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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2005 Dave Chapman
*
* 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 "codec.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 */
static uint8_t buf[3840] IDATA_ATTR;
static int16_t int16_samples[256*2] IDATA_ATTR;
static inline int16_t convert (int32_t i)
{
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]);
}
rb->yield();
while(!ci->audiobuffer_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;
int len;
while (1) {
len = end - start;
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 (!length) {
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: */
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)) {
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
for (i = 0; i < 6; i++) {
if (a52_block (state)) {
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
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)
{
size_t n;
unsigned char* filebuf;
/* Generic codec initialisation */
TEST_CODEC_API(api);
rb = api;
ci = (struct codec_api*)api;
#ifndef SIMULATOR
rb->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(DSP_DITHER, (bool *)false);
ci->configure(DSP_SET_STEREO_MODE, (int *)STEREO_INTERLEAVED);
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) {
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 */
state = a52_init (0); // Parameter is "accel"
/* The main decoding loop */
samplesdone=0;
while (1) {
if (ci->stop_codec || ci->reload_codec) {
break;
}
filebuf=ci->request_buffer(&n,BUFFER_SIZE);
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);
return CODEC_OK;
}
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