess.check_output(['avprobe', '-show_format', '-of',
  
  'json', filename])
  
  avprobe version 9.6, Copyright (c) 2007-2013 the Libav developers
  
    built on Jun  8 2013 02:44:19 with Apple LLVM version 4.2
  
  (clang-425.0.24) (based on LLVM 3.2svn)
  
  [gsm @ 0x7fe0b1806600] Estimating duration from bitrate, this may be
  
  inaccurate
  
  Input #0, gsm, from 'sample.gsm':
  
    Duration: 00:03:52.32, start: 0.000000, bitrate: 13 kb/s
  
      Stream #0.0: Audio: gsm, 8000 Hz, mono, s16, 13 kb/s
  
  >>> print result
  
  {  "format" : {
  
      "filename" : "sample.gsm",
  
      "nb_streams" : 1,
  
      "format_name" : "gsm",
  
      "format_long_name" : "raw GSM",
  
      "start_time" : "0.000000",
  
      "duration" : "232.320000",
  
      "size" : "383328.000000",
  
      "bit_rate" : "13200.000000"
  
  }}

  So I am aware that sampling rate could be a stream specific display to be shown in -show_format option results. But there isn't any other options to detect the sampling rate on a specific audio stream even though it's possible to set it with -ar when re-encoding it.

  I filed a ticket to libav but I am just curious if there is any other way to extract sampling rate from libav probing utils. I appreciate the answer beforehand.

  PS : it would be the same question for the upstream project of ffmpeg (ffprobe) in this case.

• FFmpeg cant convert mov to flv [migrated]

  26 février 2013, par 5et

  When i try to convert .mov file to .flv i get this type of output. I get error "Invalid sample format '(null)' Error opening filters !". When i convert .avi or .mp4 i get no errors. Im on windows

      error: ffmpeg -i C:/wamp/www/\ds_uploads\62\video\23c4a54cbf8bc73056cb370ae7371848.mov -y -f flv -ar 44100 -q:v 0 C:/wamp/www/\ds_uploads\62\video\23c4a54cbf8bc73056cb370ae7371848.flv 2>&1
  
  ffmpeg version N-47062-g26c531c Copyright (c) 2000-2012 the FFmpeg developers
  
    built on Nov 25 2012 12:21:26 with gcc 4.7.2 (GCC)
  
    configuration: --enable-gpl --enable-version3 --disable-pthreads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libnut --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libutvideo --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
  
    libavutil      52.  9.100 / 52.  9.100
  
    libavcodec     54. 77.100 / 54. 77.100
  
    libavformat    54. 37.100 / 54. 37.100
  
    libavdevice    54.  3.100 / 54.  3.100
  
    libavfilter     3. 23.102 /  3. 23.102
  
    libswscale      2.  1.102 /  2.  1.102
  
    libswresample   0. 17.101 /  0. 17.101
  
    libpostproc    52.  2.100 / 52.  2.100
  
  [mov,mp4,m4a,3gp,3g2,mj2 @ 01f8b200] multiple edit list entries, a/v desync might occur, patch welcome
  
  [mov,mp4,m4a,3gp,3g2,mj2 @ 01f8b200] max_analyze_duration 5000000 reached at 5015510
  
  Guessed Channel Layout for  Input Stream #0.1 : mono
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:/wamp/www/\ds_uploads\62\video\23c4a54cbf8bc73056cb370ae7371848.mov':
  
    Metadata:
  
      creation_time   : 1998-11-04 16:40:13
  
    Duration: 00:01:00.83, start: 0.000000, bitrate: 110 kb/s
  
      Stream #0:0(eng): Video: svq1 (SVQ1 / 0x31515653), yuv410p, 160x120, 90 kb/s, 7.51 fps, 7.50 tbr, 600 tbn, 600 tbc
  
      Metadata:
  
        creation_time   : 1998-11-04 16:40:13
  
        handler_name    : Apple Alias Data Handler
  
      Stream #0:1(eng): Audio: qdmc (QDMC / 0x434D4451), 44100 Hz, mono
  
      Metadata:
  
        creation_time   : 1998-11-04 16:40:13
  
        handler_name    : Apple Alias Data Handler
  
  [graph 1 input from stream 0:1 @ 02b5f040] Invalid sample format '(null)'
  
  Error opening filters!

• ffmpeg : alpha merge add audio [closed]

  21 juin 2022, par Alex

  I am alpha merging two videos for iOS/Apple devices (mov), the color.mp4 has audio, while alpha.mp4 has not. The end result has no audio ouput. What is the proper flag to grab audio from color.mp4 onto the output ?

  


   ffmpeg -y -i color.mp4 -i alpha.mp4 -f lavfi -i color=c=black:s=320x568 -filter_complex "[1:v]scale=320:568,setsar=1:1,split[vs][alpha];[0:v][vs]alphamerge[vt];[2:v][vt]overlay=shortest=1[rgb];[rgb][alpha]alphamerge" -shortest -c:v hevc_videotoolbox -allow_sw 1 -alpha_quality 0.75 -vtag hvc1 -pix_fmt yuva420p -an output.mov