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Patch #1235763 by Ryan Jackson - Resume/bookmark support for Vorbis

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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7098 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Linus Nielsen Feltzing 2005-07-11 06:47:35 +00:00
parent b1ecc8319d
commit c4b7c671f9
2 changed files with 100 additions and 74 deletions
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169
apps/codecs/vorbis.c
View file

@ -25,6 +25,8 @@
#include "dsp.h" #include "dsp.h"
#include "lib/codeclib.h" #include "lib/codeclib.h"
#define TEST_RESUME
static struct codec_api* rb; static struct codec_api* rb;
/* Some standard functions and variables needed by Tremor */ /* Some standard functions and variables needed by Tremor */
@ -88,6 +90,38 @@ long tell_handler(void *datasource)
return rb->curpos; return rb->curpos;
} }
/* This sets the DSP parameters based on the current logical bitstream
* (sampling rate, number of channels, etc). It also tries to guess
* reasonable buffer parameters based on the current quality setting.
*/
bool vorbis_set_codec_parameters(OggVorbis_File *vf)
{
vorbis_info* vi;
vi=ov_info(vf,-1);
if (vi==NULL) {
//rb->splash(HZ*2, true, "Vorbis Error");
return false;
}
if (rb->id3->frequency != NATIVE_FREQUENCY) {
rb->configure(CODEC_DSP_ENABLE, (bool *)true);
} else {
rb->configure(CODEC_DSP_ENABLE, (bool *)false);
}
rb->configure(DSP_SET_FREQUENCY, (int *)rb->id3->frequency);
if (vi->channels == 2) {
rb->configure(DSP_SET_STEREO_MODE, (int *)STEREO_INTERLEAVED);
} else if (vi->channels == 1) {
rb->configure(DSP_SET_STEREO_MODE, (int *)STEREO_MONO);
}
return true;
}
#ifdef USE_IRAM #ifdef USE_IRAM
extern char iramcopy[]; extern char iramcopy[];
extern char iramstart[]; extern char iramstart[];
@ -103,17 +137,16 @@ enum codec_status codec_start(struct codec_api* api)
{ {
ov_callbacks callbacks; ov_callbacks callbacks;
OggVorbis_File vf; OggVorbis_File vf;
vorbis_info* vi;
int error; int error;
long n; long n;
int current_section; int current_section;
int previous_section = -1;
int eof; int eof;
ogg_int64_t vf_offsets[2]; ogg_int64_t vf_offsets[2];
ogg_int64_t vf_dataoffsets; ogg_int64_t vf_dataoffsets;
ogg_uint32_t vf_serialnos; ogg_uint32_t vf_serialnos;
ogg_int64_t vf_pcmlengths[2]; ogg_int64_t vf_pcmlengths[2];
int current_stereo_mode = -1;
TEST_CODEC_API(api); TEST_CODEC_API(api);
@ -125,11 +158,19 @@ enum codec_status codec_start(struct codec_api* api)
rb->memcpy(iramstart, iramcopy, iramend-iramstart); rb->memcpy(iramstart, iramcopy, iramend-iramstart);
#endif #endif
rb->configure(CODEC_SET_FILEBUF_LIMIT, (int *)(1024*1024*2));
rb->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*64));
rb->configure(DSP_DITHER, (bool *)false); rb->configure(DSP_DITHER, (bool *)false);
rb->configure(DSP_SET_SAMPLE_DEPTH, (int *)(16)); rb->configure(DSP_SET_SAMPLE_DEPTH, (int *)(16));
/* Note: These are sane defaults for these values. Perhaps
* they should be set differently based on quality setting
*/
rb->configure(CODEC_SET_FILEBUF_LIMIT, (int *)(1024*1024*2));
/* The chunk size below is magic. If set any lower, resume
* doesn't work properly (ov_raw_seek() does the wrong thing).
