rockbox/apps/codecs/cook.c

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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 <string.h>
#include "logf.h"
#include "codeclib.h"
#include "inttypes.h"
#include "libcook/cook.h"
CODEC_HEADER
static RMContext rmctx;
static RMPacket pkt;
static COOKContext q IBSS_ATTR;
static int32_t rm_outbuf[2048];
static void init_rm(RMContext *rmctx)
{
memcpy(rmctx, (void*)(( (intptr_t)ci->id3->id3v2buf + 3 ) &~ 3), sizeof(RMContext));
}
/* this is the codec entry point */
enum codec_status codec_main(void)
{
static size_t buff_size;
int datasize, res, consumed, i, time_offset;
uint8_t *bit_buffer;
uint16_t fs,sps,h;
uint32_t packet_count;
int scrambling_unit_size, num_units;
size_t resume_offset;
next_track:
if (codec_init()) {
DEBUGF("codec init failed\n");
return CODEC_ERROR;
}
if (codec_wait_taginfo() != 0)
goto done;
resume_offset = ci->id3->offset;
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);
/* cook's sample representation is 21.11
* DSP_SET_SAMPLE_DEPTH = 11 (FRACT) + 16 (NATIVE) - 1 (SIGN) = 26 */
ci->configure(DSP_SET_SAMPLE_DEPTH, 26);
ci->configure(DSP_SET_STEREO_MODE, rmctx.nb_channels == 1 ?
STEREO_MONO : STEREO_NONINTERLEAVED);
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;
}
/* 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;
num_units = (int)resume_offset / scrambling_unit_size;
/* put number of subpackets to skip in resume_offset */
resume_offset /= (sps + PACKET_HEADER_SIZE);
ci->seek_time = (int)resume_offset * ((sps * 8 * 1000)/rmctx.bit_rate);
}
ci->set_elapsed(0);
ci->advance_buffer(rmctx.data_offset + DATA_HEADER_SIZE);
/* The main decoder loop */
seek_start :
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;
}
for(i = 0; i < rmctx.audio_pkt_cnt*(fs/sps) ; i++)
{
ci->yield();
if (ci->stop_codec || ci->new_track)
goto done;
if (ci->seek_time) {
ci->set_elapsed(ci->seek_time);
/* Do not allow seeking beyond the file's length */
if ((unsigned) ci->seek_time > ci->id3->length) {
ci->seek_complete();
goto done;
}
ci->seek_buffer(rmctx.data_offset + DATA_HEADER_SIZE);
packet_count = rmctx.nb_packets;
rmctx.audio_pkt_cnt = 0;
rmctx.frame_number = 0;
/* Seek to the start of the track */
if (ci->seek_time == 1) {
ci->set_elapsed(0);
ci->seek_complete();
goto seek_start;
}
num_units = ((ci->seek_time)/(sps*1000*8/rmctx.bit_rate))/(h*(fs/sps));
ci->seek_buffer(rmctx.data_offset + DATA_HEADER_SIZE + consumed * num_units);
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;
}
packet_count = rmctx.nb_packets - rmctx.audio_pkt_cnt * num_units;
rmctx.frame_number = ((ci->seek_time)/(sps*1000*8/rmctx.bit_rate));
while(rmctx.audiotimestamp > (unsigned) ci->seek_time) {
rmctx.audio_pkt_cnt = 0;
ci->seek_buffer(rmctx.data_offset + DATA_HEADER_SIZE + consumed * (num_units-1));
bit_buffer = (uint8_t *) ci->request_buffer(&buff_size, scrambling_unit_size);
consumed = rm_get_packet(&bit_buffer, &rmctx, &pkt);
packet_count += rmctx.audio_pkt_cnt;
num_units--;
}
time_offset = ci->seek_time - rmctx.audiotimestamp;
i = (time_offset/((sps * 8 * 1000)/rmctx.bit_rate));
ci->set_elapsed(rmctx.audiotimestamp+(1000*8*sps/rmctx.bit_rate)*i);
ci->seek_complete();
}
res = cook_decode_frame(&rmctx,&q, rm_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(rm_outbuf,
rm_outbuf+q.samples_per_channel,
q.samples_per_channel);
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);
}
done :
if (ci->request_next_track())
goto next_track;
return CODEC_OK;
}