7ad2cad173
buffer chunks. * Samples and position indication is closely associated with audio data instead of compensating by a latency constant. Alleviates problems with using the elapsed as a track indicator where it could be off by several steps. * Timing is accurate throughout track even if resampling for pitch shift, whereas before it updated during transition latency at the normal 1:1 rate. * Simpler PCM buffer with a constant chunk size, no linked lists. In converting crossfade, a minor change was made to not change the WPS until the fade-in of the incoming track, whereas before it would change upon the start of the fade-out of the outgoing track possibly having the WPS change with far too much lead time. Codec changes are to set elapsed times *before* writing next PCM frame because time and position data last set are saved in the next committed PCM chunk. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30366 a1c6a512-1295-4272-9138-f99709370657
202 lines
7.4 KiB
C
202 lines
7.4 KiB
C
/***************************************************************************
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* __________ __ ___.
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* Open \______ \ ____ ____ | | _\_ |__ _______ ___
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* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
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* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
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* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
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* \/ \/ \/ \/ \/
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* $Id$
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*
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* Copyright (C) 2009 Mohamed Tarek
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License
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* as published by the Free Software Foundation; either version 2
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* of the License, or (at your option) any later version.
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*
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* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
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* KIND, either express or implied.
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*
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****************************************************************************/
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#include <string.h>
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#include "logf.h"
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#include "codeclib.h"
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#include "inttypes.h"
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#include "libcook/cook.h"
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CODEC_HEADER
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static RMContext rmctx IBSS_ATTR_COOK_LARGE_IRAM;
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static RMPacket pkt IBSS_ATTR_COOK_LARGE_IRAM;
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static COOKContext q IBSS_ATTR;
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static int32_t rm_outbuf[2048] IBSS_ATTR_COOK_LARGE_IRAM MEM_ALIGN_ATTR;
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static void init_rm(RMContext *rmctx)
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{
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memcpy(rmctx, (void*)(( (intptr_t)ci->id3->id3v2buf + 3 ) &~ 3), sizeof(RMContext));
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}
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/* this is the codec entry point */
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enum codec_status codec_main(enum codec_entry_call_reason reason)
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{
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/* Nothing to do */
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return CODEC_OK;
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(void)reason;
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}
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/* this is called for each file to process */
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enum codec_status codec_run(void)
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{
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static size_t buff_size;
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int datasize, res, consumed, i, time_offset;
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uint8_t *bit_buffer;
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uint16_t fs,sps,h;
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uint32_t packet_count;
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int scrambling_unit_size, num_units;
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size_t resume_offset;
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intptr_t param = 0;
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enum codec_command_action action = CODEC_ACTION_NULL;
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if (codec_init()) {
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DEBUGF("codec init failed\n");
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return CODEC_ERROR;
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}
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resume_offset = ci->id3->offset;
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codec_set_replaygain(ci->id3);
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ci->memset(&rmctx,0,sizeof(RMContext));
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ci->memset(&pkt,0,sizeof(RMPacket));
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ci->memset(&q,0,sizeof(COOKContext));
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ci->seek_buffer(0);
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init_rm(&rmctx);
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ci->configure(DSP_SET_FREQUENCY, ci->id3->frequency);
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/* cook's sample representation is 21.11
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* DSP_SET_SAMPLE_DEPTH = 11 (FRACT) + 16 (NATIVE) - 1 (SIGN) = 26 */
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ci->configure(DSP_SET_SAMPLE_DEPTH, 26);
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ci->configure(DSP_SET_STEREO_MODE, rmctx.nb_channels == 1 ?
