forked from len0rd/rockbox
Cleaned up code. Now passes full precision samples to the playback engine, and DSP is enabled.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7620 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Thom Johansen
parent
b91e9fdfa9
commit
6762810ba6
1 changed files with 112 additions and 161 deletions
|
|
@ -18,208 +18,159 @@
|
||||||
****************************************************************************/
|
****************************************************************************/
|
||||||
|
|
||||||
#include "codec.h"
|
#include "codec.h"
|
||||||
|
#include "lib/codeclib.h"
|
||||||
#include <inttypes.h> /* Needed by a52.h */
|
#include <inttypes.h> /* Needed by a52.h */
|
||||||
#include <codecs/liba52/config-a52.h>
|
#include <codecs/liba52/config-a52.h>
|
||||||
#include <codecs/liba52/a52.h>
|
#include <codecs/liba52/a52.h>
|
||||||
|
|
||||||
#include "playback.h"
|
|
||||||
#include "dsp.h"
|
|
||||||
#include "lib/codeclib.h"
|
|
||||||
|
|
||||||
#define BUFFER_SIZE 4096
|
#define BUFFER_SIZE 4096
|
||||||
|
|
||||||
struct codec_api* rb;
|
struct codec_api *ci;
|
||||||
struct codec_api* ci;
|
|
||||||
|
|
||||||
static float gain = 1;
|
static a52_state_t *state;
|
||||||
static a52_state_t * state;
|
|
||||||
unsigned long samplesdone;
|
unsigned long samplesdone;
|
||||||
unsigned long frequency;
|
unsigned long frequency;
|
||||||
|
|
||||||
/* Two buffers used outside liba52 */
|
/* used outside liba52 */
|
||||||
static uint8_t buf[3840] IDATA_ATTR;
|
static uint8_t buf[3840] IDATA_ATTR;
|
||||||
static int16_t int16_samples[256*2] IDATA_ATTR;
|
|
||||||
|
|
||||||
static inline int16_t convert (int32_t i)
|
void output_audio(sample_t *samples, int flags)
|
||||||
{
|
{
|
||||||
i >>= 15;
|
flags &= A52_CHANNEL_MASK | A52_LFE;
|
||||||
return (i > 32767) ? 32767 : ((i < -32768) ? -32768 : i);
|
|
||||||
|
do {
|
||||||
|
ci->yield();
|
||||||
|
} while (!ci->pcmbuf_insert_split(&samples[0], &samples[256],
|
||||||
|
256*sizeof(sample_t)));
|
||||||
}
|
}
|
||||||
|
|
||||||
void output_audio(sample_t* samples,int flags) {
|
void a52_decode_data(uint8_t *start, uint8_t *end)
|
||||||
int i;
|
|
||||||
|
|
||||||
flags &= A52_CHANNEL_MASK | A52_LFE;
|
|
||||||
|
|
||||||
/* We may need to check the output format in flags - I'm not sure... */
|
|
||||||
for (i = 0; i < 256; i++) {
|
|
||||||
int16_samples[2*i] = convert (samples[i]);
|
|
||||||
int16_samples[2*i+1] = convert (samples[i+256]);
|
|
||||||
}
|
|
||||||
|
|
||||||
rb->yield();
|
|
||||||
while(!ci->pcmbuf_insert((unsigned char*)int16_samples,256*2*2))
|
|
||||||
rb->yield();
|
|
||||||
}
|
|
||||||
|
|
||||||
|
|
||||||
void a52_decode_data (uint8_t * start, uint8_t * end)
|
|
||||||
{
|
{
|
||||||
static uint8_t * bufptr = buf;
|
static uint8_t *bufptr = buf;
|
||||||
static uint8_t * bufpos = buf + 7;
|
static uint8_t *bufpos = buf + 7;
|
||||||
|
/*
|
||||||
|
* sample_rate and flags are static because this routine could
|
||||||
|
* exit between the a52_syncinfo() and the ao_setup(), and we want
|
||||||
|
* to have the same values when we get back !
|
||||||
|
*/
|
||||||
|
static int sample_rate;
|
||||||
|
static int flags;
|
||||||
|
int bit_rate;
|
||||||
|
int len;
|
||||||
|
|
||||||
/*
|
while (1) {
|
||||||
* sample_rate and flags are static because this routine could
|
len = end - start;
|
||||||
* exit between the a52_syncinfo() and the ao_setup(), and we want
|
if (!len)
|
||||||
* to have the same values when we get back !
