[FFmpeg-devel] [PATCH] Waveform audio grabber.

[Michael Niedermayer]

On Wed, Mar 17, 2010 at 12:50:07AM -0300, Ramiro Polla wrote:
> On Sat, Mar 13, 2010 at 6:58 AM, Reimar D?ffinger
> <Reimar.Doeffinger at gmx.de> wrote:
> > On Fri, Mar 12, 2010 at 06:08:53PM -0300, Ramiro Polla wrote:
> > I don't think adding the sample_format is particularly related to this new
> > input format and shouldn't be in the same patch.
> 
> Split into ap_sample_fmt.diff
> 
> > Also it should select a suitable format (one supported by the hardware) on
> > its own for values that are set to 0 instead of failing.
> 
> Set SAMPLE_FMT_S16 as default sample_fmt in ffmpeg.c. Tries all other
> formats that are understood by MinGW.
> 
> One thing that bothers me about this implementation is the pts. There
> is no way to get a proper pts from the MS API. What I have done here
> is to detect whenever all buffers are full (this will drop audio data)
> and use GetTickCount() to get a pts for the first frame in a sequence
> as soon as it is received. All other frames without pts following that
> one with pts are contiguous and so lavf can calculate the proper pts.
> Any other ideas?

libavformat/timefilter*

if you provide just 1 timestamp you can be sure there will e drift with
some hardware (probably all hardware people use on pcs)

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

Rewriting code that is poorly written but fully understood is good.
Rewriting code that one doesnt understand is a sign that one is less smart
then the original author, trying to rewrite it will not make it better.
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