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-rw-r--r--src/decoder/FLAC_PCM.cxx110
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);
+ }
+}