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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* 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 <string.h>
#include "logf.h"
#include "codeclib.h"
#include "inttypes.h"
#include "libcook/cook.h"
#define DATA_HEADER_SIZE 18 /* size of DATA chunk header in a rm file */
CODEC_HEADER
RMContext rmctx;
RMPacket pkt;
COOKContext q;
static void init_rm(RMContext *rmctx)
{
memcpy(rmctx, ci->id3->id3v2buf, sizeof(RMContext));
}
/* this is the codec entry point */
enum codec_status codec_main(void)
{
static size_t buff_size;
int datasize, res, consumed,i;
uint8_t *bit_buffer;
int16_t outbuf[2048] __attribute__((aligned(32)));
uint16_t fs,sps,h;
uint32_t packet_count;
int scrambling_unit_size;
next_track:
if (codec_init()) {
DEBUGF("codec init failed\n");
return CODEC_ERROR;
}
while (!*ci->taginfo_ready && !ci->stop_codec)
ci->sleep(1);
codec_set_replaygain(ci->id3);
ci->memset(&rmctx,0,sizeof(RMContext));
ci->memset(&pkt,0,sizeof(RMPacket));
ci->memset(&q,0,sizeof(COOKContext));
init_rm(&rmctx);
ci->configure(DSP_SET_FREQUENCY, ci->id3->frequency);
ci->configure(DSP_SET_SAMPLE_DEPTH, 16);
ci->configure(DSP_SET_STEREO_MODE, rmctx.nb_channels == 1 ?
STEREO_MONO : STEREO_INTERLEAVED);
packet_count = rmctx.nb_packets;
rmctx.audio_framesize = rmctx.block_align;
rmctx.block_align = rmctx.sub_packet_size;
fs = rmctx.audio_framesize;
sps= rmctx.block_align;
h = rmctx.sub_packet_h;
scrambling_unit_size = h*fs;
res =cook_decode_init(&rmctx, &q);
if(res < 0) {
DEBUGF("failed to initialize cook decoder\n");
return CODEC_ERROR;
}
ci->set_elapsed(0);
ci->advance_buffer(rmctx.data_offset + DATA_HEADER_SIZE);
/* The main decoder loop */
while (1)
{
/*if (ci->seek_time) {
ci->set_elapsed(ci->seek_time);
n = ci->seek_time/10;
memset(buf,0,BUF_SIZE);
ci->seek_complete();
}*/
while(packet_count)
{
bit_buffer = (uint8_t *) ci->request_buffer(&buff_size, scrambling_unit_size);
consumed = rm_get_packet(&bit_buffer, &rmctx, &pkt);
if(consumed < 0) {
DEBUGF("rm_get_packet failed\n");
return CODEC_ERROR;
}
/*DEBUGF(" version = %d\n"
" length = %d\n"
" stream = %d\n"
" timestamp= %d\n",pkt.version,pkt.length,pkt.stream_number,pkt.timestamp);*/
for(i = 0; i < rmctx.audio_pkt_cnt*(fs/sps) ; i++)
{
ci->yield();
if (ci->stop_codec || ci->new_track)
goto done;
res = cook_decode_frame(&rmctx,&q, outbuf, &datasize, pkt.frames[i], rmctx.block_align);
rmctx.frame_number++;
/* skip the first two frames; no valid audio */
if(rmctx.frame_number < 3) continue;
if(res != rmctx.block_align) {
DEBUGF("codec error\n");
return CODEC_ERROR;
}
ci->pcmbuf_insert(outbuf, NULL, rmctx.samples_pf_pc / rmctx.nb_channels);
ci->set_elapsed(rmctx.audiotimestamp+(1000*8*sps/rmctx.bit_rate)*i);
}
packet_count -= rmctx.audio_pkt_cnt;
rmctx.audio_pkt_cnt = 0;
ci->advance_buffer(consumed);
}
goto done;
}
done :
if (ci->request_next_track())
goto next_track;
return CODEC_OK;
}
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