rockbox/apps/codecs/vorbis.c

257 lines
7.3 KiB
C

/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2002 Björn Stenberg
*
* All files in this archive are subject to the GNU General Public License.
* See the file COPYING in the source tree root for full license agreement.
*
* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
#include "codeclib.h"
#include "Tremor/ivorbisfile.h"
#include "Tremor/ogg.h"
static struct codec_api *rb;
/* Some standard functions and variables needed by Tremor */
int errno;
size_t read_handler(void *ptr, size_t size, size_t nmemb, void *datasource)
{
(void)datasource;
return rb->read_filebuf(ptr, nmemb*size);
}
int initial_seek_handler(void *datasource, ogg_int64_t offset, int whence)
{
(void)datasource;
(void)offset;
(void)whence;
return -1;
}
int seek_handler(void *datasource, ogg_int64_t offset, int whence)
{
(void)datasource;
if (whence == SEEK_CUR) {
offset += rb->curpos;
} else if (whence == SEEK_END) {
offset += rb->filesize;
}
if (rb->seek_buffer(offset)) {
return 0;
}
return -1;
}
int close_handler(void *datasource)
{
(void)datasource;
return 0;
}
long tell_handler(void *datasource)
{
(void)datasource;
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;
}
rb->configure(DSP_SET_FREQUENCY, (int *)rb->id3->frequency);
codec_set_replaygain(rb->id3);
if (vi->channels == 2) {
rb->configure(DSP_SET_STEREO_MODE, (int *)STEREO_NONINTERLEAVED);
} else if (vi->channels == 1) {
rb->configure(DSP_SET_STEREO_MODE, (int *)STEREO_MONO);
}
return true;
}
#ifdef USE_IRAM
extern char iramcopy[];
extern char iramstart[];
extern char iramend[];
#endif
/* this is the codec entry point */
enum codec_status codec_start(struct codec_api *api)
{
ov_callbacks callbacks;
OggVorbis_File vf;
ogg_int32_t **pcm;
int error;
long n;
int current_section;
int previous_section = -1;
int eof;
ogg_int64_t vf_offsets[2];
ogg_int64_t vf_dataoffsets;
ogg_uint32_t vf_serialnos;
ogg_int64_t vf_pcmlengths[2];
TEST_CODEC_API(api);
rb = api;
#ifdef USE_IRAM
rb->memcpy(iramstart, iramcopy, iramend - iramstart);
#endif
rb->configure(CODEC_DSP_ENABLE, (bool *)true);
rb->configure(DSP_DITHER, (bool *)false);
rb->configure(DSP_SET_SAMPLE_DEPTH, (long *)24);
rb->configure(DSP_SET_CLIP_MAX, (long *)((1 << 24) - 1));
rb->configure(DSP_SET_CLIP_MIN, (long *)-((1 << 24) - 1));
/* Note: These are sane defaults for these values. Perhaps
* they should be set differently based on quality setting
*/
rb->configure(CODEC_SET_FILEBUF_LIMIT, (long *)(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, (long *)(1024*256));
/* We need to flush reserver memory every track load. */
next_track:
if (codec_init(rb)) {
return CODEC_ERROR;
}
while (!*rb->taginfo_ready && !rb->stop_codec)
rb->sleep(1);
/* Create a decoder instance */
callbacks.read_func = read_handler;
callbacks.seek_func = initial_seek_handler;
callbacks.tell_func = tell_handler;
callbacks.close_func = close_handler;
/* Open a non-seekable stream */
error = ov_open_callbacks(rb, &vf, NULL, 0, callbacks);
/* 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
* 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
* 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
* obtained by Tremor in the above ov_open call.
*
* Note that this assumes there is only ONE logical Vorbis bitstream in our
* physical Ogg bitstream. This is verified in metadata.c, well before we
* get here.
*/
if (!error) {
vf.offsets = vf_offsets;
vf.dataoffsets = &vf_dataoffsets;
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.end = rb->id3->filesize;
vf.ready_state = OPENED;
vf.links = 1;
} else {
//rb->logf("ov_open: %d", error);
return CODEC_ERROR;
}
if (rb->id3->offset) {
rb->advance_buffer(rb->id3->offset);
ov_raw_seek(&vf, rb->id3->offset);
rb->set_elapsed(ov_time_tell(&vf));
rb->set_offset(ov_raw_tell(&vf));
}
eof = 0;
while (!eof) {
rb->yield();
if (rb->stop_codec || rb->reload_codec)
break;
if (rb->seek_time) {
if (ov_time_seek(&vf, rb->seek_time)) {
//rb->logf("ov_time_seek failed");
}
rb->seek_complete();
}
/* Read host-endian signed 24-bit PCM samples */
n = ov_read_fixed(&vf, &pcm, 1024, &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) {
eof = 1;
} else if (n < 0) {
DEBUGF("Error decoding frame\n");
} else {
while (!rb->pcmbuf_insert_split(pcm[0], pcm[1],
n*sizeof(ogg_int32_t))) {
rb->sleep(1);
}
rb->set_offset(ov_raw_tell(&vf));
rb->set_elapsed(ov_time_tell(&vf));
}
}
if (rb->request_next_track()) {
/* Clean things up for the next track */
vf.dataoffsets = NULL;
vf.offsets = NULL;
vf.serialnos = NULL;
vf.pcmlengths = NULL;
ov_clear(&vf);
previous_section = -1;
goto next_track;
}
return CODEC_OK;
}