rockbox/apps/codecs/vox.c

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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2010 Yoshihisa Uchida
*
* 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 "codecs/libpcm/support_formats.h"
CODEC_HEADER
/* vox codec (Dialogic telephony file formats) */
#define PCM_SAMPLE_SIZE (2048)
static int32_t samples[PCM_SAMPLE_SIZE] IBSS_ATTR;
static struct pcm_format format;
static uint32_t bytesdone;
static uint8_t *read_buffer(size_t *realsize)
{
uint8_t *buffer = (uint8_t *)ci->request_buffer(realsize, format.chunksize);
if (bytesdone + (*realsize) > format.numbytes)
*realsize = format.numbytes - bytesdone;
bytesdone += *realsize;
ci->advance_buffer(*realsize);
return buffer;
}
/* this is the codec entry point */
enum codec_status codec_main(void)
{
int status = CODEC_OK;
uint32_t decodedsamples;
size_t n;
int bufcount;
int endofstream;
uint8_t *voxbuf;
off_t firstblockposn = 0; /* position of the first block in file */
const struct pcm_codec *codec;
/* Generic codec initialisation */
ci->configure(DSP_SET_SAMPLE_DEPTH, PCM_OUTPUT_DEPTH-1);
next_track:
if (codec_init()) {
DEBUGF("codec_init() error\n");
status = CODEC_ERROR;
goto exit;
}
while (!*ci->taginfo_ready && !ci->stop_codec)
ci->sleep(1);
codec_set_replaygain(ci->id3);
/* Need to save offset for later use (cleared indirectly by advance_buffer) */
bytesdone = ci->id3->offset;
ci->memset(&format, 0, sizeof(struct pcm_format));
/* set format */
format.channels = 1;
format.bitspersample = 4;
format.numbytes = ci->id3->filesize;
format.blockalign = 1;
/* advance to first WAVE chunk */
firstblockposn = 0;
decodedsamples = 0;
ci->advance_buffer(firstblockposn);
/*
* get codec
* supports dialogic oki adpcm only
*/
codec = get_dialogic_oki_adpcm_codec();
if (!codec)
{
DEBUGF("CODEC_ERROR: dialogic oki adpcm codec does not load.\n");
status = CODEC_ERROR;
goto done;
}
if (!codec->set_format(&format))
{
status = CODEC_ERROR;
goto done;
}
if (format.numbytes == 0) {
DEBUGF("CODEC_ERROR: data size is 0\n");
status = CODEC_ERROR;
goto done;
}
/* check chunksize */
if (format.chunksize * 2 > PCM_SAMPLE_SIZE)
format.chunksize = PCM_SAMPLE_SIZE / 2;
if (format.chunksize == 0)
{
DEBUGF("CODEC_ERROR: chunksize is 0\n");
status = CODEC_ERROR;
goto done;
}
ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO);
/* make sure we're at the correct offset */
if (bytesdone > (uint32_t) firstblockposn) {
/* Round down to previous block */
struct pcm_pos *newpos = codec->get_seek_pos(bytesdone - firstblockposn,
PCM_SEEK_POS, &read_buffer);
if (newpos->pos > format.numbytes)
goto done;
if (ci->seek_buffer(firstblockposn + newpos->pos))
{
bytesdone = newpos->pos;
decodedsamples = newpos->samples;
}
ci->seek_complete();
} else {
/* already where we need to be */
bytesdone = 0;
}
/* The main decoder loop */
endofstream = 0;
while (!endofstream) {
ci->yield();
if (ci->stop_codec || ci->new_track) {
break;
}
if (ci->seek_time) {
struct pcm_pos *newpos = codec->get_seek_pos(ci->seek_time, PCM_SEEK_TIME,
&read_buffer);
if (newpos->pos > format.numbytes)
break;
if (ci->seek_buffer(firstblockposn + newpos->pos))
{
bytesdone = newpos->pos;
decodedsamples = newpos->samples;
}
ci->seek_complete();
}
voxbuf = (uint8_t *)ci->request_buffer(&n, format.chunksize);
if (n == 0)
break; /* End of stream */
if (bytesdone + n > format.numbytes) {
n = format.numbytes - bytesdone;
endofstream = 1;
}
status = codec->decode(voxbuf, n, samples, &bufcount);
if (status == CODEC_ERROR)
{
DEBUGF("codec error\n");
goto done;
}
ci->pcmbuf_insert(samples, NULL, bufcount);
ci->advance_buffer(n);
bytesdone += n;
decodedsamples += bufcount;
if (bytesdone >= format.numbytes)
endofstream = 1;
ci->set_elapsed(decodedsamples*1000LL/ci->id3->frequency);
}
status = CODEC_OK;
done:
if (ci->request_next_track())
goto next_track;
exit:
return status;
}