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• 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%

• Parse and drop gain control data, so that SSR packets decode.

  16 février 2018, par Dale Curtis

  Parse and drop gain control data, so that SSR packets decode.
  This will result in poor quality audio for SSR streams, but they
  
  will at least demux and decode without error ; partially fixing
  
  ticket #1693.
  This pulls in the decode_gain_control() function from the
  
  ffmpeg summer-of-code repo (original author Maxim Gavrilov) at
  
  svn ://svn.mplayerhq.hu/soc/aac/aac.c with some minor modifications
  
  and adds AOT_AAC_SSR to decode_audio_specific_config_gb().
  Signed-off-by : Dale Curtis <dalecurtis@chromium.org>
  
  Co-authored-by : Maxim Gavrilov <maxim.gavrilov@gmail.com>
  

  • [DH] libavcodec/aac.h
  • [DH] libavcodec/aacdec_template.c

• Convert audio files to mp3 using ffmpeg [closed]

  24 mars, par Hrishikesh -Rishi- Choudhari

  I need to convert audio files to mp3 using ffmpeg.

  



  When I write the command as ffmpeg -i audio.ogg -acodec mp3 newfile.mp3, I get the error :

  



  FFmpeg version 0.5.2, Copyright (c) 2000-2009 Fabrice Bellard, et al.
  configuration: 
  libavutil     49.15. 0 / 49.15. 0
  libavcodec    52.20. 1 / 52.20. 1
  libavformat   52.31. 0 / 52.31. 0
  libavdevice   52. 1. 0 / 52. 1. 0
  built on Jun 24 2010 14:56:20, gcc: 4.4.1
Input #0, mp3, from 'ZHRE.mp3':
  Duration: 00:04:12.52, start: 0.000000, bitrate: 208 kb/s
    Stream #0.0: Audio: mp3, 44100 Hz, stereo, s16, 256 kb/s
Output #0, mp3, to 'audio.mp3':
    Stream #0.0: Audio: 0x0000, 44100 Hz, stereo, s16, 64 kb/s
Stream mapping:
  Stream #0.0 -> #0.0
Unsupported codec for output stream #0.0


  



  I also ran this command :

  



   ffmpeg -formats | grep mp3


  



  and got this in response :

  



  FFmpeg version 0.5.2, Copyright (c) 2000-2009 Fabrice Bellard, et al.
  configuration: 
  libavutil     49.15. 0 / 49.15. 0
  libavcodec    52.20. 1 / 52.20. 1
  libavformat   52.31. 0 / 52.31. 0
  libavdevice   52. 1. 0 / 52. 1. 0
  built on Jun 24 2010 14:56:20, gcc: 4.4.1
 DE mp3             MPEG audio layer 3
 D A    mp3             MP3 (MPEG audio layer 3)
 D A    mp3adu          ADU (Application Data Unit) MP3 (MPEG audio layer 3)
 D A    mp3on4          MP3onMP4
 text2movsub remove_extra noise mov2textsub mp3decomp mp3comp mjpegadump imxdump h264_mp4toannexb dump_extra


  



  I guess that the mp3 codec isn't installed. Am I on the right track here ?