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authorMax Kellermann <max@duempel.org>2014-08-31 16:12:26 +0200
committerMax Kellermann <max@duempel.org>2014-08-31 16:12:26 +0200
commite5a28bfd8d07fe5ca5dee1bfb55ce414c8f7e9fc (patch)
tree304e2b5bfa829d95d2c756449f010bcc76762eac /src/pcm/PcmDop.cxx
parent6e04d66a354c02910ebd849f2233e52de8b6e3c4 (diff)
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output/alsa, pcm: rename "DSD over USB" to "DoP"
The standard has been renamed since the early draft that was implemented in MPD.
Diffstat (limited to 'src/pcm/PcmDop.cxx')
-rw-r--r--src/pcm/PcmDop.cxx96
1 files changed, 96 insertions, 0 deletions
diff --git a/src/pcm/PcmDop.cxx b/src/pcm/PcmDop.cxx
new file mode 100644
index 000000000..b2096d9e4
--- /dev/null
+++ b/src/pcm/PcmDop.cxx
@@ -0,0 +1,96 @@
+/*
+ * Copyright (C) 2003-2014 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 "PcmDop.hxx"
+#include "PcmBuffer.hxx"
+#include "AudioFormat.hxx"
+#include "util/ConstBuffer.hxx"
+
+#include <assert.h>
+
+constexpr
+static inline uint32_t
+pcm_two_dsd_to_dop_marker1(uint8_t a, uint8_t b)
+{
+ return 0xff050000 | (a << 8) | b;
+}
+
+constexpr
+static inline uint32_t
+pcm_two_dsd_to_dop_marker2(uint8_t a, uint8_t b)
+{
+ return 0xfffa0000 | (a << 8) | b;
+}
+
+ConstBuffer<uint32_t>
+pcm_dsd_to_dop(PcmBuffer &buffer, unsigned channels,
+ ConstBuffer<uint8_t> _src)
+{
+ assert(audio_valid_channel_count(channels));
+ assert(!_src.IsNull());
+ assert(_src.size > 0);
+ assert(_src.size % channels == 0);
+
+ const unsigned num_src_samples = _src.size;
+ const unsigned num_src_frames = num_src_samples / channels;
+
+ /* this rounds down and discards the last odd frame; not
+ elegant, but good enough for now */
+ const unsigned num_frames = num_src_frames / 2;
+ const unsigned num_samples = num_frames * channels;
+
+ uint32_t *const dest0 = (uint32_t *)buffer.GetT<uint32_t>(num_samples),
+ *dest = dest0;
+
+ auto src = _src.data;
+ for (unsigned i = num_frames / 2; i > 0; --i) {
+ for (unsigned c = channels; c > 0; --c) {
+ /* each 24 bit sample has 16 DSD sample bits
+ plus the magic 0x05 marker */
+
+ *dest++ = pcm_two_dsd_to_dop_marker1(src[0], src[channels]);
+
+ /* seek the source pointer to the next
+ channel */
+ ++src;
+ }
+
+ /* skip the second byte of each channel, because we
+ have already copied it */
+ src += channels;
+
+ for (unsigned c = channels; c > 0; --c) {
+ /* each 24 bit sample has 16 DSD sample bits
+ plus the magic 0xfa marker */
+
+ *dest++ = pcm_two_dsd_to_dop_marker2(src[0], src[channels]);
+
+ /* seek the source pointer to the next
+ channel */
+ ++src;
+ }
+
+ /* skip the second byte of each channel, because we
+ have already copied it */
+ src += channels;
+ }
+
+ return { dest0, num_samples };
+}