0001-Add-native-DSD-support-to-MPD-0.19.10.patch

From c159b5460e267edd377ba74a3ae6154e885b0a98 Mon Sep 17 00:00:00 2001
From: Jurgen Kramer <gtmkramer@xs4all.nl>
Date: Sun, 5 Jul 2015 10:38:28 +0200
Subject: [PATCH] Add native DSD support to MPD 0.19.10

Add native DSD support to MPD 0.19 using DSD_U8 and DSD_U32_BE sample formats.

See https://github.com/lintweaker/xmos-native-dsd for more info.

Jurgen Kramer
---
 Makefile.am                             |  5 ++-
 README-DSD                              | 52 +++++++++++++++++++++++
 configure.ac                            |  2 +-
 src/AudioFormat.cxx                     |  4 +-
 src/AudioFormat.hxx                     | 18 +++++++-
 src/decoder/plugins/FlacPcm.cxx         |  4 +-
 src/output/plugins/AlsaOutputPlugin.cxx | 75 +++++++++++++++++++++++++++++++--
 src/output/plugins/OssOutputPlugin.cxx  |  4 +-
 src/pcm/ChannelsConverter.cxx           |  4 +-
 src/pcm/FallbackResampler.cxx           |  6 ++-
 src/pcm/FormatConverter.cxx             |  6 ++-
 src/pcm/PcmDsdNative.cxx                | 69 ++++++++++++++++++++++++++++++
 src/pcm/PcmDsdNative.hxx                | 38 +++++++++++++++++
 src/pcm/PcmExport.cxx                   | 21 ++++++++-
 src/pcm/PcmExport.hxx                   | 21 ++++++++-
 src/pcm/PcmFormat.cxx                   | 10 ++++-
 src/pcm/PcmMix.cxx                      |  6 ++-
 src/pcm/Volume.cxx                      |  6 ++-
 18 files changed, 330 insertions(+), 21 deletions(-)
 create mode 100644 README-DSD
 create mode 100644 src/pcm/PcmDsdNative.cxx
 create mode 100644 src/pcm/PcmDsdNative.hxx

diff --git a/Makefile.am b/Makefile.am
index 89819de..dedc881 100644
--- a/Makefile.am
+++ b/Makefile.am
@@ -495,7 +495,8 @@ PCM_LIBS = \
 if ENABLE_DSD
 libpcm_a_SOURCES += \
 	src/pcm/PcmDsd.cxx src/pcm/PcmDsd.hxx \
-	src/pcm/dsd2pcm/dsd2pcm.c src/pcm/dsd2pcm/dsd2pcm.h
+	src/pcm/dsd2pcm/dsd2pcm.c src/pcm/dsd2pcm/dsd2pcm.h \
+	src/pcm/PcmDsdNative.cxx src/pcm/PcmNativeDsd.hxx
 endif
 
 if HAVE_LIBSAMPLERATE
@@ -2092,7 +2093,7 @@ endif
 #
 
 man_MANS = doc/mpd.1 doc/mpd.conf.5
-doc_DATA = AUTHORS COPYING NEWS README doc/mpdconf.example
+doc_DATA = AUTHORS COPYING NEWS README README-DSD doc/mpdconf.example
 
 DOCBOOK_FILES = doc/protocol.xml doc/user.xml doc/developer.xml
 
diff --git a/README-DSD b/README-DSD
new file mode 100644
index 0000000..d896020
--- /dev/null
+++ b/README-DSD
@@ -0,0 +1,52 @@
+[31-jan-15]
+This version of MPD 0.19 has been enhancend with DSD native playback options.
+DSD output using DoP is mutually exclusive with native DSD playback. 
+Disable DoP output (or do not configure DoP output) if you want native DSD playback.
+
+For MPD 0.18 and older:
+
+dsd_usb	"no"
+
+For MPD 0.19:
+
+dop	"no"
+
+Native DSD playback requires ALSA driver and ALSA library support. The ALSA
+library needs patching (ALSA lib <= 1.0.28). A version with the needed DSD_U32_BE sample
+format is not released yet. See: https://github.com/lintweaker/xmos-native-dsd
+
+Add the following configuration to the appropriate audio_output section(s) of your mpd configuration file (e.g.
+/etc/mpd.conf):
+
+audio_output {
+
+	# Normal config parts omitted
+
+
+	# Disable DSD DoP MPD 0.18 or older
+	dsd_usb		"no"
+
+	# Disable DSD DoP MPD 0.19
+	dop		"no"
+
+	# Enable native DSD
+	dsd_native	"yes"
+	# Set DSD native type
+	dsd_native_type	"2"
+
+}
+
+
+Use 'dsd_native_type "2"' for USB DACs that support 32-bit DSD_U32_BE sample
+formats (e.g. XMOS based USB DACs and Marantz/Kenwood DACs).
+
+Use 'dsd_native_type "0"' for USB DACs using 8-bit DSD_U8 format like the Botic
+driver for the BeagleBone Black.
+
+Enjoy!
+Jurgen Kramer
+
+
+https://github.com/lintweaker
+
+
diff --git a/configure.ac b/configure.ac
index 1a3471f..190d53f 100644
--- a/configure.ac
+++ b/configure.ac
@@ -1,6 +1,6 @@
 AC_PREREQ(2.60)
 
