Recherche avancée

Sur d’autres sites (8389)

• EAC3 codec not supported in WAVE

  8 janvier 2014, par Tarun

  I was trying this command and stumble upon following error-
  So my question is - Is this issue limitation of FFmpeg or limitation of WAV container itself.
  Any reference would be great

  ffmpeg -i aud_track1.mp4 -acodec eac3 out.wav
  
      ffmpeg version N-59453-gd52882f Copyright (c) 2000-2013 the FFmpeg developers
  
        built on Dec 30 2013 22:07:23 with gcc 4.8.2 (GCC)
  
        configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-li
  
      bcaca --enable-libfreetype --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enab
  
      le-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack -
  
      -enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
  
        libavutil      52. 59.100 / 52. 59.100
  
        libavcodec     55. 47.100 / 55. 47.100
  
        libavformat    55. 22.101 / 55. 22.101
  
        libavdevice    55.  5.102 / 55.  5.102
  
        libavfilter     4.  0.103 /  4.  0.103
  
        libswscale      2.  5.101 /  2.  5.101
  
        libswresample   0. 17.104 /  0. 17.104
  
        libpostproc    52.  3.100 / 52.  3.100
  
      Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'aud_track1.mp4':
  
        Metadata:
  
          major_brand     : isom
  
          minor_version   : 512
  
          compatible_brands: isomiso2mp41
  
          encoder         : Lavf54.15.102
  
        Duration: 00:01:59.64, start: 0.000000, bitrate: 129 kb/s
  
          Stream #0:0(und): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 128 kb/s (default)
  
          Metadata:
  
            handler_name    : SoundHandler
  
      File 'out.wav' already exists. Overwrite ? [y/N] y
  
      [wav @ 00000000027df6c0] eac3 codec not supported in WAVE format
  
      Output #0, wav, to 'out.wav':
  
        Metadata:
  
          major_brand     : isom
  
          minor_version   : 512
  
          compatible_brands: isomiso2mp41
  
          encoder         : Lavf55.22.101
  
          Stream #0:0(und): Audio: eac3, 48000 Hz, stereo, fltp, 192 kb/s (default)
  
          Metadata:
  
            handler_name    : SoundHandler
  
      Stream mapping:
  
        Stream #0:0 -> #0:0 (aac -> eac3)
  
      Could not write header for output file #0 (incorrect codec parameters ?): Error number -1 occurred

• Revision 507fd5220b : Enable encode_breakout in real time encoding In real time encoding, we enable e

  4 février 2014, par Yunqing Wang

  Changed Paths :
   Modify /vp9/encoder/vp9_encodeframe.c

  
   Modify /vp9/encoder/vp9_firstpass.c

  
   Modify /vp9/encoder/vp9_onyx_if.c

  
   Modify /vp9/encoder/vp9_onyx_int.h

  
   Modify /vp9/encoder/vp9_rdopt.c

  
  
  Enable encode_breakout in real time encoding

  In real time encoding, we enable encode_breakout to make encoding
  fast. A speed feature "use_encode_breakout" is defined to set
  encode_breakout thresholds for different speeds.

  However, currently, static_thresh is an encoder option. The encode_
  breakout can be turned off if user sets static_thresh=0 specifically.

  The rtc set borg test result : (need to set —static_thresh=1)
  speed -5, psnr loss -3.543% ;
  speed -4, psnr loss -2.358% ;
  speed -3, psnr loss -0.771%.

  Encoding speed test :
  speed -5, 11% - 60% speedup ;
  speed -4, 5.5% - 28% speedup ;
  speed -3, 0.8% - 7% speedup.

  Change-Id : Icde592ffbe77eac7446f872a2e9eb2051733677b

• Segment Broadcast WAVE file with FFMPEG

  4 octobre 2018, par yekootmada

  I have a Broadcast WAV file that is duration 4 hours with a timecode track starting at 00:00:00 and ending at 04:00:00 (obviously).

  I need to extract a section of the file, for example from 01:45:00 to 03:55:00 whilst maintaining the original timecode data for that section of the file.

  When i run this FFMPEG script, I get the correction section of the file in "out.wav", however the start timecode of the file is always 00:00:00, and not 01:45:00 as you would expect.

  Here is my current code :

  ffmpeg -ss 01:45:00 -i in.wav -to 03:55:00 -c copy -write_bext 1 out.wav

  How do you tell ffmpeg to take the timecode data from the section of the file and not copy it from the beginning ?

  Thanks a lot, hope that is clear.