*/
rb->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*256));
/* We need to flush reserver memory every track load. */ /* We need to flush reserver memory every track load. */
next_track: next_track:
@ -139,14 +180,7 @@ enum codec_status codec_start(struct codec_api* api)
while (!*rb->taginfo_ready && !rb->stop_codec) while (!*rb->taginfo_ready && !rb->stop_codec)
rb->yield(); rb->yield();
if (rb->id3->frequency != NATIVE_FREQUENCY) {
rb->configure(DSP_SET_FREQUENCY, (long *)(rb->id3->frequency));
rb->configure(CODEC_DSP_ENABLE, (bool *)true);
} else {
rb->configure(CODEC_DSP_ENABLE, (bool *)false);
}
/* Create a decoder instance */ /* Create a decoder instance */
callbacks.read_func=read_handler; callbacks.read_func=read_handler;
callbacks.seek_func=initial_seek_handler; callbacks.seek_func=initial_seek_handler;
@ -158,7 +192,7 @@ enum codec_status codec_start(struct codec_api* api)
/* If the non-seekable open was successful, we need to supply the missing /* If the non-seekable open was successful, we need to supply the missing
* data to make it seekable. This is a hack, but it's reasonable since we * data to make it seekable. This is a hack, but it's reasonable since we
* don't want to read the whole file into the buffer before we start * don't want to run the whole file through the buffer before we start
* playing. Using Tremor's seekable open routine would cause us to do * playing. Using Tremor's seekable open routine would cause us to do
* this, so we pretend not to be seekable at first. Then we fill in the * this, so we pretend not to be seekable at first. Then we fill in the
* missing fields of vf with 1) information in rb->id3, and 2) info * missing fields of vf with 1) information in rb->id3, and 2) info
@ -169,54 +203,32 @@ enum codec_status codec_start(struct codec_api* api)
* get here. * get here.
*/ */
if ( !error ) { if ( !error ) {
//rb->logf("no error"); vf.offsets = vf_offsets;
/* FIXME Should these be dynamically allocated? */ vf.dataoffsets = &vf_dataoffsets;
vf.offsets = vf_offsets; vf.serialnos = &vf_serialnos;
vf.dataoffsets = &vf_dataoffsets; vf.pcmlengths = vf_pcmlengths;
vf.serialnos = &vf_serialnos;
vf.pcmlengths = vf_pcmlengths;
vf.offsets[0] = 0;
vf.offsets[1] = rb->id3->filesize;
vf.dataoffsets[0] = vf.offset;
vf.pcmlengths[0] = 0;
vf.pcmlengths[1] = rb->id3->samples;
vf.serialnos[0] = vf.current_serialno;
vf.callbacks.seek_func=seek_handler;
vf.seekable = 1;
vf.offset = 58; /* length of Ogg header */
vf.end = rb->id3->filesize;
vf.ready_state = OPENED;
vf.links = 1;
/*if(ov_raw_seek(&vf,0)){
rb->logf("seek err");
}
*/
vf.offsets[0] = 0;
vf.offsets[1] = rb->id3->filesize;
vf.dataoffsets[0] = vf.offset;
vf.pcmlengths[0] = 0;
vf.pcmlengths[1] = rb->id3->samples;
vf.serialnos[0] = vf.current_serialno;
vf.callbacks.seek_func=seek_handler;
vf.seekable = 1;
vf.end = rb->id3->filesize;
vf.ready_state = OPENED;
vf.links = 1;
} else { } else {
//rb->logf("ov_open: %d", error); //rb->logf("ov_open: %d", error);
} return CODEC_ERROR;
vi=ov_info(&vf,-1);
if (vi==NULL) {
//rb->splash(HZ*2, true, "Vorbis Error");