-AC_INIT(mpd, 0.19.10, musicpd-dev-team@lists.sourceforge.net)
+AC_INIT(mpd, 0.19.10-dsd, gtmkramer@xs4all.nl)
 
 VERSION_MAJOR=0
 VERSION_MINOR=19
diff --git a/src/AudioFormat.cxx b/src/AudioFormat.cxx
index edfb9d8..5b23ad9 100644
--- a/src/AudioFormat.cxx
+++ b/src/AudioFormat.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * Copyright (C) 2003-2015 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -48,6 +48,7 @@ sample_format_to_string(SampleFormat format)
 		return "?";
 
 	case SampleFormat::S8:
+	case SampleFormat::DSD_U8:
 		return "8";
 
 	case SampleFormat::S16:
@@ -57,6 +58,7 @@ sample_format_to_string(SampleFormat format)
 		return "24";
 
 	case SampleFormat::S32:
+	case SampleFormat::DSD_U32:
 		return "32";
 
 	case SampleFormat::FLOAT:
diff --git a/src/AudioFormat.hxx b/src/AudioFormat.hxx
index 0937ab8..a4d330f 100644
--- a/src/AudioFormat.hxx
+++ b/src/AudioFormat.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * Copyright (C) 2003-2015 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -56,6 +56,18 @@ enum class SampleFormat : uint8_t {
 	 * byte (8 samples) per channel.
 	 */
 	DSD,
+
+	/**
+	 * DSD native output, 1 bit samples. Each frame carries 1 byte
+	 * per channel
+	 */
+	DSD_U8,
+
+	/**
+	 * DSD native output, 1 bit samples. Each frames carries 4 DSD
+	 * 1 byte samples
+	 */
+	DSD_U32,
 };
 
 #if defined(WIN32) && GCC_CHECK_VERSION(4,6)
@@ -198,6 +210,8 @@ audio_valid_sample_format(SampleFormat format)
 	case SampleFormat::S32:
 	case SampleFormat::FLOAT:
 	case SampleFormat::DSD:
+	case SampleFormat::DSD_U8:
+	case SampleFormat::DSD_U32:
 		return true;
 
 	case SampleFormat::UNDEFINED:
@@ -256,9 +270,11 @@ sample_format_size(SampleFormat format)
 	case SampleFormat::S24_P32:
 	case SampleFormat::S32:
 	case SampleFormat::FLOAT:
+	case SampleFormat::DSD_U32:
 		return 4;
 
 	case SampleFormat::DSD:
+	case SampleFormat::DSD_U8:
 		/* each frame has 8 samples per channel */
 		return 1;
 
diff --git a/src/decoder/plugins/FlacPcm.cxx b/src/decoder/plugins/FlacPcm.cxx
index 311500f..3403298 100644
--- a/src/decoder/plugins/FlacPcm.cxx
+++ b/src/decoder/plugins/FlacPcm.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * Copyright (C) 2003-2015 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -104,6 +104,8 @@ flac_convert(void *dest,
 	case SampleFormat::FLOAT:
 	case SampleFormat::DSD:
 	case SampleFormat::UNDEFINED:
+	case SampleFormat::DSD_U8:
+	case SampleFormat::DSD_U32:
 		assert(false);
 		gcc_unreachable();
 	}
diff --git a/src/output/plugins/AlsaOutputPlugin.cxx b/src/output/plugins/AlsaOutputPlugin.cxx
index 28c374a..55b22ef 100644
--- a/src/output/plugins/AlsaOutputPlugin.cxx
+++ b/src/output/plugins/AlsaOutputPlugin.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * Copyright (C) 2003-2015 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -68,6 +68,17 @@ struct AlsaOutput {
 	 */
 	bool dop;
 
+	/**
+	 * Enable native DSD playback (requires ALSA driver support)
+	 */
+	bool dsd_native;
+
+	/**
+	 * dsd_native_type
+	 * 0 = regular, uses DSD_U8, 1 = reserved, 2 uses DSD_U32_BE
+	 */
+	unsigned int dsd_native_type;
+
 	/** libasound's buffer_time setting (in microseconds) */
 	unsigned int buffer_time;
 
@@ -156,6 +167,27 @@ AlsaOutput::Configure(const config_param &param, Error &error)
 		/* legacy name from MPD 0.18 and older: */
 		param.GetBlockValue("dsd_usb", false);
 
