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• Renaming Carrierwave extension before saving it [on hold]

  2 septembre 2013, par Jan

  I am trying to get a mp3 version of a wav which gets watermarked by using ffmpeg.

  Spoken in steps :

  • upload wav (works)
  • watermark it (works also)
  • make a mp3 version (does not work)

  The upload and watermarking step is done correctly.

  My problem is, that ffmpeg evaluates the destination format by reading its file extension name (which is wav in my case). How can I rename the extension before it get's saved ?

  ffmpeg -i watermarked.wav -acodec libmp3lame -f mp3 watermarked.wav
  
                       HOW CAN I RENAME THIS BEFORE IT GET SAVED? ^^^

  The above snip (-f forcing the codec and format) does NOT it's job and

  def full_filename(for_file=file)
  
    super.chomp(File.extname(super)) + '.mp3'
  
  end 

  is happening too late (after processing)

  Do I have to make a seperate (second) tempfile and remove the watermarked.wav ?
  Or do I have to make a "seperate process" and take ?
  And when yes, how ?

  I am trying this since weeks...

  many, many, many thanks in advance

• How to detect audio sampling rate with avprobe / ffprobe ?

  8 août 2013, par Devy

  I am using libav 9.6, installed via Homebrew.

  $ avprobe -version
  
  avprobe version 9.6, Copyright (c) 2007-2013 the Libav developers
  
    built on Jun  8 2013 02:44:19 with Apple LLVM version 4.2 (clang-425.0.24) (based on LLVM 3.2svn)
  
  avprobe 9.6
  
  libavutil     52.  3. 0 / 52.  3. 0
  
  libavcodec    54. 35. 0 / 54. 35. 0
  
  libavformat   54. 20. 3 / 54. 20. 3
  
  libavdevice   53.  2. 0 / 53.  2. 0
  
  libavfilter    3.  3. 0 /  3.  3. 0
  
  libavresample  1.  0. 1 /  1.  0. 1
  
  libswscale     2.  1. 1 /  2.  1. 1

  Even though the sampling rate is displayed in the stdout in the command line output, the -show_format option doesn't surface the sampling rate information for the audio file at all.

  Here is the BASH terminal output :

  $ avprobe  -v verbose -show_format -of json  sample.gsm
  
  avprobe version 9.6, Copyright (c) 2007-2013 the Libav developers
  
    built on Jun  8 2013 02:44:19 with Apple LLVM version 4.2 (clang-425.0.24)
  
  (based on LLVM 3.2svn)
  
    configuration: --prefix=/usr/local/Cellar/libav/9.6 --enable-shared
  
  --enable-pthreads --enable-gpl --enable-version3 --enable-nonfree
  
  --enable-hardcoded-tables --enable-avresample --enable-vda --enable-gnutls
  
  --enable-runtime-cpudetect --disable-indev=jack --cc=cc --host-cflags=
  
  --host-ldflags= --enable-libx264 --enable-libfaac --enable-libmp3lame
  
  --enable-libxvid --enable-avplay
  
    libavutil     52.  3. 0 / 52.  3. 0
  
    libavcodec    54. 35. 0 / 54. 35. 0
  
    libavformat   54. 20. 3 / 54. 20. 3
  
    libavdevice   53.  2. 0 / 53.  2. 0
  
    libavfilter    3.  3. 0 /  3.  3. 0
  
    libavresample  1.  0. 1 /  1.  0. 1
  
    libswscale     2.  1. 1 /  2.  1. 1
  
  [gsm @ 0x7f8012806600] Estimating duration from bitrate, this may be inaccurate
  
  Input #0, gsm, from 'sample.gsm':
  
    Duration: 00:03:52.32, start: 0.000000, bitrate: 13 kb/s
  
      Stream #0.0: Audio: gsm, 8000 Hz, mono, s16, 13 kb/s
  
  {  "format" : {
  
      "filename" : "sample.gsm",
  
      "nb_streams" : 1,
  
      "format_name" : "gsm",
  
      "format_long_name" : "raw GSM",
  
      "start_time" : "0.000000",
  
      "duration" : "232.320000",
  
      "size" : "383328.000000",
  
      "bit_rate" : "13200.000000"
  
    }}

  And the python code example :

  >>> filename = 'sample.gsm'
  
