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/*
* Copyright (C) 2003-2011 The Music Player Daemon Project
* http://www.musicpd.org
*
* 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 program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with this program; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include "config.h"
#include "flac_pcm.h"
#include <assert.h>
static void flac_convert_stereo16(int16_t *dest,
const FLAC__int32 * const buf[],
unsigned int position, unsigned int end)
{
for (; position < end; ++position) {
*dest++ = buf[0][position];
*dest++ = buf[1][position];
}
}
static void
flac_convert_16(int16_t *dest,
unsigned int num_channels,
const FLAC__int32 * const buf[],
unsigned int position, unsigned int end)
{
unsigned int c_chan;
for (; position < end; ++position)
for (c_chan = 0; c_chan < num_channels; c_chan++)
*dest++ = buf[c_chan][position];
}
/**
* Note: this function also handles 24 bit files!
*/
static void
flac_convert_32(int32_t *dest,
unsigned int num_channels,
const FLAC__int32 * const buf[],
unsigned int position, unsigned int end)
{
unsigned int c_chan;
for (; position < end; ++position)
for (c_chan = 0; c_chan < num_channels; c_chan++)
*dest++ = buf[c_chan][position];
}
static void
flac_convert_8(int8_t *dest,
unsigned int num_channels,
const FLAC__int32 * const buf[],
unsigned int position, unsigned int end)
{
unsigned int c_chan;
for (; position < end; ++position)
for (c_chan = 0; c_chan < num_channels; c_chan++)
*dest++ = buf[c_chan][position];
}
void
flac_convert(void *dest,
unsigned int num_channels, enum sample_format sample_format,
const FLAC__int32 *const buf[],
unsigned int position, unsigned int end)
{
switch (sample_format) {
case SAMPLE_FORMAT_S16:
if (num_channels == 2)
flac_convert_stereo16((int16_t*)dest, buf,
position, end);
else
flac_convert_16((int16_t*)dest, num_channels, buf,
position, end);
break;
case SAMPLE_FORMAT_S24_P32:
case SAMPLE_FORMAT_S32:
flac_convert_32((int32_t*)dest, num_channels, buf,
position, end);
break;
case SAMPLE_FORMAT_S8:
flac_convert_8((int8_t*)dest, num_channels, buf,
position, end);
break;
case SAMPLE_FORMAT_S24:
case SAMPLE_FORMAT_FLOAT:
case SAMPLE_FORMAT_DSD:
case SAMPLE_FORMAT_DSD_LSBFIRST:
case SAMPLE_FORMAT_UNDEFINED:
/* unreachable */
assert(false);
}
}
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