/* * Copyright (C) 2003-2010 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_UNDEFINED: /* unreachable */ assert(false); } }