rockbox/lib/rbcodec/codecs/asap.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

146 lines
4.1 KiB
C

/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2008 Dominik Wenger
*
* 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 "libasap/asap.h"
CODEC_HEADER
#define CHUNK_SIZE (1024*2)
static byte samples[CHUNK_SIZE] IBSS_ATTR; /* The sample buffer */
static ASAP_State asap IBSS_ATTR; /* asap codec state */
/* 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)
{
int n_bytes;
int song;
int duration;
char* module;
int bytesPerSample =2;
intptr_t param;
if (codec_init()) {
DEBUGF("codec init failed\n");
return CODEC_ERROR;
}
param = ci->id3->elapsed;
codec_set_replaygain(ci->id3);
int bytes_done =0;
size_t filesize;
ci->seek_buffer(0);
module = ci->request_buffer(&filesize, ci->filesize);
if (!module || (size_t)filesize < (size_t)ci->filesize)
{
DEBUGF("loading error\n");
return CODEC_ERROR;
}
/*Init ASAP */
if (!ASAP_Load(&asap, ci->id3->path, module, filesize))
{
DEBUGF("%s: format not supported",ci->id3->path);
return CODEC_ERROR;
}
/* Make use of 44.1khz */
ci->configure(DSP_SET_FREQUENCY, 44100);
/* Sample depth is 16 bit little endian */
ci->configure(DSP_SET_SAMPLE_DEPTH, 16);
/* Stereo or Mono output ? */
if(asap.module_info->channels ==1)
{
ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO);
bytesPerSample = 2;
}
else
{
ci->configure(DSP_SET_STEREO_MODE, STEREO_INTERLEAVED);
bytesPerSample = 4;
}
song = asap.module_info->default_song;
duration = asap.module_info->durations[song];
if (duration < 0)
duration = 180 * 1000;
/* set id3 length, because metadata parse might not have done it */
ci->id3->length = duration;
ASAP_PlaySong(&asap, song, duration);
ASAP_MutePokeyChannels(&asap, 0);
if (param)
goto resume_start;
ci->set_elapsed(0);
/* The main decoder loop */
while (1) {
enum codec_command_action action = ci->get_command(&param);
if (action == CODEC_ACTION_HALT)
break;
if (action == CODEC_ACTION_SEEK_TIME) {
resume_start:
/* New time is ready in param */
/* seek to pos */
ASAP_Seek(&asap,param);
/* update bytes_done */
bytes_done = param*44.1*2;
/* update elapsed */
ci->set_elapsed((bytes_done / 2) / 44.1);
/* seek ready */
ci->seek_complete();
}
/* Generate a buffer full of Audio */
#ifdef ROCKBOX_LITTLE_ENDIAN
n_bytes = ASAP_Generate(&asap, samples, sizeof(samples), ASAP_FORMAT_S16_LE);
#else
n_bytes = ASAP_Generate(&asap, samples, sizeof(samples), ASAP_FORMAT_S16_BE);
#endif
ci->pcmbuf_insert(samples, NULL, n_bytes /bytesPerSample);
bytes_done += n_bytes;
ci->set_elapsed((bytes_done / 2) / 44.1);
if(n_bytes != sizeof(samples))
break;
}
return CODEC_OK;
}