return CODEC_ERROR;
} }
rb->configure(DSP_SET_FREQUENCY, (int *)rb->id3->frequency); if ( rb->id3->offset ) {
rb->advance_buffer(rb->id3->offset);
if (vi->channels == 2) { ov_raw_seek(&vf,rb->id3->offset);
if (current_stereo_mode != STEREO_INTERLEAVED) { rb->id3->offset = ov_raw_tell(&vf);
rb->configure(DSP_SET_STEREO_MODE, (int *)STEREO_INTERLEAVED); rb->set_elapsed(ov_time_tell(&vf));
current_stereo_mode = STEREO_INTERLEAVED;
}
} else if (vi->channels == 1) {
if (current_stereo_mode != STEREO_MONO) {
rb->configure(DSP_SET_STEREO_MODE, (int *)STEREO_MONO);
current_stereo_mode = STEREO_MONO;
}
} }
eof=0; eof=0;
@ -226,9 +238,9 @@ enum codec_status codec_start(struct codec_api* api)
break ; break ;
if (rb->seek_time) { if (rb->seek_time) {
if (ov_time_seek(&vf, rb->seek_time)) { if (ov_time_seek(&vf, rb->seek_time)) {
//rb->logf("ov_time_seek failed"); //rb->logf("ov_time_seek failed");
} }
rb->seek_time = 0; rb->seek_time = 0;
} }
@ -236,6 +248,15 @@ enum codec_status codec_start(struct codec_api* api)
/* Read host-endian signed 16 bit PCM samples */ /* Read host-endian signed 16 bit PCM samples */
n=ov_read(&vf,pcmbuf,sizeof(pcmbuf),&current_section); n=ov_read(&vf,pcmbuf,sizeof(pcmbuf),&current_section);
/* Change DSP and buffer settings for this bitstream */
if ( current_section != previous_section ) {
if (!vorbis_set_codec_parameters(&vf)) {
return CODEC_ERROR;
} else {
previous_section = current_section;
}
}
if (n==0) { if (n==0) {
eof=1; eof=1;
} else if (n < 0) { } else if (n < 0) {
@ -243,29 +264,31 @@ enum codec_status codec_start(struct codec_api* api)
} else { } else {
while (!rb->audiobuffer_insert(pcmbuf, n)) { while (!rb->audiobuffer_insert(pcmbuf, n)) {
rb->yield(); rb->yield();
if ( rb->seek_time ) { if ( rb->seek_time ) {
/* Hmmm, a seek was requested. Throw out the /* Hmmm, a seek was requested. Throw out the
* buffer and go back to the top of the loop. * buffer and go back to the top of the loop.
*/ */
break; break;
} }
} }
if ( !rb->seek_time ) { if ( !rb->seek_time ) {
rb->set_elapsed(ov_time_tell(&vf)); rb->id3->offset = ov_raw_tell(&vf);
rb->yield(); rb->set_elapsed(ov_time_tell(&vf));
} rb->yield();
}
} }
} }
if (rb->request_next_track()) { if (rb->request_next_track()) {
/* Clean things up for the next track */ /* Clean things up for the next track */
vf.dataoffsets = NULL; vf.dataoffsets = NULL;
vf.offsets = NULL; vf.offsets = NULL;
vf.serialnos = NULL; vf.serialnos = NULL;
vf.pcmlengths = NULL; vf.pcmlengths = NULL;
ov_clear(&vf); ov_clear(&vf);
goto next_track; goto next_track;
} }
return CODEC_OK; return CODEC_OK;
} }

5
apps/playback.c
View file

@ -477,7 +477,7 @@ static bool rebuffer_and_seek(int newpos)
if (ci.stop_codec) if (ci.stop_codec)
return false; return false;
} }
return true; return true;
} }
@ -962,6 +962,9 @@ bool audio_load_track(int offset, bool start_play, int peek_offset)
ci.curpos = offset; ci.curpos = offset;
tracks[track_widx].start_pos = offset; tracks[track_widx].start_pos = offset;
break; break;
case AFMT_OGG_VORBIS:
tracks[track_widx].id3.offset = offset;
break;
} }
} }