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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2009 Mohamed Tarek
*
* 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 <codecs/librm/rm.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;
static unsigned long samplesdone;
static unsigned long frequency;
static RMContext rmctx;
static RMPacket pkt;
static void init_rm(RMContext *rmctx)
{
memcpy(rmctx, (void*)(( (intptr_t)ci->id3->id3v2buf + 3 ) &~ 3), sizeof(RMContext));
}
/* used outside liba52 */
static uint8_t buf[3840] IBSS_ATTR;
/* The following two functions, a52_decode_data and output_audio are taken from apps/codecs/a52.c */
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(enum codec_entry_call_reason reason)
{
if (reason == CODEC_LOAD) {
/* Generic codec initialisation */
ci->configure(DSP_SET_STEREO_MODE, STEREO_NONINTERLEAVED);
ci->configure(DSP_SET_SAMPLE_DEPTH, 28);
}
else if (reason == CODEC_UNLOAD) {
if (state)
a52_free(state);
}
return CODEC_OK;
}
/* this is called for each file to process */
enum codec_status codec_run(void)
{
size_t n;
uint8_t *filebuf;
int consumed, packet_offset;
int playback_on = -1;
size_t resume_offset;
intptr_t param;
enum codec_command_action action = CODEC_ACTION_NULL;
if (codec_init()) {
return CODEC_ERROR;
}
resume_offset = ci->id3->offset;
ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
codec_set_replaygain(ci->id3);
ci->seek_buffer(ci->id3->first_frame_offset);
/* Intializations */
state = a52_init(0);
ci->memset(&rmctx,0,sizeof(RMContext));
ci->memset(&pkt,0,sizeof(RMPacket));
init_rm(&rmctx);
/* check for a mid-track resume and force a seek time accordingly */
if(resume_offset > rmctx.data_offset + DATA_HEADER_SIZE) {
resume_offset -= rmctx.data_offset + DATA_HEADER_SIZE;
/* put number of subpackets to skip in resume_offset */
resume_offset /= (rmctx.block_align + PACKET_HEADER_SIZE);
param = (int)resume_offset * ((rmctx.block_align * 8 * 1000)/rmctx.bit_rate);
action = CODEC_ACTION_SEEK_TIME;
}
else {
/* Seek to the first packet */
ci->set_elapsed(0);
ci->advance_buffer(rmctx.data_offset + DATA_HEADER_SIZE );
}
/* The main decoding loop */
while((unsigned)rmctx.audio_pkt_cnt < rmctx.nb_packets) {
if (action == CODEC_ACTION_NULL)
action = ci->get_command(¶m);
if (action == CODEC_ACTION_HALT)
break;
if (action == CODEC_ACTION_SEEK_TIME) {
packet_offset = param / ((rmctx.block_align*8*1000)/rmctx.bit_rate);
ci->seek_buffer(rmctx.data_offset + DATA_HEADER_SIZE +
packet_offset*(rmctx.block_align + PACKET_HEADER_SIZE));
rmctx.audio_pkt_cnt = packet_offset;
samplesdone = (rmctx.sample_rate/1000 * param);
ci->set_elapsed(samplesdone/(frequency/1000));
ci->seek_complete();
}
action = CODEC_ACTION_NULL;
filebuf = ci->request_buffer(&n, rmctx.block_align + PACKET_HEADER_SIZE);
consumed = rm_get_packet(&filebuf, &rmctx, &pkt);
if(consumed < 0 && playback_on != 0) {
if(playback_on == -1) {
/* Error only if packet-parsing failed and playback hadn't started */
DEBUGF("rm_get_packet failed\n");
return CODEC_ERROR;
}
else {
break;
}
}
playback_on = 1;
a52_decode_data(filebuf, filebuf + rmctx.block_align);
ci->advance_buffer(pkt.length);
}
return CODEC_OK;
}
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