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• avdevice/decklink : move general code of decklink encoder to common file

  22 septembre 2014, par Deti Fliegl

  avdevice/decklink : move general code of decklink encoder to common file
  Signed-off-by : Michael Niedermayer <michaelni@gmx.at>
  

  • [DH] libavdevice/Makefile
  • [DH] libavdevice/decklink_common.cpp
  • [DH] libavdevice/decklink_common.h
  • [DH] libavdevice/decklink_common_c.h
  • [DH] libavdevice/decklink_enc.cpp
  • [DH] libavdevice/decklink_enc.h
  • [DH] libavdevice/decklink_enc_c.c

• ffmpef cannot open a simple microsoft wav file exported with Audacity

  23 juillet 2013, par sebpiq

  I have exported a sound file to microsoft wav using Audacity.
  I am trying to open this file with ffmpeg :

  ffmpeg -i steps-stereo-16b-44khz.wav /tmp/test.ogg

  and here's the ouput I get :

  fmpeg version 1.2.1 Copyright (c) 2000-2013 the FFmpeg developers
  
    built on Jun 12 2013 13:46:11 with Apple clang version 4.1 (tags/Apple/clang-421.11.66) (based on LLVM 3.1svn)
  
    configuration: --prefix=/opt/local --enable-swscale --enable-avfilter --enable-libmp3lame --enable-libvorbis --enable-libopus --enable-libtheora --enable-libschroedinger --enable-libopenjpeg --enable-libmodplug --enable-libvpx --enable-libspeex --enable-libass --enable-libbluray --enable-gnutls --enable-libfreetype --mandir=/opt/local/share/man --enable-shared --enable-pthreads --cc=/usr/bin/clang --arch=x86_64 --enable-yasm --enable-gpl --enable-postproc --enable-libx264 --enable-libxvid
  
    libavutil      52. 18.100 / 52. 18.100
  
    libavcodec     54. 92.100 / 54. 92.100
  
    libavformat    54. 63.104 / 54. 63.104
  
    libavdevice    54.  3.103 / 54.  3.103
  
    libavfilter     3. 42.103 /  3. 42.103
  
    libswscale      2.  2.100 /  2.  2.100
  
    libswresample   0. 17.102 /  0. 17.102
  
    libpostproc    52.  2.100 / 52.  2.100
  
  [dca @ 0x7fd30c013600] Not a valid DCA frame
  
  
  
  ... SNIP ...
  
  
  
  [dca @ 0x7fd5bc013600] Invalid bit allocation index
  
  [dca @ 0x7fd5bc013600] error decoding block
  
      Last message repeated 3 times
  
  [dca @ 0x7fd5bc013600] Didn't get subframe DSYNC
  
  [dca @ 0x7fd5bc013600] error decoding block
  
  [wav @ 0x7fd5bc013000] max_analyze_duration 5000000 reached at 5009070 microseconds
  
  [wav @ 0x7fd5bc013000] decoding for stream 0 failed
  
  [wav @ 0x7fd5bc013000] Could not find codec parameters for stream 0 (Audio: dts ([1][0][0][0] / 0x0001), 192000 Hz, 2 channels, fltp, 0 kb/s): no decodable DTS frames
  
  Consider increasing the value for the 'analyzeduration' and 'probesize' options
  
  steps-stereo-16b-44khz.wav: could not find codec parameters

  If I export the same file to .ogg or .aiff, no problem, the following works fine :

  ffmpeg -i steps-stereo-16b-44khz.aiff /tmp/test.ogg

  Any idea what could be wrong ?

  A link to my wav file so you can try to reproduce.

  NB my final goal is to slice the audio file. I know I can export file directly to .ogg with audacity. This is just a test case.

• ffprobe Input buffer exhausted before END element found

  9 juillet 2021, par puneet jain

  While fetching the stream details with ffprobe getting Input buffer exhausted before END element found saw few other similar errors reported but the solution was more of to fix it using the audio separately and kind of recompiling it.
How can I handle it without adding a new compilation ?

  


  ffprobe -loglevel error -select_streams v:0 -show_entries stream=width,height:stream_tags=rotate  -of json testVideo.mp4

{
[aac @ 0x7fc06201cc00] Input buffer exhausted before END element found
    "programs": [

    ],
    "streams": [
        {
            "width": 640,
            "height": 640,
            "tags": {

            }
        }
    ]
}