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STEREO_MONO : STEREO_NONINTERLEAVED);
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packet_count = rmctx.nb_packets;
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rmctx.audio_framesize = rmctx.block_align;
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rmctx.block_align = rmctx.sub_packet_size;
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fs = rmctx.audio_framesize;
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sps= rmctx.block_align;
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h = rmctx.sub_packet_h;
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scrambling_unit_size = h * (fs + PACKET_HEADER_SIZE);
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res =cook_decode_init(&rmctx, &q);
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if(res < 0) {
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DEBUGF("failed to initialize cook decoder\n");
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return CODEC_ERROR;
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}
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/* check for a mid-track resume and force a seek time accordingly */
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if(resume_offset > rmctx.data_offset + DATA_HEADER_SIZE) {
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resume_offset -= rmctx.data_offset + DATA_HEADER_SIZE;
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num_units = (int)resume_offset / scrambling_unit_size;
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/* put number of subpackets to skip in resume_offset */
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resume_offset /= (sps + PACKET_HEADER_SIZE);
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param = (int)resume_offset * ((sps * 8 * 1000)/rmctx.bit_rate);
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action = CODEC_ACTION_SEEK_TIME;
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}
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else {
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ci->set_elapsed(0);
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}
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ci->advance_buffer(rmctx.data_offset + DATA_HEADER_SIZE);
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/* The main decoder loop */
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seek_start :
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while(packet_count)
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{
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bit_buffer = (uint8_t *) ci->request_buffer(&buff_size, scrambling_unit_size);
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consumed = rm_get_packet(&bit_buffer, &rmctx, &pkt);
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if(consumed < 0) {
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DEBUGF("rm_get_packet failed\n");
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return CODEC_ERROR;
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}
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for(i = 0; i < rmctx.audio_pkt_cnt*(fs/sps) ; i++)
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{
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if (action == CODEC_ACTION_NULL)
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action = ci->get_command(¶m);
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if (action == CODEC_ACTION_HALT)
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return CODEC_OK;
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if (action == CODEC_ACTION_SEEK_TIME) {
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/* Do not allow seeking beyond the file's length */
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if ((unsigned) param > ci->id3->length) {
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ci->set_elapsed(ci->id3->length);
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ci->seek_complete();
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return CODEC_OK;
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}
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ci->seek_buffer(rmctx.data_offset + DATA_HEADER_SIZE);
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packet_count = rmctx.nb_packets;
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rmctx.audio_pkt_cnt = 0;
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rmctx.frame_number = 0;
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/* Seek to the start of the track */
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if (param == 0) {
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ci->set_elapsed(0);
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ci->seek_complete();
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action = CODEC_ACTION_NULL;
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goto seek_start;
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}
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num_units = (param/(sps*1000*8/rmctx.bit_rate))/(h*(fs/sps));
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ci->seek_buffer(rmctx.data_offset + DATA_HEADER_SIZE + consumed * num_units);
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bit_buffer = (uint8_t *) ci->request_buffer(&buff_size, scrambling_unit_size);
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consumed = rm_get_packet(&bit_buffer, &rmctx, &pkt);
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if(consumed < 0) {
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DEBUGF("rm_get_packet failed\n");
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ci->seek_complete();
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return CODEC_ERROR;
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}
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packet_count = rmctx.nb_packets - rmctx.audio_pkt_cnt * num_units;
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rmctx.frame_number = (param/(sps*1000*8/rmctx.bit_rate));
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while(rmctx.audiotimestamp > (unsigned) param) {
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rmctx.audio_pkt_cnt = 0;
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ci->seek_buffer(rmctx.data_offset + DATA_HEADER_SIZE + consumed * (num_units-1));
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bit_buffer = (uint8_t *) ci->request_buffer(&buff_size, scrambling_unit_size);
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consumed = rm_get_packet(&bit_buffer, &rmctx, &pkt);
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packet_count += rmctx.audio_pkt_cnt;
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num_units--;
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}
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time_offset = param - rmctx.audiotimestamp;
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i = (time_offset/((sps * 8 * 1000)/rmctx.bit_rate));
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ci->set_elapsed(rmctx.audiotimestamp+(1000*8*sps/rmctx.bit_rate)*i);
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ci->seek_complete();
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}
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action = CODEC_ACTION_NULL;
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res = cook_decode_frame(&rmctx,&q, rm_outbuf, &datasize, pkt.frames[i], rmctx.block_align);
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rmctx.frame_number++;
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/* skip the first two frames; no valid audio */
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if(rmctx.frame_number < 3) continue;
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if(res != rmctx.block_align) {
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DEBUGF("codec error\n");
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return CODEC_ERROR;
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}
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ci->pcmbuf_insert(rm_outbuf,
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rm_outbuf+q.samples_per_channel,
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q.samples_per_channel);
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ci->set_elapsed(rmctx.audiotimestamp+(1000*8*sps/rmctx.bit_rate)*i);
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}
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packet_count -= rmctx.audio_pkt_cnt;
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rmctx.audio_pkt_cnt = 0;
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ci->advance_buffer(consumed);
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}
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return CODEC_OK;
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}
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