[Mplayer-cvslog] CVS: main dec_audio.c,1.85,1.86
Alban Bedel CVS
albeu at mplayer.dev.hu
Sat Mar 16 15:32:05 CET 2002
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Update of /cvsroot/mplayer/main
In directory mplayer:/var/tmp.root/cvs-serv11764
Modified Files:
dec_audio.c
Log Message:
Initial ogg demuxer. No seeking, a/v sync is broken. Support avi
with ogg/vorbis audio.
Index: dec_audio.c
===================================================================
RCS file: /cvsroot/mplayer/main/dec_audio.c,v
retrieving revision 1.85
retrieving revision 1.86
diff -u -r1.85 -r1.86
--- dec_audio.c 15 Mar 2002 23:16:32 -0000 1.85
+++ dec_audio.c 16 Mar 2002 14:31:52 -0000 1.86
@@ -73,13 +73,8 @@
#include <math.h>
#include <vorbis/codec.h>
+// This struct is also defined in demux_ogg.c => common header ?
typedef struct ov_struct_st {
- ogg_sync_state oy; /* sync and verify incoming physical bitstream */
- ogg_stream_state os; /* take physical pages, weld into a logical
- stream of packets */
- ogg_page og; /* one Ogg bitstream page. Vorbis packets are inside */
- ogg_packet op; /* one raw packet of data for decode */
-
vorbis_info vi; /* struct that stores all the static vorbis bitstream
settings */
vorbis_comment vc; /* struct that stores all the bitstream user comments */
@@ -389,6 +384,7 @@
driver=0;
#else
/* OggVorbis audio via libvorbis, compatible with files created by nandub and zorannt codec */
+ // Is there always 1024 samples/frame ? ***** Albeu
sh_audio->audio_out_minsize=1024*4; // 1024 samples/frame
#endif
break;
@@ -759,124 +755,61 @@
}
#ifdef HAVE_OGGVORBIS
case AFM_VORBIS: {
- // OggVorbis Audio:
-#if 0 /* just here for reference - atmos */
- ogg_sync_state oy; /* sync and verify incoming physical bitstream */
- ogg_stream_state os; /* take physical pages, weld into a logical
- stream of packets */
- ogg_page og; /* one Ogg bitstream page. Vorbis packets are inside */
- ogg_packet op; /* one raw packet of data for decode */
-
- vorbis_info vi; /* struct that stores all the static vorbis bitstream
- settings */
- vorbis_comment vc; /* struct that stores all the bitstream user comments */
- vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */
- vorbis_block vb; /* local working space for packet->PCM decode */
-#else
- /* nix, nada, rien, nothing, nem, nüx */
-#endif
-
- uint32_t hdrsizes[3];/* stores vorbis header sizes from AVI audio header,
- maybe use ogg_uint32_t */
- //int i;
- int ret;
- char *buffer;
- ogg_packet hdr;
- //ov_struct_t *s=&sh_audio->ov;
- sh_audio->ov=malloc(sizeof(ov_struct_t));
- //s=&sh_audio->ov;
-
- vorbis_info_init(&sh_audio->ov->vi);
- vorbis_comment_init(&sh_audio->ov->vc);
-
- mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"OggVorbis: cbsize: %i\n", sh_audio->wf->cbSize);
- memcpy(hdrsizes, ((unsigned char*)sh_audio->wf)+2*sizeof(WAVEFORMATEX), 3*sizeof(uint32_t));
- mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"OggVorbis: Read header sizes: initial: %i comment: %i codebook: %i\n", hdrsizes[0], hdrsizes[1], hdrsizes[2]);
- /*for(i=12; i <= 40; i+=2) { // header bruteforce :)
- memcpy(hdrsizes, ((unsigned char*)sh_audio->wf)+i, 3*sizeof(uint32_t));
- printf("OggVorbis: Read header sizes (%i): %ld %ld %ld\n", i, hdrsizes[0], hdrsizes[1], hdrsizes[2]);
- }*/
-
- /* read headers */ // FIXME disable sound on errors here, we absolutely need this headers! - atmos
- hdr.packet=NULL;
- hdr.b_o_s = 1; /* beginning of stream for first packet */
- hdr.bytes = hdrsizes[0];
- hdr.packet = realloc(hdr.packet,hdr.bytes);
- memcpy(hdr.packet,((unsigned char*)sh_audio->wf)+2*sizeof(WAVEFORMATEX)+3*sizeof(uint32_t),hdr.bytes);
