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authorMax Kellermann <max@duempel.org>2012-10-02 10:56:44 +0200
committerMax Kellermann <max@duempel.org>2012-10-02 10:59:56 +0200
commitcbdd042adc8131c4ef890195447760d229eb1f87 (patch)
treef32b528e955b7bdf468168b8c490052bd33a702a /src/decoder/flac_pcm.c
parenta9419da09c27986708002fcb6a9032371eb1ce2b (diff)
downloadmpd-cbdd042adc8131c4ef890195447760d229eb1f87.tar.gz
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decoder/flac: use C++ compiler
Diffstat (limited to 'src/decoder/flac_pcm.c')
-rw-r--r--src/decoder/flac_pcm.c110
1 files changed, 0 insertions, 110 deletions
diff --git a/src/decoder/flac_pcm.c b/src/decoder/flac_pcm.c
deleted file mode 100644
index 6964d8ac6..000000000
--- a/src/decoder/flac_pcm.c
+++ /dev/null
@@ -1,110 +0,0 @@
-/*
- * 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_FLOAT:
- case SAMPLE_FORMAT_DSD:
- case SAMPLE_FORMAT_UNDEFINED:
- /* unreachable */
- assert(false);
- }
-}