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• FFmpgeg and raw file data input

  6 avril 2014, par fitsum

  My goal is to create "glitch" art by using image files as input and audio processing applications as conversion tool.

  I'm able to import uncompressed image files into Audacity, apply effects and export the result to an uncompressed image format. Is it possible for ffmpeg to, I guess, generate an audio stream from raw data of uncompressed input ?

• C++ FFMPEG not writing AVCC box information

  5 février 2017, par awr

  I’m trying to encode raw H264 into an mp4 container using the FFMPEG API in C++. It all works fine, however the AVCC box is empty, and it returns the error :
  [iso file] Box "avcC" size 8 invalid

  If I then use the command line tool on the output file :
  ffmpeg -i output.mp4 -vcodec copy fixed.mp4

  The output file works and AVCC is populated with the required information. I’m at a loss as to why this command line argument works but I’m unable to produce the same result using the API.

  What I do in the C++ code (also do things in between the function calls) :

  outputFormat_ = av_guess_format( "mp4", NULL, NULL ); //AV_CODEC_H264
  
  formatContext_ = avformat_alloc_context();
  
  formatContext_->oformat = outputFormat_;
  
  ...
  
  AVDictionary *opts = NULL;
  
  char tmpstr[50]; sprintf(tmpstr, "%i", muxRate * KILOBYTESTOBYTES);
  
  av_dict_set(&opts, "muxrate", tmpstr, 0);
  
  avformat_write_header( formatContext_, &opts);
  
  av_write_trailer(formatContext_);

  The output of this is correct, except it’s missing the AVCC information. Adding this is manually (and fixing the box lengths accordingly) lets me playback the video fine. Any idea why the API calls are not generating the AVCC info ?

  For reference, here’s the chars from the mp4 before the fix :
  .avc1.........................€.8.H...H..........................................ÿÿ....avcC....stts

  and after :
  avc1.........................€.8.H...H..........................................ÿÿ...!avcC.B€(ÿá..gB€(Ú.à.—•...hÎ<€....stts

• change wav, aiff or mov audio sample rate of MOV or WAV WITHOUT changing number of samples

  6 mars 2013, par John Pilgrim

  I need a very precise way to speed up audio.
  I am preparing films for OpenDCP, an open-source tool to make Digital Cinema Packages, for screening in theaters.
  My source files are usually quicktime MOV files at 23.976fps with 48.000kHz audio.
  Sometimes my audio is a separate 48.000kHz WAV.
  (FWIW, the video frame rate of the source is actually 24/100.1 frames per second, which is a repeating decimal.)

  The DCP standard is based around a 24.000fps and 48.000kHz program, so the audio and video of the source need to be sped up.
  The image processing workflow inherently involves converting the MOV to a TIF sequence, frame-per-frame, which is then assumed to be 24.000fps, so I don't have to get involved in the internals of the QT Video Media Handler.

  But speeding up the audio to match is proving to be difficult. Most audio programs cannot get the number of audio samples to line up with the retimed image frames. A 0.1% speed increase in Audacity results in the wrong number of samples. The only pathway that I have found that works is to use Apple Cinema Tools to conform the 23.976fps/48.000kHz MOV to 24.000fps/48.048kHz (which it does by changing the Quicktime headers) and then using Quicktime Player to export the audio from that file at 48.000kHz, resampling it. This is frame accurate.

  So my question is : are there settings in ffmpeg or sox that will precisely speed up the audio in a MOV or in a WAV or AIFF precisely ? I would like a cross platform solution, so I am not dependent on Cinema Tools, which is only MacOS.

  I know this is a LOT of background. Feel free to ask clarifying questions !