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• ffmpeg Error converting multi channel WAV (13 channels) to AAC ?

  21 mars 2018, par AndySP

  I’m trying to convert a bunch of multi channel wavs that I have into multi channel AAC M4A’s to save space.

  FFMPEG is giving me an error which I don’t understand. Does any one have any ideas ?

  Here is the FFMPEG terminal dump including the command I’m trying to use, the source file is a 13 channel WAV that I exported using Cockos Reaper on the Mac.

  ffmpeg -i ACDC\ -\ Let\ There\ Be\ Rock.wav -c:a libfdk_aac -b:a 640k ACDC.m4a
  
  
  
  ffmpeg version 3.4.2 Copyright (c) 2000-2018 the FFmpeg developers
  
    built with Apple LLVM version 9.0.0 (clang-900.0.39.2)
  
    configuration: --prefix=/usr/local/Cellar/ffmpeg/3.4.2 --enable-shared --enable-pthreads --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --disable-jack --enable-gpl --enable-chromaprint --enable-ffplay --enable-libass --enable-libfdk-aac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenh264 --enable-librsvg --enable-librtmp --enable-librubberband --enable-libsnappy --enable-libsoxr --enable-libssh --enable-libtesseract --enable-libvidstab --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-libzimg --enable-libzmq --enable-opencl --enable-videotoolbox --enable-openssl --enable-lzma --enable-libopenjpeg --disable-decoder=jpeg2000 --extra-cflags=-I/usr/local/Cellar/openjpeg/2.3.0/include/openjpeg-2.3 --enable-nonfree
  
    libavutil      55. 78.100 / 55. 78.100
  
    libavcodec     57.107.100 / 57.107.100
  
    libavformat    57. 83.100 / 57. 83.100
  
    libavdevice    57. 10.100 / 57. 10.100
  
    libavfilter     6.107.100 /  6.107.100
  
    libavresample   3.  7.  0 /  3.  7.  0
  
    libswscale      4.  8.100 /  4.  8.100
  
    libswresample   2.  9.100 /  2.  9.100
  
    libpostproc    54.  7.100 / 54.  7.100
  
  Input #0, wav, from 'ACDC - Let There Be Rock.wav':
  
    Metadata:
  
      encoded_by      : REAPER
  
      date            : 2018-03-21
  
      creation_time   : 12-24-20
  
      time_reference  : 0
  
    Duration: 00:06:10.42, bitrate: 13759 kb/s
  
      Stream #0:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 44100 Hz, 13 channels, s32 (24 bit), 13759 kb/s
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (pcm_s24le (native) -> aac (libfdk_aac))
  
  Press [q] to stop, [?] for help
  
  [auto_resampler_0 @ 0x7fad4d490b60] [SWR @ 0x7fad4e84a200] Rematrix is needed between 13 channels and 7.1(wide) but there is not enough information to do it
  
  [auto_resampler_0 @ 0x7fad4d490b60] Failed to configure output pad on auto_resampler_0
  
  Error reinitializing filters!
  
  Failed to inject frame into filter network: Invalid argument
  
  Error while processing the decoded data for stream #0:0
  
  Conversion failed!

  Any help greatly appreciated, cheers

• ffmpeg Unrecognized option 'var_stream_map'

  3 août 2022, par 陈永林

  I try :

  



  ffmpeg -re -i ./2898654.mp4 -b:a:0 32k -b:a:1 64k -b:v:0 1000k -b:v:1 3000k  \
  -map 0:a -map 0:a -map 0:v -map 0:v -f hls \
  -var_stream_map "a:0,agroup:aud_low a:1,agroup:aud_high v:0,agroup:aud_low v:1,agroup:aud_high" \
  -master_pl_name master.m3u8 \
  ./out_%v.m3u8


  



  error info :

  



  ffmpeg version 3.4.2-1~16.04.york0.2 Copyright (c) 2000-2018 the FFmpeg developers
  built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.9) 20160609
  configuration: --prefix=/usr --extra-version='1~16.04.york0.2' --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
  libavutil      55. 78.100 / 55. 78.100
  libavcodec     57.107.100 / 57.107.100
  libavformat    57. 83.100 / 57. 83.100
  libavdevice    57. 10.100 / 57. 10.100
  libavfilter     6.107.100 /  6.107.100
  libavresample   3.  7.  0 /  3.  7.  0
  libswscale      4.  8.100 /  4.  8.100
  libswresample   2.  9.100 /  2.  9.100
  libpostproc    54.  7.100 / 54.  7.100
Unrecognized option 'var_stream_map'.
Error splitting the argument list: Option not found


  


• AWS Lambda - ffmpeg erroneously outputs distorted/clipped mp3

  23 mars 2018, par Eric Amshukov

  How do I get an mp3 output without any distortion using ffmpeg ?

