[FFmpeg-devel] Audio decoding in api-example.c broken?

[Martin Storsjö]

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:
> > 
> > > On Mon, May 24, 2010 at 10:42:41AM +0300, Martin Storsj? wrote:
> > > > On Mon, 24 May 2010, Michael Niedermayer wrote:
> > > > 
> > > > > a AVParser could be used or you could implement some buffer that is refilled
> > > > > after each frame is decoded
> > > > 
> > > > An attempt at solving it by refilling the buffer is attached.
> > > > 
> > > > // Martin
> > > >  api-example.c |   12 ++++++------
> > > >  1 file changed, 6 insertions(+), 6 deletions(-)
> > > > f62159abacf81c6c15a0a4f310284b9f6d2859e4  0001-Refill-the-input-audio-buffer-after-decoding-each-pa.patch
> > > > From 25f9c3322ebe7381623ddb546d61c67540267835 Mon Sep 17 00:00:00 2001
> > > > From: Martin Storsjo <martin at martin.st>
> > > > Date: Mon, 24 May 2010 10:23:25 +0300
> > > > Subject: [PATCH] Refill the input audio buffer after decoding each packet
> > > > 
> > > > This avoids trying to decode incomplete frames.
> > > > ---
> > > >  libavcodec/api-example.c |   12 ++++++------
> > > >  1 files changed, 6 insertions(+), 6 deletions(-)
> > > > 
> > > > diff --git a/libavcodec/api-example.c b/libavcodec/api-example.c
> > > > index fb48b1f..576826d 100644
> > > > --- a/libavcodec/api-example.c
> > > > +++ b/libavcodec/api-example.c
> > > > @@ -155,12 +155,8 @@ static void audio_decode_example(const char *outfilename, const char *filename)
> > > >  
> > > >      /* decode until eof */
> > > >      avpkt.data = inbuf;
> > > > -    for(;;) {
> > > >          avpkt.size = fread(inbuf, 1, INBUF_SIZE, f);
> > > > -        if (avpkt.size == 0)
> > > > -            break;
> > > >  
> > > > -        avpkt.data = inbuf;
> > > >          while (avpkt.size > 0) {
> > > >              out_size = AVCODEC_MAX_AUDIO_FRAME_SIZE;
> > > >              len = avcodec_decode_audio3(c, (short *)outbuf, &out_size, &avpkt);
> > > > @@ -173,9 +169,13 @@ static void audio_decode_example(const char *outfilename, const char *filename)
> > > >                  fwrite(outbuf, 1, out_size, outfile);
> > > >              }
> > > >              avpkt.size -= len;
> > > > -            avpkt.data += len;
> > > > +            memmove(avpkt.data, avpkt.data + len, avpkt.size);
> > > > +            /* Refill the input buffer, to avoid trying to decode
> > > > +             * incomplete frames. */
> > > > +            len = fread(avpkt.data + avpkt.size, 1, INBUF_SIZE - avpkt.size, f);
> > > > +            if (len > 0)
> > > > +                avpkt.size += len;
> > > 
> > > as this is an example people will use for their applications
> > > it should mention parsers and demuxers as alternative solutions
> > 
> > Ok, I could add comments mentioning that.
> > 
> > > 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.

> codecs have a maximum packet size in general

So, where can I get such a maximum packet size for MP2, in a generic way 
suitable for api-example.c?

// Martin