a2b5db0003
DSD-over-USB should not be a MPD core format, because it is not a "natural" format; it is just a temnporary over-the-wire format. This format has been implemented in pcm_export, and does not need to be supported by pcm_convert.
111 lines
2.8 KiB
C
111 lines
2.8 KiB
C
/*
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* Copyright (C) 2003-2011 The Music Player Daemon Project
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* http://www.musicpd.org
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with this program; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#include "config.h"
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#include "flac_pcm.h"
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#include <assert.h>
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static void flac_convert_stereo16(int16_t *dest,
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const FLAC__int32 * const buf[],
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unsigned int position, unsigned int end)
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{
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for (; position < end; ++position) {
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*dest++ = buf[0][position];
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*dest++ = buf[1][position];
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}
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}
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static void
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flac_convert_16(int16_t *dest,
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unsigned int num_channels,
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const FLAC__int32 * const buf[],
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unsigned int position, unsigned int end)
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{
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unsigned int c_chan;
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for (; position < end; ++position)
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for (c_chan = 0; c_chan < num_channels; c_chan++)
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*dest++ = buf[c_chan][position];
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}
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/**
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* Note: this function also handles 24 bit files!
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*/
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static void
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flac_convert_32(int32_t *dest,
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unsigned int num_channels,
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const FLAC__int32 * const buf[],
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unsigned int position, unsigned int end)
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{
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unsigned int c_chan;
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for (; position < end; ++position)
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for (c_chan = 0; c_chan < num_channels; c_chan++)
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*dest++ = buf[c_chan][position];
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}
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static void
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flac_convert_8(int8_t *dest,
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unsigned int num_channels,
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const FLAC__int32 * const buf[],
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unsigned int position, unsigned int end)
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{
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unsigned int c_chan;
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for (; position < end; ++position)
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for (c_chan = 0; c_chan < num_channels; c_chan++)
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*dest++ = buf[c_chan][position];
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}
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void
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flac_convert(void *dest,
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unsigned int num_channels, enum sample_format sample_format,
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const FLAC__int32 *const buf[],
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unsigned int position, unsigned int end)
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{
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switch (sample_format) {
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case SAMPLE_FORMAT_S16:
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if (num_channels == 2)
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flac_convert_stereo16((int16_t*)dest, buf,
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position, end);
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else
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flac_convert_16((int16_t*)dest, num_channels, buf,
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position, end);
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break;
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case SAMPLE_FORMAT_S24_P32:
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case SAMPLE_FORMAT_S32:
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flac_convert_32((int32_t*)dest, num_channels, buf,
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position, end);
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break;
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case SAMPLE_FORMAT_S8:
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flac_convert_8((int8_t*)dest, num_channels, buf,
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position, end);
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break;
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case SAMPLE_FORMAT_FLOAT:
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case SAMPLE_FORMAT_DSD:
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case SAMPLE_FORMAT_UNDEFINED:
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/* unreachable */
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assert(false);
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}
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}
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