|
break;
|
||||||
*/
|
if (len > bufpos - bufptr)
|
||||||
|
len = bufpos - bufptr;
|
||||||
|
memcpy(bufptr, start, len);
|
||||||
|
bufptr += len;
|
||||||
|
start += len;
|
||||||
|
if (bufptr == bufpos) {
|
||||||
|
if (bufpos == buf + 7) {
|
||||||
|
int length;
|
||||||
|
|
||||||
static int sample_rate;
|
length = a52_syncinfo(buf, &flags, &sample_rate, &bit_rate);
|
||||||
static int flags;
|
if (!length) {
|
||||||
int bit_rate;
|
//DEBUGF("skip\n");
|
||||||
int len;
|
for (bufptr = buf; bufptr < buf + 6; bufptr++)
|
||||||
|
bufptr[0] = bufptr[1];
|
||||||
|
continue;
|
||||||
|
}
|
||||||
|
bufpos = buf + length;
|
||||||
|
} else {
|
||||||
|
/* The following two defaults are taken from audio_out_oss.c: */
|
||||||
|
level_t level = 1 << 26;
|
||||||
|
sample_t bias = 0;
|
||||||
|
int i;
|
||||||
|
|
||||||
while (1) {
|
/* This is the configuration for the downmixing: */
|
||||||
len = end - start;
|
flags = A52_STEREO | A52_ADJUST_LEVEL | A52_LFE;
|
||||||
if (!len)
|
|
||||||
break;
|
|
||||||
if (len > bufpos - bufptr)
|
|
||||||
len = bufpos - bufptr;
|
|
||||||
memcpy (bufptr, start, len);
|
|
||||||
bufptr += len;
|
|
||||||
start += len;
|
|
||||||
if (bufptr == bufpos) {
|
|
||||||
if (bufpos == buf + 7) {
|
|
||||||
int length;
|
|
||||||
|
|
||||||
length = a52_syncinfo (buf, &flags, &sample_rate, &bit_rate);
|
if (a52_frame(state, buf, &flags, &level, bias))
|
||||||
if (!length) {
|
goto error;
|
||||||
DEBUGF("skip\n");
|
|
||||||
for (bufptr = buf; bufptr < buf + 6; bufptr++)
|
|
||||||
bufptr[0] = bufptr[1];
|
|
||||||
continue;
|
|
||||||
}
|
|
||||||
bufpos = buf + length;
|
|
||||||
} else {
|
|
||||||
// The following two defaults are taken from audio_out_oss.c:
|
|
||||||
level_t level;
|
|
||||||
sample_t bias;
|
|
||||||
int i;
|
|
||||||
|
|
||||||
/* This is the configuration for the downmixing: */
|
frequency = sample_rate;
|
||||||
flags=A52_STEREO|A52_ADJUST_LEVEL|A52_LFE;
|
|
||||||
level=(1 << 26);
|
|
||||||
bias=0;
|
|
||||||
|
|
||||||
level = (level_t) (level * gain);
|
/* An A52 frame consists of 6 blocks of 256 samples
|
||||||
|
So we decode and output them one block at a time */
|
||||||
if (a52_frame (state, buf, &flags, &level, bias)) {
|
for (i = 0; i < 6; i++) {
|
||||||
goto error;
|
if (a52_block(state))
|
||||||
}
|
goto error;
|
||||||
|
output_audio(a52_samples(state), flags);
|
||||||
// file_info->frames_decoded++;
|
samplesdone += 256;
|
||||||
|
}
|
||||||
// /* We assume this never changes */
|
ci->set_elapsed(samplesdone/(frequency/1000));
|
||||||
// file_info->samplerate=sample_rate;
|
bufptr = buf;
|
||||||
frequency=sample_rate;
|
bufpos = buf + 7;
|
||||||
|
continue;
|
||||||
// An A52 frame consists of 6 blocks of 256 samples
|
error:
|
||||||
// So we decode and output them one block at a time
|
//logf("Error decoding A52 stream\n");
|
||||||
for (i = 0; i < 6; i++) {
|
bufptr = buf;
|
||||||
if (a52_block (state)) {
|
bufpos = buf + 7;
|
||||||
goto error;
|
}
|
||||||
}
|
}
|
||||||
|
|
||||||
output_audio(a52_samples (state),flags);
|
|
||||||
samplesdone+=256;
|
|
||||||
}
|
|
||||||
ci->set_elapsed(samplesdone/(frequency/1000));
|
|
||||||
bufptr = buf;
|
|
||||||
bufpos = buf + 7;
|
|
||||||
continue;
|
|
||||||
|
|
||||||
error:
|
|
||||||
|
|
||||||
//logf("Error decoding A52 stream\n");
|
|
||||||
bufptr = buf;
|
|
||||||
bufpos = buf + 7;
|
|
||||||
}
|
|
||||||
}
|
}
|
||||||
}
|
|
||||||
}
|
}
|
||||||
|
|
||||||
#ifndef SIMULATOR
|
#ifdef USE_IRAM
|
||||||
extern char iramcopy[];
|
extern char iramcopy[];
|
||||||
