[FFmpeg-devel] Audio decoding in api-example.c broken?
Mon May 24 16:19:40 CEST 2010
On Mon, 24 May 2010, Michael Niedermayer wrote:
> On Mon, May 24, 2010 at 03:46:08PM +0300, Martin Storsj? wrote:
> > On Mon, 24 May 2010, Michael Niedermayer wrote:
> > > On Mon, May 24, 2010 at 02:27:42PM +0300, Martin Storsj? wrote:
> > > > On Mon, 24 May 2010, Michael Niedermayer wrote:
> > > >
> > > > > and the refill must be implemented so as to minimize memmove/cpy
> > > > > moving is only needed once we get close to the end of the buffer ...
> > > >
> > > > How do we know that we're "close to the end of the buffer"?
> > >
> > > How do you know your buffer is large enough to begin with?
> > I don't, I just hope it is.
> > > > Something like
> > > > if (avpkt.size < 2*len)?
> > >
> > > if len was the last packets len then no this is not good
> > > sizes can surely vary by more
> > That's why I always refilled the buffer in the previous suggestion.
> its quite inefficient if you have a 4096 buffer and 40 byte packets
> and as this is example code such hacks are obviously something we cant
> you could allocate a 100k buffer and keep filling it so there is at least
> 4k there, and once you hit the end memmove the <=4k left to the begin
> this still is a hack without us knowing for sure 4k is enough in the
> first place but its better than what we have now
In this particular case, the data it decodes is just 41 KB so that part of
the code would never be used. But if you'd accept such a version of the
example anyway, I can try to rewrite it in that way.
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