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Cleaned up code. Now passes full precision samples to the playback engine, and DSP is enabled.

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git-svn-id: svn://svn.rockbox.org/rockbox/trunk@7620 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
Thom Johansen 2005-10-10 23:37:30 +00:00
parent b91e9fdfa9
commit 6762810ba6
1 changed files with 112 additions and 161 deletions
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117
apps/codecs/a52.c
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@ -18,63 +18,41 @@
****************************************************************************/
#include "codec.h"
#include "lib/codeclib.h"
#include <inttypes.h> /* Needed by a52.h */
#include <codecs/liba52/config-a52.h>
#include <codecs/liba52/a52.h>
#include "playback.h"
#include "dsp.h"
#include "lib/codeclib.h"
#define BUFFER_SIZE 4096
struct codec_api* rb;
struct codec_api *ci;
static float gain = 1;
static a52_state_t *state;
unsigned long samplesdone;
unsigned long frequency;
/* Two buffers used outside liba52 */
/* used outside liba52 */
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;
return (i > 32767) ? 32767 : ((i < -32768) ? -32768 : i);
}
void output_audio(sample_t* samples,int flags) {
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]);
do {
ci->yield();
} while (!ci->pcmbuf_insert_split(&samples[0], &samples[256],
256*sizeof(sample_t)));
}
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 *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;
@ -95,42 +73,31 @@ void a52_decode_data (uint8_t * start, uint8_t * end)
length = a52_syncinfo(buf, &flags, &sample_rate, &bit_rate);
if (!length) {
DEBUGF("skip\n");
//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;
/* The following two defaults are taken from audio_out_oss.c: */
level_t level = 1 << 26;
sample_t bias = 0;
int i;
/* This is the configuration for the downmixing: */
flags = A52_STEREO | A52_ADJUST_LEVEL | A52_LFE;
level=(1 << 26);
bias=0;
level = (level_t) (level * gain);
if (a52_frame (state, buf, &flags, &level, bias)) {
if (a52_frame(state, buf, &flags, &level, bias))
goto error;
}
// file_info->frames_decoded++;
// /* We assume this never changes */
// file_info->samplerate=sample_rate;
frequency = sample_rate;
// An A52 frame consists of 6 blocks of 256 samples
// So we decode and output them one block at a time
/* An A52 frame consists of 6 blocks of 256 samples
So we decode and output them one block at a time */
for (i = 0; i < 6; i++) {
if (a52_block (state)) {
if (a52_block(state))
goto error;
}
output_audio(a52_samples(state), flags);
samplesdone += 256;
}
@ -138,9 +105,7 @@ void a52_decode_data (uint8_t * start, uint8_t * end)
bufptr = buf;
bufpos = buf + 7;
continue;
error:
//logf("Error decoding A52 stream\n");
bufptr = buf;
bufpos = buf + 7;
@ -149,7 +114,7 @@ void a52_decode_data (uint8_t * start, uint8_t * end)
}
}
#ifndef SIMULATOR
#ifdef USE_IRAM
extern char iramcopy[];
extern char iramstart[];
extern char iramend[];
@ -163,63 +128,49 @@ enum codec_status codec_start(struct codec_api* api)
/* Generic codec initialisation */
TEST_CODEC_API(api);
ci = api;
rb = api;
ci = (struct codec_api*)api;
#ifndef SIMULATOR
rb->memcpy(iramstart, iramcopy, iramend-iramstart);
#ifdef USE_IRAM
ci->memcpy(iramstart, iramcopy, iramend - iramstart);
#endif
ci->configure(CODEC_SET_FILEBUF_LIMIT, (int *)(1024*1024*2));
ci->configure(CODEC_SET_FILEBUF_CHUNKSIZE, (int *)(1024*128));
ci->configure(CODEC_DSP_ENABLE, (bool *)true);
ci->configure(DSP_DITHER, (bool *)false);
ci->configure(DSP_SET_STEREO_MODE, (int *)STEREO_INTERLEAVED);
ci->configure(DSP_SET_SAMPLE_DEPTH, (int *)(16));
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));
next_track:
if (codec_init(api)) {
if (codec_init(api))
return CODEC_ERROR;
}
while (!rb->taginfo_ready)
rb->yield();
while (!ci->taginfo_ready)
ci->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);
}
ci->configure(DSP_SET_FREQUENCY, (long *)(ci->id3->frequency));
/* Intialise the A52 decoder and check for success */
state = a52_init (0); // Parameter is "accel"
state = a52_init(0);
/* The main decoding loop */
samplesdone = 0;
while (1) {
if (ci->stop_codec || ci->reload_codec) {
if (ci->stop_codec || ci->reload_codec)
break;
}
filebuf = ci->request_buffer(&n, BUFFER_SIZE);
if (n==0) { /* End of Stream */
if (n == 0) /* End of Stream */
break;
}
a52_decode_data(filebuf, filebuf + n);
ci->advance_buffer(n);
}
if (ci->request_next_track())
goto next_track;
//NOT NEEDED??: a52_free (state);
a52_free(state);
return CODEC_OK;
}