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• find in bash to convert files with ffmpeg - paths being changed

  20 avril 2017, par J V

  Trying to convert a bunch of files with ffmpeg and find.

  find -name "*.mkv" | while read f
  
  do ffmpeg -i "$f" -c copy -y "${f%.*}.mp4"
  
  done

  This should work right ? It results in :

  + read f
  
  + ffmpeg -i '/file.mkv' -c copy -y '/file.mp4'
  
  ffmpeg version N-61155-g61ff043 Copyright (c) 2000-2014 the FFmpeg developers
  
    built on Mar  7 2014 19:17:59 with gcc 4.8 (Debian 4.8.2-16)
  
  /file.mkv: No such file or directory

  So I naturally try sticking the missing dot in front of the variable :

  find -name "*.mkv" | while read f
  
  do ffmpeg -i ".$f" -c copy -y ".${f%.*}.mp4"
  
  done

  The result :

  + read f
  
  + ffmpeg -i '../file.mkv' -c copy -y '../file.mp4'
  
  ffmpeg version N-61155-g61ff043 Copyright (c) 2000-2014 the FFmpeg developers
  
    built on Mar  7 2014 19:17:59 with gcc 4.8 (Debian 4.8.2-16)
  
  ../file.mkv: No such file or directory

  It’s specifically removing the dot at the worst possible time. Any idea what’s causing this and how to fix it ?

  PS : -print0 in find has the same problem, and none of the files have newlines etc.

  Edit : As requested, the echo. The echo works as expected :

  find -name "*.mkv" | while read f
  
  do echo ffmpeg -i "$f" -c copy -y "${f%.*}.mp4"
  
  done
  
  
  
  + read f
  
  + echo ffmpeg -i './file.mkv' -c copy -y './file.mp4'
  
  ffmpeg -i ./file.mkv -c copy -y ./file.mp4

  Edit2 : With a file with a space and one without, the filename without is changed. If I rm the one with a space the one without works fine.

  $ ls
  
  + ls
  
  file.mkv  file two.mkv
  
  $ find -name "*.mkv" | while read f; do ffmpeg -i "$f" -c copy -y "${f%.*}.mp4"; done
  
  + find -name '*.mkv'
  
  + read f
  
  + ffmpeg -i './file two.mkv' -c copy -y './file two.mp4'
  
  ffmpeg version N-61155-g61ff043 Copyright (c) 2000-2014 the FFmpeg developers
  
    built on Mar  7 2014 19:17:59 with gcc 4.8 (Debian 4.8.2-16)
  
    configuration: --arch=amd64 --enable-pthreads --enable-libopencv --enable-librtmp --enable-libopenjpeg --enable-libopus --enable-libspeex --enable-libtheora --enable-vaapi --enable-runtime-cpudetect --enable-libvorbis --enable-zlib --enable-swscale --enable-libcdio --enable-bzlib --enable-libdc1394 --enable-frei0r --enable-gnutls --enable-libgsm --enable-libmp3lame --enable-libpulse --enable-vdpau --enable-libvpx --enable-gpl --enable-x11grab --enable-libx264 --enable-filters
  
    libavutil      52. 66.101 / 52. 66.101
  
    libavcodec     55. 52.102 / 55. 52.102
  
    libavformat    55. 33.101 / 55. 33.101
  
    libavdevice    55. 11.100 / 55. 11.100
  
    libavfilter     4.  3.100 /  4.  3.100
  
    libswscale      2.  5.101 /  2.  5.101
  
    libswresample   0. 18.100 /  0. 18.100
  
    libpostproc    52.  3.100 / 52.  3.100
  
  Input #0, matroska,webm, from './file two.mkv':
  
    Metadata:
  
      COMPATIBLE_BRANDS: isommp42
  
      MAJOR_BRAND     : mp42
  
      MINOR_VERSION   : 0
  
      ENCODER         : Lavf55.33.101
  
    Duration: 00:08:15.54, start: 0.000000, bitrate: 194 kb/s
  
      Stream #0:0(und): Audio: aac, 44100 Hz, stereo, fltp (default)
  
      Metadata:
  