- if(vorbis_synthesis_headerin(&sh_audio->ov->vi,&sh_audio->ov->vc,&hdr)<0)
- mp_msg(MSGT_DECAUDIO,MSGL_WARN,"OggVorbis: initial (identification) header broken!\n");
- hdr.b_o_s = 0;
- hdr.bytes = hdrsizes[1];
- hdr.packet = realloc(hdr.packet,hdr.bytes);
- memcpy(hdr.packet,((unsigned char*)sh_audio->wf)+2*sizeof(WAVEFORMATEX)+3*sizeof(uint32_t)+hdrsizes[0],hdr.bytes);
- if(vorbis_synthesis_headerin(&sh_audio->ov->vi,&sh_audio->ov->vc,&hdr)<0)
- mp_msg(MSGT_DECAUDIO,MSGL_WARN,"OggVorbis: comment header broken!\n");
- hdr.bytes = hdrsizes[2];
- hdr.packet = realloc(hdr.packet,hdr.bytes);
- memcpy(hdr.packet,((unsigned char*)sh_audio->wf)+2*sizeof(WAVEFORMATEX)+3*sizeof(uint32_t)+hdrsizes[0]+hdrsizes[1],hdr.bytes);
- if(vorbis_synthesis_headerin(&sh_audio->ov->vi,&sh_audio->ov->vc,&hdr)<0)
+ ogg_packet op;
+ vorbis_comment vc;
+ struct ov_struct_st *ov;
+
+ /// Init the decoder with the 3 header packets
+ ov = (struct ov_struct_st*)malloc(sizeof(struct ov_struct_st));
+ vorbis_info_init(&ov->vi);
+ vorbis_comment_init(&vc);
+ op.bytes = ds_get_packet(sh_audio->ds,&op.packet);
+ op.b_o_s = 1;
+ /// Header
+ if(vorbis_synthesis_headerin(&ov->vi,&vc,&op) <0) {
+ mp_msg(MSGT_DECAUDIO,MSGL_ERR,"OggVorbis: initial (identification) header broken!\n");
+ driver = 0;
+ free(ov);
+ break;
+ }
+ op.bytes = ds_get_packet(sh_audio->ds,&op.packet);
+ op.b_o_s = 0;
+ /// Comments
+ if(vorbis_synthesis_headerin(&ov->vi,&vc,&op) <0) {
+ mp_msg(MSGT_DECAUDIO,MSGL_ERR,"OggVorbis: comment header broken!\n");
+ driver = 0;
+ free(ov);
+ break;
+ }
+ op.bytes = ds_get_packet(sh_audio->ds,&op.packet);
+ //// Codebook
+ if(vorbis_synthesis_headerin(&ov->vi,&vc,&op)<0) {
mp_msg(MSGT_DECAUDIO,MSGL_WARN,"OggVorbis: codebook header broken!\n");
- hdr.bytes=0;
- hdr.packet = realloc(hdr.packet,hdr.bytes); /* free */
- /* done with the headers */
-
-
- /* Throw the comments plus a few lines about the bitstream we're
- decoding */
- {
- char **ptr=sh_audio->ov->vc.user_comments;
+ driver = 0;
+ free(ov);
+ break;
+ } else { /// Print the infos
+ char **ptr=vc.user_comments;
while(*ptr){
mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbisComment: %s\n",*ptr);
++ptr;
}
- mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Bitstream is %d channel, %ldHz, %ldkbit/s %cBR\n",sh_audio->ov->vi.channels,sh_audio->ov->vi.rate,sh_audio->ov->vi.bitrate_nominal/1000, (sh_audio->ov->vi.bitrate_lower!=sh_audio->ov->vi.bitrate_nominal)||(sh_audio->ov->vi.bitrate_upper!=sh_audio->ov->vi.bitrate_nominal)?'V':'C');
- mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Encoded by: %s\n",sh_audio->ov->vc.vendor);
+ mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Bitstream is %d channel, %ldHz, %ldkbit/s %cBR\n",ov->vi.channels,ov->vi.rate,ov->vi.bitrate_nominal/1000, (ov->vi.bitrate_lower!=ov->vi.bitrate_nominal)||(ov->vi.bitrate_upper!=ov->vi.bitrate_nominal)?'V':'C');
+ mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Encoded by: %s\n",vc.vendor);
}
- sh_audio->channels=sh_audio->ov->vi.channels;
- sh_audio->samplerate=sh_audio->ov->vi.rate;
- sh_audio->i_bps=sh_audio->ov->vi.bitrate_nominal/8;
-
-// printf("[\n");
-// sh_audio->a_buffer_len=sh_audio->audio_out_minsize;///ov->vi.channels;
-// printf("]\n");
-
- /* OK, got and parsed all three headers. Initialize the Vorbis
- packet->PCM decoder. */
- vorbis_synthesis_init(&sh_audio->ov->vd,&sh_audio->ov->vi); /* central decode state */
- vorbis_block_init(&sh_audio->ov->vd,&sh_audio->ov->vb); /* local state for most of the decode
- so multiple block decodes can
- proceed in parallel. We could init
- multiple vorbis_block structures
- for vd here */
- //printf("OggVorbis: synthesis and block init done.\n");
