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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2005 Dave Chapman
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
#include "codeclib.h"
#include <inttypes.h> /* Needed by a52.h */
#include <codecs/liba52/config-a52.h>
#include <codecs/liba52/a52.h>
CODEC_HEADER
#define BUFFER_SIZE 4096
#define A52_SAMPLESPERFRAME (6*256)
static a52_state_t *state;
unsigned long samplesdone;
unsigned long frequency;
/* used outside liba52 */
static uint8_t buf[3840] IBSS_ATTR;
static inline void output_audio(sample_t *samples)
{
ci->yield();
ci->pcmbuf_insert(&samples[0], &samples[256], 256);
}
static 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 {
/* Unity gain is 1 << 26, and we want to end up on 28 bits
of precision instead of the default 30.
*/
level_t level = 1 << 24;
sample_t bias = 0;
int i;
/* This is the configuration for the downmixing: */
flags = A52_STEREO | A52_ADJUST_LEVEL;
if (a52_frame(state, buf, &flags, &level, bias))
goto error;
a52_dynrng(state, NULL, NULL);
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));
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;
}
}
}
}
/* this is the codec entry point */
enum codec_status codec_main(void)
{
size_t n;
unsigned char *filebuf;
int sample_loc;
int retval;
/* Generic codec initialisation */
ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
ci->configure(DSP_SET_SAMPLE_DEPTH, 28);
next_track:
if (codec_init()) {
retval = CODEC_ERROR;
goto exit;
}
while (!ci->taginfo_ready)
ci->yield();
ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
codec_set_replaygain(ci->id3);
/* Intialise the A52 decoder and check for success */
state = a52_init(0);
/* The main decoding loop */
if (ci->id3->offset) {
if (ci->seek_buffer(ci->id3->offset)) {
samplesdone = (ci->id3->offset / ci->id3->bytesperframe) *
A52_SAMPLESPERFRAME;
ci->set_elapsed(samplesdone/(ci->id3->frequency / 1000));
}
}
else {
samplesdone = 0;
}
while (1) {
if (ci->stop_codec || ci->new_track)
break;
if (ci->seek_time) {
sample_loc = (ci->seek_time - 1)/1000 * ci->id3->frequency;
if (ci->seek_buffer((sample_loc/A52_SAMPLESPERFRAME)*ci->id3->bytesperframe)) {
samplesdone = sample_loc;
ci->set_elapsed(samplesdone/(ci->id3->frequency/1000));
}
ci->seek_complete();
}
filebuf = ci->request_buffer(&n, BUFFER_SIZE);
if (n == 0) /* End of Stream */
break;
a52_decode_data(filebuf, filebuf + n);
ci->advance_buffer(n);
}
retval = CODEC_OK;
if (ci->request_next_track())
goto next_track;
exit:
a52_free(state);
return retval;
}
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