        CREATION_TIME   : 2014-03-07 04:33:17
  
        LANGUAGE        : und
  
        HANDLER_NAME    : IsoMedia File Produced by Google, 5-11-2011
  
  Output #0, mp4, to './file two.mp4':
  
    Metadata:
  
      COMPATIBLE_BRANDS: isommp42
  
      MAJOR_BRAND     : mp42
  
      MINOR_VERSION   : 0
  
      encoder         : Lavf55.33.101
  
      Stream #0:0(und): Audio: aac ([64][0][0][0] / 0x0040), 44100 Hz, stereo (default)
  
      Metadata:
  
        CREATION_TIME   : 2014-03-07 04:33:17
  
        LANGUAGE        : und
  
        HANDLER_NAME    : IsoMedia File Produced by Google, 5-11-2011
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)
  
  Press [q] to stop, [?] for help
  
  size=   11698kB time=00:08:15.53 bitrate= 193.4kbits/s    
  
  video:0kB audio:11614kB subtitle:0 data:0 global headers:0kB muxing overhead 0.724668%
  
  + read f
  
  + ffmpeg -i /file.mkv -c copy -y /file.mp4
  
  ffmpeg version N-61155-g61ff043 Copyright (c) 2000-2014 the FFmpeg developers
  
    built on Mar  7 2014 19:17:59 with gcc 4.8 (Debian 4.8.2-16)
  
    configuration: --arch=amd64 --enable-pthreads --enable-libopencv --enable-librtmp --enable-libopenjpeg --enable-libopus --enable-libspeex --enable-libtheora --enable-vaapi --enable-runtime-cpudetect --enable-libvorbis --enable-zlib --enable-swscale --enable-libcdio --enable-bzlib --enable-libdc1394 --enable-frei0r --enable-gnutls --enable-libgsm --enable-libmp3lame --enable-libpulse --enable-vdpau --enable-libvpx --enable-gpl --enable-x11grab --enable-libx264 --enable-filters
  
    libavutil      52. 66.101 / 52. 66.101
  
    libavcodec     55. 52.102 / 55. 52.102
  
    libavformat    55. 33.101 / 55. 33.101
  
    libavdevice    55. 11.100 / 55. 11.100
  
    libavfilter     4.  3.100 /  4.  3.100
  
    libswscale      2.  5.101 /  2.  5.101
  
    libswresample   0. 18.100 /  0. 18.100
  
    libpostproc    52.  3.100 / 52.  3.100
  
  /file.mkv: No such file or directory
  
  + read f

• change wav sample rate from 22khz to 44khz with ffmpeg [on hold]

  17 mars 2014, par Marc Maxson

  I have wav files that the web browser has recorded using recorder.js at 22.05khz (in order to limit filesize). On the server, I'm using ffmpeg to convert them to 44.1khz MP3 files.

  ffmpeg has pretty good documentation, and I think this ought to do what I want :

  ffmpeg -i ~/webapps/s/audiostory/absquatulate-ko6vc-2014-03-14.wav -ar 44100 test.mp3

  but the file still sounds slow. I probed :

  > ffprobe /home/djotjog/webapps/s/audiostory/absquatulate-ko6vc-2014-03-14.wav
  
  Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 22050 Hz, 2 channels, s16, 705 kb/s
  
  > ffprobe test.mp3
  
  Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16, 128 kb/s

  What am I doing wrong ? I've tried a bunch of other tricks, including :

  ffmpeg -i /webapps/s/audiostory/absquatulate-ko6vc-2014-03-14.wav -ar 44100 -filter:a "aresample=44100" test2.wav
  (should resample the file)

  and

  ffmpeg -i /webapps/s/audiostory/absquatulate-ko6vc-2014-03-14.wav -ar 44100 -filter:a "atempo=2.0" aresample=44100 test2.wav
  (should double the tempo of the file)

  but they had no effect either.

• FFmpeg/Libav audio decode example

  19 mai 2015, par Walid Baccari

  I am trying to get the same output as that of ffmpeg/avconv from converting an MP2 file to raw PCM using code. I used the audio_decode_example function from the tutorial included in the FFmpeg/Libav documentation but the outputs were different.