- ogg_sync_init(&sh_audio->ov->oy); /* Now we can read pages */
-
- while((ret = ogg_sync_pageout(&sh_audio->ov->oy,&sh_audio->ov->og))!=1) {
- if(ret == -1)
- mp_msg(MSGT_DECAUDIO,MSGL_WARN,"OggVorbis: Pageout: not properly synced, had to skip some bytes.\n");
- else
- if(ret == 0) {
- mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Pageout: need more data to verify page, reading more data.\n");
- /* submit a a_buffer_len block to libvorbis' Ogg layer */
- buffer=ogg_sync_buffer(&sh_audio->ov->oy,256);
- ogg_sync_wrote(&sh_audio->ov->oy,demux_read_data(sh_audio->ds,buffer,256));
- }
- }
- mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Pageout: successfull.\n");
- ogg_stream_pagein(&sh_audio->ov->os,&sh_audio->ov->og); /* we can ignore any errors here
- as they'll also become apparent
- at packetout */
-
- /* Get the serial number and set up the rest of decode. */
- /* serialno first; use it to set up a logical stream */
- ogg_stream_init(&sh_audio->ov->os,ogg_page_serialno(&sh_audio->ov->og));
-
- mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Init OK!\n");
- break;
-}
+ // Setup the decoder
+ sh_audio->channels=ov->vi.channels;
+ sh_audio->samplerate=ov->vi.rate;
+ sh_audio->i_bps=ov->vi.bitrate_nominal/8;
+ sh_audio->context = ov;
+
+ /// Finish the decoder init
+ vorbis_synthesis_init(&ov->vd,&ov->vi);
+ vorbis_block_init(&ov->vd,&ov->vb);
+ mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Init OK!\n");
+} break;
#endif
-
#ifdef USE_LIBMAD
case AFM_MAD:
{
@@ -982,117 +915,64 @@
// len=MP3_DecodeFrame(buf,3);
break;
#ifdef HAVE_OGGVORBIS
- case AFM_VORBIS: { // OggVorbis
- /* note: good minlen would be 4k or 8k IMHO - atmos */
- int ret;
- char *buffer;
- int bytes;
- int samples;
- float **pcm;
- //ogg_int16_t convbuffer[4096];
-// int convsize;
- int readlen=1024;
- len=0;
-// convsize=minlen/sh_audio->ov->vi.channels;
-
- while(len < minlen) { /* double loop allows for break in inner loop */
- while(len < minlen) { /* without aborting the outer loop - atmos */
- ret=ogg_stream_packetout(&sh_audio->ov->os,&sh_audio->ov->op);
- if(ret==0) {
- int xxx=0;
- //printf("OggVorbis: Packetout: need more data, paging!\n");
- while((ret = ogg_sync_pageout(&sh_audio->ov->oy,&sh_audio->ov->og))!=1) {
- if(ret == -1)
- mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Pageout: not properly synced, had to skip some bytes.\n");
- else
- if(ret == 0) {
- //printf("OggVorbis: Pageout: need more data to verify page, reading more data.\n");
- /* submit a readlen k block to libvorbis' Ogg layer */
- buffer=ogg_sync_buffer(&sh_audio->ov->oy,readlen);
- bytes=demux_read_data(sh_audio->ds,buffer,readlen);
- xxx+=bytes;
- ogg_sync_wrote(&sh_audio->ov->oy,bytes);
- if(bytes==0)
- mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: 0Bytes written, possible End of Stream\n");
- }
- }
- mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"\n[sync: %d ]\n",xxx);
- //printf("OggVorbis: Pageout: successfull, pagin in.\n");
- if(ogg_stream_pagein(&sh_audio->ov->os,&sh_audio->ov->og)<0)
- mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Pagein failed!\n");
- break;
- } else if(ret<0) {
- mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Packetout: missing or corrupt data, skipping packet!\n");
- break;
- } else {
-
- /* we have a packet. Decode it */
-
- if(vorbis_synthesis(&sh_audio->ov->vb,&sh_audio->ov->op)==0) /* test for success! */
- vorbis_synthesis_blockin(&sh_audio->ov->vd,&sh_audio->ov->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(&sh_audio->ov->vd,&pcm))>0){
- int i,j;
- int clipflag=0;
- int convsize=(maxlen-len)/(2*sh_audio->ov->vi.channels); // max size!
