rockbox/lib/rbcodec/codecs/cook.c
Michael Sevakis 31b7122867 Implement time-based resume and playback start.
This complements offset-based resume and playback start funcionality.
The implementation is global on both HWCODEC and SWCODEC.

Basically, if either the specified elapsed or offset are non-zero,
it indicates a mid-track resume.

To resume by time only, set elapsed to nonzero and offset to zero.
To resume by offset only, set offset to nonzero and elapsed to zero.

Which one the codec uses and which has priority is up to the codec;
however, using an elapsed time covers more cases:

* Codecs not able to use an offset such as VGM or other atomic
formats

* Starting playback at a nonzero elapsed time from a source that
contains no offset, such as a cuesheet

The change re-versions pretty much everything from tagcache to nvram.

Change-Id: Ic7aebb24e99a03ae99585c5e236eba960d163f38
Reviewed-on: http://gerrit.rockbox.org/516
Reviewed-by: Michael Sevakis <jethead71@rockbox.org>
Tested: Michael Sevakis <jethead71@rockbox.org>
2014-03-10 04:12:30 +01:00

207 lines
7.4 KiB
C

/***************************************************************************
* __________ __ ___.
* 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 IBSS_ATTR_COOK_LARGE_IRAM;
static RMPacket pkt IBSS_ATTR_COOK_LARGE_IRAM;
static COOKContext q IBSS_ATTR;
static int32_t rm_outbuf[2048] IBSS_ATTR_COOK_LARGE_IRAM MEM_ALIGN_ATTR;
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(enum codec_entry_call_reason reason)
{
/* Nothing to do */
return CODEC_OK;
(void)reason;
}
/* this is called for each file to process */
enum codec_status codec_run(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;
intptr_t param;
enum codec_command_action action;
if (codec_init()) {
DEBUGF("codec init failed\n");
return CODEC_ERROR;
}
action = CODEC_ACTION_NULL;
param = ci->id3->elapsed;
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));
ci->seek_buffer(0);
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 + PACKET_HEADER_SIZE);
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) {
resume_offset -= MIN(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);
param = (int)resume_offset * ((sps * 8 * 1000)/rmctx.bit_rate);
}
if (param) {
action = CODEC_ACTION_SEEK_TIME;
}
else {
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++)
{
if (action == CODEC_ACTION_NULL)
action = ci->get_command(&param);
if (action == CODEC_ACTION_HALT)
return CODEC_OK;
if (action == CODEC_ACTION_SEEK_TIME) {
/* Do not allow seeking beyond the file's length */
if ((unsigned) param > ci->id3->length) {
ci->set_elapsed(ci->id3->length);
ci->seek_complete();
return CODEC_OK;
}
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 (param == 0) {
ci->set_elapsed(0);
ci->seek_complete();
action = CODEC_ACTION_NULL;
goto seek_start;
}
num_units = (param/(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");
ci->seek_complete();
return CODEC_ERROR;
}
packet_count = rmctx.nb_packets - rmctx.audio_pkt_cnt * num_units;
rmctx.frame_number = (param/(sps*1000*8/rmctx.bit_rate));
while(rmctx.audiotimestamp > (unsigned) param) {
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 = param - 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();
}
action = CODEC_ACTION_NULL;
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);
}
return CODEC_OK;
}