I'm trying to adapt decoder_example to to support more of an event driven
model. I'm building an RTSP client/server where the client will have a
function called everytime a new packet comes in. So, I want to queue
up these packets to be played.
My current attack is as follows (note this isn't very robust and I
realized that):
For first and second packet i grab all the header stuff in the same way
as decoder_example. From the third packet on, I send it to a bunch of
code:
float **pcm;
int samples;
buffer=ogg_sync_buffer(&oy,4096);
bytes = 4096;
buffer = memcpy (buffer, data, bytes);
ogg_sync_wrote(&oy,bytes);
ogg_sync_pageout(&oy,&og);
ogg_stream_pagein(&os,&og); /* can safely ignore errors at
this point */
/* we have a packet. Decode it */
if(vorbis_synthesis(&vb,&op)==0) /* test for success! */
vorbis_synthesis_blockin(&vd,&vb);
/*
**pcm is a multichannel float vector. In stereo, for
example, pcm[0] is left, and pcm[1] is right. samples is
the size of each channel. Convert the float values
(-1.<=range<=1.) to whatever PCM format and write it out */
while((samples=vorbis_synthesis_pcmout(&vd,&pcm))>0) {
int i,j;
int clipflag=0;
int bout=(samples<convsize?samples:convsize);
fprintf (stderr,
"\nhoebag, i made it the while loop");
/* convert floats to 16 bit signed ints (host order) and
interleave */
for(i=0;i<vi.channels;i++){
ogg_int16_t *ptr=convbuffer+i;
float *mono=pcm[i];
for(j=0;j<bout;j++){
#if 1
int val=mono[j]*32767.f;
#else /* optional dither */
int val=mono[j]*32767.f+drand48()-0.5f;
#endif
/* might as well guard against clipping */
if(val>32767){
val=32767;
clipflag=1;
}
if(val<-32768){
val=-32768;
clipflag=1;
}
*ptr=val;
ptr+=2;
}
}
if(clipflag)
fprintf(stderr,"Clipping in frame
%ld\n",(long)(vd.sequence));
write (mywav, convbuffer, 2*vi.channels*bout);
vorbis_synthesis_read(&vd,bout); /* tell libvorbis how
many samples we
actually consumed */
}
---
So, the first packets come in fine because I can retrieve the title, etc
properly. All of the packets come in correctly because I write them out
to disk and it is the same file as the original file. However, when I put
the packets through the above code, I lose a lot of information and the
mywav file is a small portion of what it should be.
If anyone has some ideas how I can solve this problem, I would appreciate
any help.
Thanks,
Wayne
--
Wayne Davis - wfdavis@seas.upenn.edu - PGP Key Available
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