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• Compress video with ffmpeg programatically - i.e. using C libraries instead of command line or wrapper

  29 décembre 2014, par Mick

  I am trying to find some examples or guidelines that help show how to compress a video programatically using ffmpeg API’s (i.e. the C API’s not the command line).

  In other words I would like to achieve the equivalent of the following ffmpeg command line :

  ffmpeg -i inoutVideo.mp4 -strict experimental -acodec aac CompressedVideo_aac.mp4

  but using the appropriate API calls from a C program.

  I can already do this by creating a wrapper around the ’ffmpeg.c’ program but I am trying to understand how to achieve this with the C libraries directly.

  I realise I can simply(!) read the source code of ffmpeg.c to see what it does, but I was hoping for something that either provides and overview or a specific example for video compression.

  Any pointers gratefully appreciated.

• cocoa ffmpeg seamless loop video

  13 décembre 2012, par Marius

  OK i've seen many people having issues with getting videos to loop seamlessly on iOS. If it's a video that you provide with your app (in other words if you have control over the file itself) - there's a way of getting around the problem - by providing the video as self-contained looped MOV file. And it works like a charm. The way you do it is :

  • open an MP4 video in QT (QT7 works fine)
  • re-save it as self-contained movie
  • CMD+A, CMD+C and then CMD+V a bunch of times
  • save

  This will create a slightly bigger file than the original.

  So now i've just gotten curious : could the same thing be done in ffmpeg ? If so - how would one take an MP4 video, save it as self contained MOV and loop it for, let's say, around 60 minutes, without actually cloning video data and increasing file size just slightly ?

  Thanks for your thoughts.

• Something missing during transcoding audio-only pieces from FLV format to AAC format ?

  21 octobre 2012, par MaiTiano

  I have ten consecutive flv pieces cut from a song in flv format. Each flv piece has about 10 seconds long.

  After I transcode these flv format audio piece into wav format by ffmpeg, I drag the new 10 wav file into foobar player and play them automatically(one-by-one played automatically). I found there is no "broken/pause" feeling in during switching from the end part of previous wav file to the start part of the next wav file.

  However, after I transcode these flv format audio piece into AAC format by ffmpeg, and do the similar listening test in the foobar player, I can hear/feel obvious breakpoint when file changing.

  Is it possible for ffmpeg to miss some audio signal during flv->aac transcoding ? Because, the transcoded aac files are supposed to be played one-by-one contiguously just like there is one file playing, in other words, there should not be the breakpoint feeling happened between two consecutive file playing !

  Any advices ? Many thanks.