Recherche avancée

Sur d’autres sites (4097)

• FFMPEG : Invalid Pixel Format String -1 when converting MP3 to SWF

  22 janvier 2013, par James Riley

  I'm having an issue with FFMPEG - receiving an error when trying to convert an MP3 into an SWF. The full output contains :

  $ ffmpeg -i track1.mp3 -y -ar '44100' -ab '96k' 'sample.swf'
  
      FFmpeg version SVN-r26402, Copyright (c) 2000-2011 the FFmpeg developers
  
      configuration: --enable-libmp3lame --enable-libvorbis --enable-shared --disable-mmx
  
      libavutil     50.36. 0 / 50.36. 0
  
      libavcore      0.16. 1 /  0.16. 1
  
      libavcodec    52.108. 0 / 52.108. 0
  
      libavformat   52.93. 0 / 52.93. 0
  
      libavdevice   52. 2. 3 / 52. 2. 3
  
      libavfilter    1.74. 0 /  1.74. 0
  
      libswscale     0.12. 0 /  0.12. 0
  
      Input #0, nsv, from 'track1.mp3':
  
       Metadata:
  
         encoder         : Lavf52.93.0
  
         artist          : Test
  
         title           : Track02
  
         album           : Mixtape
  
         disc            : 1/1
  
         genre           : Acoustic
  
         date            : 2013
  
         track           : 2/2
  
         album_artist    : Test
  
         Duration: 00:00:00.00, start: 0.000000, bitrate: -2147483 kb/s
  
         Stream #0.0: Video: [216][195]2[138] / 0x8A32C3D8, 44318x50468, 24 fps, 24 tbr, 24 tbn, 24 tbc
  
         Stream #0.1: Audio: h[228][146][142] / 0x8E92E468, 0 channels
  
      [buffer @ 0x175cd00] Invalid pixel format string '-1'
  
      Error opening filters!

  I'm aware the FFmpeg version here is old, but the issue is only occuring for one of the MP3s I am testing here. I have many others that are working just fine.

  What can be causing this issue and what steps can I take to resolve this ?

  From what I understand, the pixel format relates to the video output, yet in this case, the SWF is only to be used for audio. I've tested this on a newer version of FFmpeg and the issue isn't occuring - I certainly plan to upgrade FFmpeg on the test server I'm using, but would like to get to the bottom of why this is happening and try to get this particular MP3 converting, where its currently failing despite many others not having the same issue.

  Thanks for any help !

• Audacity vocal removal failed when ffmpeg-conversion was involved

  10 mars 2018, par fyang

  I downloaded some songs coded with FLAC, and Audacity could remove the vocals quite well.

  When I downloaded songs coded with ALAC, I must use ffmpeg to convert them to some other forms because Audacity didn’t recognise .m4a files.

  I used the command ffmpeg -i "song 01.m4a" -f flac "song 01.flac". Now Audacity could load the song, but its vocal removal failed to remove the vocals.

  I tried again with this command in order to be precise, ffmpeg -i "song 01.m4a" -af "pan=stereo|c0=c0|c1=c1" -f flac "song 01.flac", and vocal removal did not work either.

  I tried to do it manually by splitting, inverting and changing both channels to mono, but the vocals were still there.

  I think the problem lies with the ffmpeg conversion step. Is there any fix ? Thanks !

  Below is the result of the conversion :

  ffmpeg -i "song 01.m4a" -af "pan=stereo|c0=c0|c1=c1" -f flac "song 01.flac"
  
  ffmpeg version N-90143-gb6652f5100 Copyright (c) 2000-2018 the FFmpeg developers
  
    built with gcc 7.3.0 (GCC)
  
    configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-bzlib --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-libmfx --enable-amf --enable-cuda --enable-cuvid --enable-d3d11va --enable-nvenc --enable-dxva2 --enable-avisynth
  
    libavutil      56.  7.101 / 56.  7.101
  
    libavcodec     58. 12.102 / 58. 12.102
  
    libavformat    58.  9.100 / 58.  9.100
  
    libavdevice    58.  2.100 / 58.  2.100
  
    libavfilter     7. 12.100 /  7. 12.100
  
    libswscale      5.  0.101 /  5.  0.101
  
    libswresample   3.  0.101 /  3.  0.101
  
    libpostproc    55.  0.100 / 55.  0.100
  
  [mov,mp4,m4a,3gp,3g2,mj2 @ 0000019f2b258000] stream 0, timescale not set
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'song 01.m4a':
  
    Metadata:
  
      major_brand     : M4A
  
      minor_version   : 0
  
      compatible_brands: M4A mp42isom
  
      creation_time   : 2009-12-27T00:15:23.000000Z
  
      track           : 1/10
  
      genre           : 
  
      album           : 
  
      artist          : 
  
      comment         : ExactAudioCopy v0.95b4
  
      DISCID          : 
  
      iTunNORM        :  00000F32 00000E1D 0000547D 00005B93 0006C3CA 0006C43E 00007FF8 00007FFF 00058227 0003593B
  
      title           : song 01
  
      encoder         : iTunes 9.0.2.25
  
      date            : 2005
  
      album_artist    : 
  
      lyrics          : 
  
    Duration: 00:08:10.84, start: 0.000000, bitrate: 921 kb/s
  
      Stream #0:0(und): Audio: alac (alac / 0x63616C61), 44100 Hz, stereo, s16p, 920 kb/s (default)
  
      Metadata:
  
        creation_time   : 2009-12-27T00:15:23.000000Z
  
      Stream #0:1: Video: mjpeg, yuvj444p(pc, bt470bg/unknown/unknown), 300x300 [SAR 100:100 DAR 1:1], 90k tbr, 90k tbn, 90k tbc
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (alac (native) -> flac (native))
  
