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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%

• ffplay attempt to subscribe to rtmp server failing with : RTMP_ReadPacket, failed to read RTMP packet header

  8 mars 2018, par johnnydonna

  I have an nginx rtmp server loaded with this docker image : https://github.com/DvdGiessen/nginx-rtmp-docker.

  In general I can stream to it fine with ffmpeg and most of the time connect to the stream fine as well with ffplay. However, for some people, they are unable to subscribe to the RTMP stream at all.

  ffmpeg hosts with this command :

  ffmpeg.exe -f,gdigrab,-framerate,20,-draw_mouse,1,-i,desktop,-c:v,h264_nvenc,-profile:v,main,-delay,0,-preset,default,-rc,vbr,-cq,36,-vf,scale=1024:-2,format=yuv420p,-r,20,-g,40,-y,-f,flv,rtmp://url

  ffplay subscribes with this command :

  ffplay.exe -fflags,nobuffer,-flags,low_delay,-an,-window_title,Screen of User,-framedrop,rtmp://url

  The URL does match the url to which the host is streaming from. What happens is that for about 30 seconds, nothing happens with the following ffplay output :

  nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0<br />
nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0<br />
nan    :  0.000 fd=   0 aq=    0KB vq=    0KB sq=    0B f=0/0

  which repeats until after a while I get the following error :

  RTMP_ReadPacket, failed to read RTMP packet header

  2018/mm/dd 12:--:--:-- [web] rtmp://url: Invalid data found when processing input

  I tried doing what this recommended in regards to the NGINX server setup here : https://github.com/arut/nginx-rtmp-module/issues/1039, setting my worker_processes to 1 which did not change anything.

  It seems like it may just be ffplay timing out but I cannot tell why it occurs only for a few users and not widely. If it is ffplay timing out, what can be done to fix the problem ? It doesn’t seem like an internet speed issue, as these subscribers have pretty good internet. I cannot replicate across different machines, only those few who have continued to have this problem. Any and all help would be appreciated !

• NODE.JS using audioconcat , configured ffmpeg but still have prob

  11 mai 2018, par Adnan Khan

  Want to concatenate two audio files. i used an npm package known as audioconcat but when i installed and configured the below code i am confronted with the following error

  Error: Error: Cannot find ffmpeg
  
      at E:\VoiceMan\registercheck\node_modules\fluent-ffmpeg\lib\processor.js:136:22
  
      at E:\VoiceMan\registercheck\node_modules\fluent-ffmpeg\lib\capabilities.js:123:9
  
      at E:\VoiceMan\registercheck\node_modules\async\dist\async.js:356:16
  
      at nextTask (E:\VoiceMan\registercheck\node_modules\async\dist\async.js:5057:29)
  
      at E:\VoiceMan\registercheck\node_modules\async\dist\async.js:5064:13
  
      at apply (E:\VoiceMan\registercheck\node_modules\async\dist\async.js:21:25)
  
      at E:\VoiceMan\registercheck\node_modules\async\dist\async.js:56:12
  
      at E:\VoiceMan\registercheck\node_modules\async\dist\async.js:840:16
  
      at E:\VoiceMan\registercheck\node_modules\fluent-ffmpeg\lib\capabilities.js:116:11
  
      at E:\VoiceMan\registercheck\node_modules\fluent-ffmpeg\lib\utils.js:223:16
  
  ffmpeg stderr: undefined

  Then I put my problem on stackoverflow. A kind developer suggest me to install ffmpeg also. which i successfully installed and set there path variables but now i am having another issue which tells me that no such file are directry found..i placed my audio files in the same folder of this module.
  here is the error

  working11
  
  working1123423423423
  
  ffmpeg process started: ffmpeg -i concat:audio/a(1).m4a|audio/a(2).m4a|audio/a(3).m4a -y -acodec copy all.m4a
  
  Error: Error: ffmpeg exited with code 1: concat:audio/a(1).m4a|audio/a(2).m4a|audio/a(3).m4a: No such file or directory
  
  
  
      at ChildProcess.<anonymous> (C:\Projects\audio\node_modules\fluent-ffmpeg\lib\processor.js:182:22)
  
      at emitTwo (events.js:126:13)
  
      at ChildProcess.emit (events.js:214:7)
  
      at Process.ChildProcess._handle.onexit (internal/child_process.js:198:12)
  
  ffmpeg stderr: ffmpeg version N-90173-gfa0c9d69d3 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-libas
  
  s --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --ena
  
  ble-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-libvidst
  
  ab --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. 13.100 / 58. 13.100
  
    libavformat    58. 10.100 / 58. 10.100
  
    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
  
  concat:audio/a(1).m4a|audio/a(2).m4a|audio/a(3).m4a: No such file or directory
  
  </anonymous>

  here is my code :

  var audioconcat = require('audioconcat')
  
  
  
  
  
  var songs = [
  
    'a(1).mp3',
  
    'a(2).mp3',
  
    'a(3).mp3'
  
  ]
  
   console.log("working11")
  
  audioconcat(songs)
  
  
  
    .concat('all.mp3') 
  
    .on('start', function (command) {
  
      console.log('ffmpeg process started:', command)
  
    })
  
    .on('error', function (err, stdout, stderr) {
  
      console.error('Error:', err)
  
      console.error('ffmpeg stderr:', stderr)
  
    })
  
    .on('end', function (output) {
  
      console.error('Audio created in:', output)
  
    })
  
     console.log("working1123423423423")