- int bout=(samples<convsize?samples:convsize);
+ case AFM_VORBIS: { // Vorbis
+ int samples;
+ float **pcm;
+ ogg_packet op;
+ char* np;
+ struct ov_struct_st *ov = sh_audio->context;
+ len = 0;
+ op.b_o_s = op.e_o_s = 0;
+ while(len < minlen) {
+ op.bytes = ds_get_packet(sh_audio->ds,&op.packet);
+ if(!op.packet)
+ break;
+ if(vorbis_synthesis(&ov->vb,&op)==0) /* test for success! */
+ vorbis_synthesis_blockin(&ov->vd,&ov->vb);
+ while((samples=vorbis_synthesis_pcmout(&ov->vd,&pcm))>0){
+ int i,j;
+ int clipflag=0;
+ int convsize=(maxlen-len)/(2*ov->vi.channels); // max size!
+ int bout=(samples<convsize?samples:convsize);
- if(bout<=0) break;
+ if(bout<=0) break;
- /* convert floats to 16 bit signed ints (host order) and
- interleave */
- for(i=0;i<sh_audio->ov->vi.channels;i++){
- ogg_int16_t *convbuffer=(ogg_int16_t *)(&buf[len]);
- ogg_int16_t *ptr=convbuffer+i;
- float *mono=pcm[i];
- for(j=0;j<bout;j++){
+ /* convert floats to 16 bit signed ints (host order) and
+ interleave */
+ for(i=0;i<ov->vi.channels;i++){
+ ogg_int16_t *convbuffer=(ogg_int16_t *)(&buf[len]);
+ ogg_int16_t *ptr=convbuffer+i;
+ float *mono=pcm[i];
+ for(j=0;j<bout;j++){
#if 1
- int val=mono[j]*32767.f;
+ int val=mono[j]*32767.f;
#else /* optional dither */
- int val=mono[j]*32767.f+drand48()-0.5f;
+ int val=mono[j]*32767.f+drand48()-0.5f;
#endif
- /* might as well guard against clipping */
- if(val>32767){
- val=32767;
- clipflag=1;
+ /* might as well guard against clipping */
+ if(val>32767){
+ val=32767;
+ clipflag=1;
+ }
+ if(val<-32768){
+ val=-32768;
+ clipflag=1;
+ }
+ *ptr=val;
+ ptr+=ov->vi.channels;
}
- if(val<-32768){
- val=-32768;
- clipflag=1;
- }
- *ptr=val;
- ptr+=sh_audio->ov->vi.channels;
}
- }
- if(clipflag)
- mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"Clipping in frame %ld\n",(long)(sh_audio->ov->vd.sequence));
-
- //fwrite(convbuffer,2*sh_audio->ov->vi.channels,bout,stderr); //dump pcm to file for debugging
- //memcpy(buf+len,convbuffer,2*sh_audio->ov->vi.channels*bout);
- len+=2*sh_audio->ov->vi.channels*bout;
-
- mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"\n[decoded: %d / %d ]\n",bout,samples);
-
- vorbis_synthesis_read(&sh_audio->ov->vd,bout); /* tell libvorbis how
- many samples we
- actually consumed */
- }
- } // from else, packetout ok
- } // while len
- } // outer while len
- if(ogg_page_eos(&sh_audio->ov->og))
- mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: End of Stream reached!\n"); // FIXME clearup decoder, notify mplayer - atmos
-
- mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"\n[len: %d ]\n",len);
-
- break;
- }
+ if(clipflag)
+ mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"Clipping in frame %ld\n",(long)(ov->vd.sequence));
+ len+=2*ov->vi.channels*bout;
+ mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"\n[decoded: %d / %d ]\n",bout,samples);
+ vorbis_synthesis_read(&ov->vd,bout); /* tell libvorbis how
+ many samples we
+ actually consumed */
+ }
+ }
+ } break;
#endif
case AFM_PCM: // AVI PCM
len=demux_read_data(sh_audio->ds,buf,minlen);
@@ -1337,13 +1217,6 @@
MP3_DecodeFrame(NULL,-2); // resync
MP3_DecodeFrame(NULL,-2); // resync
break;
-#ifdef HAVE_OGGVORBIS
- case AFM_VORBIS:
- //printf("OggVorbis: resetting stream.\n");
- ogg_sync_reset(&sh_audio->ov->oy);
- ogg_stream_reset(&sh_audio->ov->os);
- break;
-#endif
#ifdef USE_LIBAC3
case AFM_AC3:
ac3_bitstream_reset(); // reset AC3 bitstream buffer
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