[FFmpeg-devel] [RFC] libavfilter audio API and related issues

[Michael Niedermayer]

On Wed, Jun 16, 2010 at 01:00:26AM +0200, Stefano Sabatini wrote:
> On date Sunday 2010-06-13 19:20:55 -0700, S.N. Hemanth Meenakshisundaram encoded:
> > 
> > On 06/05/2010 01:24 PM, S.N. Hemanth Meenakshisundaram wrote:
> > >
> > >>>>>On 05/02/2010 12:08 PM, Stefano Sabatini wrote:
> > >>>>>>On date Wednesday 2010-04-28 07:07:54 +0000, S.N. Hemanth
> > >>>>>>Meenakshisundaram encoded:
> > >>>>>>
> > >>>>>>>Stefano Sabatini<stefano.sabatini-lala<at>   poste.it>   writes:
> > >>>>>>>
> > >>>>>>>>Follow some notes about a possible design for the audio support in
> > >>>>>>>>libavfilter.
> > >>>>>>>>[...]
> > >Hi All,
> > >
> > >Attached is a rough draft of the audio API. [...]
> > 
> > Hi All,
> > 
> > I now have a working audio filter framework. Made some changes to
> > the audio API from last week and the earlier ffplay changes based on
> > comments and some problems encountered along the way. Now when
> > playing a video/audio file, the input and output filters are created
> > and the audio data passes through these two filters and plays out.
> > The video filters work without any breakages.
> > 
> > Here are the changes and comments incorporated:
> > 
> > 1. AVFilterSamplesRef now has a pts field as suggested. I can't find
> > a pos field for audio in existing code so I haven't added that yet.
> 
> Maybe pts should be moved to AVFilterBuffer, same for pos, see also
> the Michael comment.

pts can be changed by a filter so one buffer can depending on filter
have more than one pts

[...]
-- 
Michael     GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB

Old school: Use the lowest level language in which you can solve the problem
            conveniently.
New school: Use the highest level language in which the latest supercomputer
            can solve the problem without the user falling asleep waiting.
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