  >>> result = subprocess.check_output(['avprobe', '-show_format', '-of',
  
  'json', filename])
  
  avprobe version 9.6, Copyright (c) 2007-2013 the Libav developers
  
    built on Jun  8 2013 02:44:19 with Apple LLVM version 4.2
  
  (clang-425.0.24) (based on LLVM 3.2svn)
  
  [gsm @ 0x7fe0b1806600] Estimating duration from bitrate, this may be
  
  inaccurate
  
  Input #0, gsm, from 'sample.gsm':
  
    Duration: 00:03:52.32, start: 0.000000, bitrate: 13 kb/s
  
      Stream #0.0: Audio: gsm, 8000 Hz, mono, s16, 13 kb/s
  
  >>> print result
  
  {  "format" : {
  
      "filename" : "sample.gsm",
  
      "nb_streams" : 1,
  
      "format_name" : "gsm",
  
      "format_long_name" : "raw GSM",
  
      "start_time" : "0.000000",
  
      "duration" : "232.320000",
  
      "size" : "383328.000000",
  
      "bit_rate" : "13200.000000"
  
  }}

  So I am aware that sampling rate could be a stream specific display to be shown in -show_format option results. But there isn't any other options to detect the sampling rate on a specific audio stream even though it's possible to set it with -ar when re-encoding it.

  I filed a ticket to libav but I am just curious if there is any other way to extract sampling rate from libav probing utils. I appreciate the answer beforehand.

  PS : it would be the same question for the upstream project of ffmpeg (ffprobe) in this case.

• extracting h264 raw video stream from mp4 or flv with ffmpeg generate an invalid stream

  10 octobre 2013, par neo2006

  I'm trying to extract the video stream from an mp4 or flv h264 video (youtube video) using ffmpeg. The original video (test.flv) play without trouble with ffplay , ffprobe gives an error as follow :

  ffprobe version N-55515-gbbbd959 Copyright (c) 2007-2013 the FFmpeg developers
  
  built on Aug 13 2013 18:06:32 with gcc 4.7.3 (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-libcaca --enable-libfreetyp
  
  e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --ena
  
  ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-l
  
  ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp
  
  eex --enable-libtheora --enable-libtwolame --enable-libvo-aacenc --enable-libvo-
  
  amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --
  
  enable-libxvid --enable-zlib
  
    libavutil      52. 42.100 / 52. 42.100
  
    libavcodec     55. 27.100 / 55. 27.100
  
    libavformat    55. 13.102 / 55. 13.102
  
    libavdevice    55.  3.100 / 55.  3.100
  
    libavfilter     3. 82.100 /  3. 82.100
  
    libswscale      2.  4.100 /  2.  4.100
  
    libswresample   0. 17.103 /  0. 17.103
  
    libpostproc    52.  3.100 / 52.  3.100
  
  [flv @ 000000000031ea80] Stream discovered after head already parsed
  
  Input #0, flv, from 'test.flv':
  
    Metadata:
  
      starttime       : 0
  
      totalduration   : 142
  
      totaldatarate   : 692
  
      bytelength      : 12286492
  
      canseekontime   : true
  
      sourcedata      : B42B95507HH1381414522145462
  
      purl            :
  
      pmsg            :
  
    Duration: 00:02:22.02, start: 0.000000, bitrate: 692 kb/s
  
      Stream #0:0: Video: h264 (Main), yuv420p, 640x268, 568 kb/s, 23.98 tbr, 1k t
  
  bn, 47.95 tbc
  
      Stream #0:1: Audio: aac, 44100 Hz, stereo, fltp, 131 kb/s
  
      Stream #0:2: Data: none
  
  Unsupported codec with id 0 for input stream 2

  to get rid of the extra streams ( I only needs the video) I used the following ffmpeg command line :

  ffmpeg -i test.flv -map 0:0 -vcodec copy -an -f h264 test.h264

  The new stream is unreadable by any player including ffplay and gives an error with ffprobe :
  test.h264 : Invalid data found when processing inputq= 0B f=0/0

  Any body have an idea about what am I doing wrong ?

  I also tried simpler youtube command line :

  ffmpeg -i test.flv -vcodec copy -an test.h264

  if I use another format (avi for example) :

  ffmpeg -i test.flv -vcodec copy -an test.avi

  the output video is valid.

  If I transcode the video

  ffmpeg -i test.flv -an test.h264

  the output is also valid

  Any suggestions ?