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Diffstat (limited to '')
-rw-r--r-- | src/decoder/FLAC_PCM.cxx | 110 |
1 files changed, 110 insertions, 0 deletions
diff --git a/src/decoder/FLAC_PCM.cxx b/src/decoder/FLAC_PCM.cxx new file mode 100644 index 000000000..303530aa7 --- /dev/null +++ b/src/decoder/FLAC_PCM.cxx @@ -0,0 +1,110 @@ +/* + * Copyright (C) 2003-2012 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.hxx" + +#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_FLOAT: + case SAMPLE_FORMAT_DSD: + case SAMPLE_FORMAT_UNDEFINED: + /* unreachable */ + assert(false); + } +} |