  What steps does the ffmpeg/avconv command perform that I need to add to my program to get the same output ?

  Edit :
  This is the output from avconv -v debug -i test.mp2 test.sw :

  avconv version v10_beta1-302-g606cc8a, Copyright (c) 2000-2014 the Libav developers
  
  built on Mar 20 2014 20:28:32 with gcc 4.8 (Ubuntu/Linaro 4.8.1-10ubuntu9)
  
  configuration: 
  
  libavutil     53.  6. 0 / 53.  6. 0
  
  libavcodec    55. 34. 1 / 55. 34. 1
  
  libavformat   55. 12. 0 / 55. 12. 0
  
  libavdevice   54.  0. 0 / 54.  0. 0
  
  libavfilter    4.  3. 0 /  4.  3. 0
  
  libavresample  1.  2. 0 /  1.  2. 0
  
  libswscale     2.  1. 2 /  2.  1. 2
  
  Splitting the commandline.
  
  Reading option '-v' ... matched as option 'v' (set libav* logging level) with argument 'debug'.
  
  Reading option '-i' ... matched as input file with argument 'test.mp2'.
  
  Reading option 'test1.sw' ... matched as output file.                                  
  
  Finished splitting the commandline.                                       
  
  Parsing a group of options: global .                
  
  Applying option v (set libav* logging level) with argument debug.
  
  Successfully parsed a group of options.
  
  Parsing a group of options: input file test.mp2.
  
  Successfully parsed a group of options.
  
  Opening an input file: test.mp2.
  
  [mp3 @ 0x27e5060] Probed with size=2048 and score=51
  
  [mp3 @ 0x27e5060] Estimating duration from bitrate, this may be inaccurate
  
  Guessed Channel Layout for  Input Stream #0.0 : stereo
  
  Input #0, mp3, from 'test.mp2':
  
    Duration: 00:00:04.80, start: 0.000000, bitrate: 64 kb/s
  
      Stream #0.0, 200, 1/14112000: Audio: mp2, 48000 Hz, stereo, s16p, 64 kb/s
  
  Successfully opened the file.
  
  Parsing a group of options: output file test1.sw.
  
  Successfully parsed a group of options.
  
  Opening an output file: test1.sw.
  
  Successfully opened the file.
  
  Detected 2 logical cores.
  
  [abuffer @ 0x27e5860] tb:1/48000 samplefmt:s16p samplerate: 48000 ch layout:0x3
  
  [abuffersink @ 0x2807040] auto-inserting filter 'auto-inserted fifo 0' between the filter 'audio format for output stream 0:0' and the filter 'output stream 0:0'
  
  [aformat @ 0x2806bc0] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed filter 0 anull' and the filter 'audio format for output stream 0:0'
  
  [AVAudioResampleContext @ 0x28092a0] audio_convert: found function: s16p to s16  (C)
  
  [AVAudioResampleContext @ 0x28092a0] audio_convert: found function: s16p to s16  (SSE2)
  
  [resample @ 0x2808560] fmt:s16p srate:48000 cl:stereo -> fmt:s16 srate:48000 cl:stereo
  
  Output #0, s16le, to 'test1.sw':
  
    Metadata:
  
      encoder         : Lavf55.12.0
  
      Stream #0.0, 0, 1/90000: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
  
    Stream mapping:
  
      Stream #0:0 -> #0:0 (mp2 -> pcm_s16le)
  
  Press ctrl-c to stop encoding
  
  Input stream #0:0 frame changed from rate:48000 fmt:s16p ch:2 chl:stereo to rate:48000 fmt:s16 ch:2 chl:stereo
  
  Detected 2 logical cores.
  
  [abuffer @ 0x2806e40] tb:1/48000 samplefmt:s16 samplerate: 48000 ch layout:0x3
  
  [abuffersink @ 0x2806fa0] auto-inserting filter 'auto-inserted fifo 0' between the filter 'audio format for output stream 0:0' and the filter 'output stream 0:0'
  
  No more output streams to write to, finishing.
  
  size=     900kB time=4.80 bitrate=1536.0kbits/s    
  
  video:0kB audio:0kB global headers:0kB muxing overhead 0.000000%