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/***************************************************************************
* __________ __ ___.
* 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) {
long action = ci->get_command(¶m);
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;
}
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