git-svn-id: file:///var/local/repositories/ffmpeg/trunk@2978 9553f0bf-9b14-0410-a0b8-cfaf0461ba5b

git-svn-id: file:///var/local/repositories/ffmpeg/trunk@2977 9553f0bf-9b14-0410-a0b8-cfaf0461ba5b

git-svn-id: file:///var/local/repositories/ffmpeg/trunk@2976 9553f0bf-9b14-0410-a0b8-cfaf0461ba5b

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git-svn-id: file:///var/local/repositories/ffmpeg/trunk@2973 9553f0bf-9b14-0410-a0b8-cfaf0461ba5b

10l (updating LRU delta pts for type 1/2 frames)
ensure that checksumming isnt enabled if its not needed
search for next startcode and then search backward to the next valid frame if an inconsistancy is detected
deal with non-frame startcodes in the middle of the stream


git-svn-id: file:///var/local/repositories/ffmpeg/trunk@2972 9553f0bf-9b14-0410-a0b8-cfaf0461ba5b

   * making it possible to specify recording date and time in a stream


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git-svn-id: file:///var/local/repositories/ffmpeg/trunk@2962 9553f0bf-9b14-0410-a0b8-cfaf0461ba5b

some random id for vorbis so we can do some experiments with vorbis in various containers, anyone knows if vorbis in WAV/AVI has a official id?


git-svn-id: file:///var/local/repositories/ffmpeg/trunk@2961 9553f0bf-9b14-0410-a0b8-cfaf0461ba5b

use 3 pts predictors for vorbis, slightly more efficient


git-svn-id: file:///var/local/repositories/ffmpeg/trunk@2960 9553f0bf-9b14-0410-a0b8-cfaf0461ba5b

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git-svn-id: file:///var/local/repositories/ffmpeg/trunk@2953 9553f0bf-9b14-0410-a0b8-cfaf0461ba5b

fix vorbis in ogg decoding


git-svn-id: file:///var/local/repositories/ffmpeg/trunk@2952 9553f0bf-9b14-0410-a0b8-cfaf0461ba5b

 conform to latest spec
 remove get/put_s() not needed anymore
 update_packetheader() fixed (wasnt always working)
 store/load extradata & aspect ratio
 memleak fix
 dont use vorbis by default yet
 use frame_size from the codec instead of hardcoding it in the source


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split stream into valid mp3 frames, at least flv & nut absolutely need this, but probably most other formats too


git-svn-id: file:///var/local/repositories/ffmpeg/trunk@2942 9553f0bf-9b14-0410-a0b8-cfaf0461ba5b

   * .mov files with uncompressed audio can't be correctly processed
      because of the sample_size == 1 and MINOLTA hack relying on
      the information. So in a way, it's a hack of a hack.

      btw, if somebody knows why in the world even Apple's software
      thinks that for PCM 16bit sample_size == 1 please let me know.
      It clearly isn't documented that way.


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