  I am utilizing ffmpeg on AWS Lambda linux using the static build provided by https://www.johnvansickle.com/ffmpeg/ (x86_64 build).

  After running the following command, the mp3 output has terrible clipping/distortion.

  ffmpeg -loglevel verbose -ss 0 -t 30 -y -i /tmp/ick_20180323005225.wav -codec:a libmp3lame -qscale:a 7 /tmp/ick_20180323005225-opa.mp3

  Edit : here is the sample file that I used :

  http://www.brainybetty.com/FacebookFans/Feb112010/strings.wav

  Here is the log coming from Lambda :

  Executing command '/tmp/ffmpeg -loglevel verbose -ss 0 -t 30 -y -i /tmp/ick_20180323005225.wav -codec:a libmp3lame -qscale:a 7 /tmp/ick_20180323005225-opa.mp3' ...
  
  STDERR:
  
  ffmpeg version 3.4.2-static https://johnvansickle.com/ffmpeg/ Copyright (c) 2000-2018 the FFmpeg developers
  
  built with gcc 6.3.0 (Debian 6.3.0-18) 20170516
  
  configuration: --enable-gpl --enable-version3 --enable-static --disable-debug --disable-ffplay --disable-indev=sndio --disable-outdev=sndio --cc=gcc-6 --enable-fontconfig --enable-frei0r --enable-gnutls --enable-gray --enable-libfribidi --enable-libass --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librubberband --enab
  
  libavutil 55. 78.100 / 55. 78.100
  
  libavcodec 57.107.100 / 57.107.100
  
  libavformat 57. 83.100 / 57. 83.100
  
  libavdevice 57. 10.100 / 57. 10.100
  
  libavfilter 6.107.100 / 6.107.100
  
  libswscale 4. 8.100 / 4. 8.100
  
  libswresample 2. 9.100 / 2. 9.100
  
  libpostproc 54. 7.100 / 54. 7.100
  
  [wav @ 0x4bbdf40] parser not found for codec pcm_s16le, packets or times may be invalid.
  
  Guessed Channel Layout for Input Stream #0.0 : stereo
  
  Input #0, wav, from '/tmp/ick_20180323005225.wav':
  
  Duration: 00:00:05.00, bitrate: 1411 kb/s
  
  Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
  
  Stream mapping:
  
  Stream #0:0 -> #0:0 (pcm_s16le (native) -> mp3 (libmp3lame))
  
  Press [q] to stop, [?] for help
  
  [graph_0_in_0_0 @ 0x4bc64e0] tb:1/44100 samplefmt:s16 samplerate:44100 chlayout:0x3
  
  [format_out_0_0 @ 0x4bc6360] auto-inserting filter 'auto_resampler_0' between the filter 'Parsed_anull_0' and the filter 'format_out_0_0'
  
  [auto_resampler_0 @ 0x4bd2ee0] ch:2 chl:stereo fmt:s16 r:44100Hz -> ch:2 chl:stereo fmt:s16p r:44100Hz
  
  Output #0, mp3, to '/tmp/ick_20180323005225-opa.mp3':
  
  Metadata:
  
  TSSE : Lavf57.83.100
  
  Stream #0:0: Audio: mp3 (libmp3lame), 44100 Hz, stereo, s16p, delay 1105
  
  Metadata:
  
  encoder : Lavc57.107.100 libmp3lame
  
  size= 11kB time=00:00:00.73 bitrate= 119.5kbits/s speed=1.41x
  
  size= 23kB time=00:00:01.67 bitrate= 114.5kbits/s speed=1.61x
  
  size= 36kB time=00:00:02.61 bitrate= 113.4kbits/s speed=1.65x
  
  size= 48kB time=00:00:03.55 bitrate= 111.0kbits/s speed=1.69x
  
  size= 60kB time=00:00:04.46 bitrate= 109.6kbits/s speed=1.71x
  
  No more output streams to write to, finishing.
  
  size= 67kB time=00:00:05.01 bitrate= 108.9kbits/s speed=1.75x
  
  video:0kB audio:66kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.371917%
  
  Input file #0 (/tmp/ick_20180323005225.wav):
  
  Input stream #0:0 (audio): 216 packets read (881988 bytes); 216 frames decoded (220497 samples);
  
  Total: 216 packets (881988 bytes) demuxed
  
  Output file #0 (/tmp/ick_20180323005225-opa.mp3):
  
  Output stream #0:0 (audio): 192 frames encoded (220497 samples); 193 packets muxed (68026 bytes);
  
  Total: 193 packets (68026 bytes) muxed
  
  Executed command '/tmp/ffmpeg -loglevel verbose -ss 0 -t 30 -y -i /tmp/ick_20180323005225.wav -codec:a libmp3lame -qscale:a 7 /tmp/ick_20180323005225-opa.mp3' with code: 0.