extern char iramstart[];
|
extern char iramstart[];
|
||||||
extern char iramend[];
|
extern char iramend[];
|
||||||
#endif
|
#endif
|
||||||
|
|
||||||
/* this is the codec entry point */
|
/* this is the codec entry point */
|
||||||
enum codec_status codec_start(struct codec_api* api)
|
enum codec_status codec_start(struct codec_api *api)
|
||||||
{
|
{
|
||||||
size_t n;
|
size_t n;
|
||||||
unsigned char* filebuf;
|
unsigned char *filebuf;
|
||||||
|
|
||||||
/* Generic codec initialisation */
|
/* Generic codec initialisation */
|
||||||
TEST_CODEC_API(api);
|
TEST_CODEC_API(api);
|
||||||
|
ci = api;
|
||||||
|
|
||||||
rb = api;
|
#ifdef USE_IRAM
|
||||||
ci = (struct codec_api*)api;
|
ci->memcpy(iramstart, iramcopy, iramend - iramstart);
|
||||||
|
#endif
|
||||||
|
|
||||||
#ifndef SIMULATOR
|
ci->configure(CODEC_DSP_ENABLE, (bool *)true);
|
||||||
rb->memcpy(iramstart, iramcopy, iramend-iramstart);
|
ci->configure(DSP_DITHER, (bool *)false);
|
||||||
#endif
|
ci->configure(DSP_SET_STEREO_MODE, (long *)STEREO_NONINTERLEAVED);
|
||||||
|
ci->configure(DSP_SET_SAMPLE_DEPTH, (long *)30);
|
||||||
|
ci->configure(DSP_SET_CLIP_MAX, (long *)((1 << 30) - 1));
|
||||||
|
ci->configure(DSP_SET_CLIP_MIN, (long *)-(1 << 30));
|
||||||
|
ci->configure(CODEC_SET_FILEBUF_LIMIT, (long *)(1024*1024*2));
|
||||||
|
ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (long *)(1024*128));
|
||||||
|
|
||||||
ci->configure(CODEC_SET_FILEBUF_LIMIT, (int *)(1024*1024*2));
|
next_track:
|
||||||
ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*128));
|
if (codec_init(api))
|
||||||
|
return CODEC_ERROR;
|
||||||
|
|
||||||
ci->configure(DSP_DITHER, (bool *)false);
|
while (!ci->taginfo_ready)
|
||||||
ci->configure(DSP_SET_STEREO_MODE, (int *)STEREO_INTERLEAVED);
|
ci->yield();
|
||||||
ci->configure(DSP_SET_SAMPLE_DEPTH, (int *)(16));
|
|
||||||
|
|
||||||
next_track:
|
|
||||||
|
|
||||||
if (codec_init(api)) {
|
|
||||||
return CODEC_ERROR;
|
|
||||||
}
|
|
||||||
|
|
||||||
while (!rb->taginfo_ready)
|
|
||||||
rb->yield();
|
|
||||||
|
|
||||||
if (rb->id3->frequency != NATIVE_FREQUENCY) {
|
ci->configure(DSP_SET_FREQUENCY, (long *)(ci->id3->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);
|
|
||||||
}
|
|
||||||
|
|
||||||
/* Intialise the A52 decoder and check for success */
|
/* Intialise the A52 decoder and check for success */
|
||||||
state = a52_init (0); // Parameter is "accel"
|
state = a52_init(0);
|
||||||
|
|
||||||
/* The main decoding loop */
|
/* The main decoding loop */
|
||||||
|
samplesdone = 0;
|
||||||
|
while (1) {
|
||||||
|
if (ci->stop_codec || ci->reload_codec)
|
||||||
|
break;
|
||||||
|
|
||||||
samplesdone=0;
|
filebuf = ci->request_buffer(&n, BUFFER_SIZE);
|
||||||
while (1) {
|
|
||||||
if (ci->stop_codec || ci->reload_codec) {
|
|
||||||
break;
|
|
||||||
}
|
|
||||||
|
|
||||||
filebuf=ci->request_buffer(&n,BUFFER_SIZE);
|
if (n == 0) /* End of Stream */
|
||||||
|
break;
|
||||||
if (n==0) { /* End of Stream */
|
|
||||||
break;
|
|
||||||
}
|
|
||||||
|
|
||||||
a52_decode_data(filebuf,filebuf+n);
|
a52_decode_data(filebuf, filebuf + n);
|
||||||
|
ci->advance_buffer(n);
|
||||||
ci->advance_buffer(n);
|
}
|
||||||
}
|
if (ci->request_next_track())
|
||||||
|
goto next_track;
|
||||||
if (ci->request_next_track())
|
a52_free(state);
|
||||||
goto next_track;
|
return CODEC_OK;
|
||||||
|
|
||||||
//NOT NEEDED??: a52_free (state);
|
|
||||||
|
|
||||||
return CODEC_OK;
|
|
||||||
}
|
}
|
||||||
|
|
|
||||||
Loading…
Add table
Add a link
Reference in a new issue