  Press [q] to stop, [?] for help
  
  [Parsed_pan_0 @ 0000019f2b2a6fc0] Pure channel mapping detected: 0 1
  
  Output #0, flac, to 'song 01.flac':
  
    Metadata:
  
      major_brand     : M4A
  
      minor_version   : 0
  
      compatible_brands: M4A mp42isom
  
      lyrics          : 
  
      TRACKNUMBER     : 1/10
  
      genre           : 
  
      album           : 
  
      artist          : 
  
      DESCRIPTION     : ExactAudioCopy v0.95b4
  
      DISCID          : 
  
      iTunNORM        :  00000F32 00000E1D 0000547D 00005B93 0006C3CA 0006C43E 00007FF8 00007FFF 00058227 0003593B
  
      title           : song 01
  
      ALBUMARTIST     : 
  
      date            : 2005
  
      encoder         : Lavf58.9.100
  
      Stream #0:0(und): Audio: flac, 44100 Hz, stereo, s16, 128 kb/s (default)
  
      Metadata:
  
        creation_time   : 2009-12-27T00:15:23.000000Z
  
        encoder         : Lavc58.12.102 flac
  
  size=   54518kB time=00:08:10.84 bitrate= 909.9kbits/s speed=  35x
  
  video:0kB audio:54508kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.018294%

• FFMPEG Concat protocol Error : Found duplicated MOOV Atom. Skipped it

  11 septembre 2020, par user3541092

  I'm concatenating mp3 files into a single m4a.

  



  To save time, I'm doing the mp3 to m4a conversion first (multi-threaded) and then concatenating the output files (m4a) into a single m4a file.

  



  The concat works when I do the conversion (mp3 -> m4a) and cancat in one swoop. But it takes forever to run.

  



  Example :

  



  ffmpeg -i "concat:001.mp3|002.mp3|003.mp3" -codec:a aac -c copy output.m4a


  



  
If I take those same mp3 files and convert them to m4a in parallel and then concat, I can save myself a significant amount time.

  



  The problem is if I run the concat command on the converted m4a files

  



  Example cmd :

  



  ffmpeg -i "concat:001.m4a|002.m4a|003.m4a" -c copy output.m4a


  



  I getting the following error :

  



  


  [mov,mp4,m4a,3gp,3g2,mj2 @ 000000000052f680] Found duplicated MOOV
 Atom. Skipped it

  


  



  
UPDATE : Ouput when concatenating m4a files (two track example) :

  



  C:\_ffmpeg>ffmpeg -i "concat:001.m4a|002.m4a" -c copy output.m4a
ffmpeg version N-79630-g9ac154d Copyright (c) 2000-2016 the FFmpeg developers
  built with gcc 5.3.0 (GCC)
  configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av
isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab
le-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --
enable-libfreetype --enable-libgme --enable-libgsm --enable-libilbc --enable-lib
modplug --enable-libmfx --enable-libmp3lame --enable-libopencore-amrnb --enable-
libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enabl
e-libschroedinger --enable-libsnappy --enable-libsoxr --enable-libspeex --enable
-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-amrwbenc --ena
ble-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx
264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-libzimg --enable
-lzma --enable-decklink --enable-zlib
  libavutil      55. 22.101 / 55. 22.101
  libavcodec     57. 38.100 / 57. 38.100
  libavformat    57. 34.103 / 57. 34.103
  libavdevice    57.  0.101 / 57.  0.101
  libavfilter     6. 44.100 /  6. 44.100
  libswscale      4.  1.100 /  4.  1.100
  libswresample   2.  0.101 /  2.  0.101
  libpostproc    54.  0.100 / 54.  0.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 000000000048a9a0] Found duplicated MOOV Atom. Skipped
 it
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'concat:001.m4a|002.m4a':
  Metadata:
    track           : 1
    major_brand     : M4A
    minor_version   : 512
    title           : 001 Moon Over Soho
    artist          : Ben Aaronovitch
    album_artist    : Narrated by Kobna Holdbrook-Smith
    composer        : Narrated by Kobna Holdbrook-Smith
    album           : Moon Over Soho
    date            : 2012
    encoder         : Lavf57.34.103
    genre           : Speech
    copyright       : Tantor Audio
    compatible_brands: isomiso2
  Duration: 00:44:03.83, start: 0.000000, bitrate: 251 kb/s
    Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, flt
p, 128 kb/s (default)
    Metadata:
      handler_name    : SoundHandler
[ipod @ 000000000048bb20] Using AVStream.codec to pass codec parameters to muxer
s is deprecated, use AVStream.codecpar instead.
Output #0, ipod, to 'output.m4a':
  Metadata:
    track           : 1
    major_brand     : M4A
    minor_version   : 512
    title           : 001 Moon Over Soho
    artist          : Ben Aaronovitch
    album_artist    : Narrated by Kobna Holdbrook-Smith
    composer        : Narrated by Kobna Holdbrook-Smith
    album           : Moon Over Soho
    date            : 2012
    compatible_brands: isomiso2
    genre           : Speech
    copyright       : Tantor Audio
    encoder         : Lavf57.34.103
    Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, stereo, 128
 kb/s (default)
    Metadata:
      handler_name    : SoundHandler
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
size=   41892kB time=00:44:03.80 bitrate= 129.8kbits/s speed=2.21e+003x
video:0kB audio:41445kB subtitle:0kB other streams:0kB global headers:0kB muxing
 overhead: 1.077122%

C:\_ffmpeg>pause
Press any key to continue . . .