+	dsd_native = param.GetBlockValue("dsd_native", false);
+
+
+	/* If native DSD is enabled check for requested output type */
+	if (dsd_native) {
+		dsd_native_type = param.GetBlockValue("dsd_native_type", 255);
+		switch(dsd_native_type) {
+		case 0:
+		case 2:
+			break;
+		case 1:
+			dsd_native = false;
+		default:
+			dsd_native = false;
+		}
+	}
+
+	/* If both dop and dsd_native are enabled, fall back to dop */
+	if (dop && dsd_native)
+		dsd_native = false;
+
 	buffer_time = param.GetBlockValue("buffer_time",
 					      MPD_ALSA_BUFFER_TIME_US);
 	period_time = param.GetBlockValue("period_time", 0u);
@@ -270,6 +302,12 @@ get_bitformat(SampleFormat sample_format)
 
 	case SampleFormat::FLOAT:
 		return SND_PCM_FORMAT_FLOAT;
+
+	case SampleFormat::DSD_U8:
+		return SND_PCM_FORMAT_DSD_U8;
+
+	case SampleFormat::DSD_U32:
+		return SND_PCM_FORMAT_DSD_U32_BE;
 	}
 
 	assert(false);
@@ -643,9 +681,31 @@ alsa_setup_dop(AlsaOutput *ad, const AudioFormat audio_format,
 	assert(ad->dop);
 	assert(audio_format.format == SampleFormat::DSD);
 
+	AudioFormat dop_format = audio_format;
+
+	/* DSD native type 0 -> DSD_U8 */
+	if (ad->dsd_native && ad->dsd_native_type == 0) {
+
+		dop_format.format = SampleFormat::DSD_U8;
+		if (!alsa_setup(ad, dop_format, packed_r,
+		     reverse_endian_r, error))
+			return false;
+		return true;
+	}
+
+	/* DSD native type 2 -> DSD_U32_BE */
+	if (ad->dsd_native && ad->dsd_native_type == 2) {
+
+		dop_format.format = SampleFormat::DSD_U32;
+		dop_format.sample_rate /= 4;
+		if (!alsa_setup(ad, dop_format, packed_r,
+		    reverse_endian_r, error))
+			return false;
+		return true;
+	}
+
 	/* pass 24 bit to alsa_setup() */
 
-	AudioFormat dop_format = audio_format;
 	dop_format.format = SampleFormat::S24_P32;
 	dop_format.sample_rate /= 2;
 
@@ -684,7 +744,13 @@ alsa_setup_or_dop(AlsaOutput *ad, AudioFormat &audio_format,
 
 	const bool dop = ad->dop &&
 		audio_format.format == SampleFormat::DSD;
-	const bool success = dop
+
+	const bool dsd_native = ad->dsd_native &&
+		audio_format.format == SampleFormat::DSD;
+
+	const bool dsd_enabled = dop || dsd_native;
+
+	const bool success = dsd_enabled
 		? alsa_setup_dop(ad, audio_format,
 				 &shift8, &packed, &reverse_endian,
 				 error)
@@ -695,7 +761,8 @@ alsa_setup_or_dop(AlsaOutput *ad, AudioFormat &audio_format,
 
 	ad->pcm_export->Open(audio_format.format,
 			     audio_format.channels,
-			     dop, shift8, packed, reverse_endian);
+			     dop, shift8, packed, reverse_endian,
+			     dsd_native, ad->dsd_native_type);
 	return true;
 }
 
diff --git a/src/output/plugins/OssOutputPlugin.cxx b/src/output/plugins/OssOutputPlugin.cxx
index 39d87fc..c98304a 100644
--- a/src/output/plugins/OssOutputPlugin.cxx
+++ b/src/output/plugins/OssOutputPlugin.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * Copyright (C) 2003-2015 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -419,6 +419,8 @@ sample_format_to_oss(SampleFormat format)
 	case SampleFormat::UNDEFINED:
 	case SampleFormat::FLOAT:
 	case SampleFormat::DSD:
+	case SampleFormat::DSD_U8:
+	case SampleFormat::DSD_U32:
 		return AFMT_QUERY;
 
 	case SampleFormat::S8:
diff --git a/src/pcm/ChannelsConverter.cxx b/src/pcm/ChannelsConverter.cxx
index 7146137..b41ad86 100644
--- a/src/pcm/ChannelsConverter.cxx
+++ b/src/pcm/ChannelsConverter.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * Copyright (C) 2003-2015 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -68,6 +68,8 @@ PcmChannelsConverter::Convert(ConstBuffer<void> src, gcc_unused Error &error)
 	case SampleFormat::UNDEFINED:
 	case SampleFormat::S8:
 	case SampleFormat::DSD:
+	case SampleFormat::DSD_U8:
+	case SampleFormat::DSD_U32:
 		assert(false);
 		gcc_unreachable();
 
diff --git a/src/pcm/FallbackResampler.cxx b/src/pcm/FallbackResampler.cxx
index bd3f20d..e136f75 100644
--- a/src/pcm/FallbackResampler.cxx
+++ b/src/pcm/FallbackResampler.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * Copyright (C) 2003-2015 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -31,6 +31,8 @@ FallbackPcmResampler::Open(AudioFormat &af, unsigned new_sample_rate,
 
 	switch (af.format) {
 	case SampleFormat::UNDEFINED:
+	case SampleFormat::DSD_U8:
+	case SampleFormat::DSD_U32:
 		assert(false);
 		gcc_unreachable();
 
@@ -122,6 +124,8 @@ FallbackPcmResampler::Resample(ConstBuffer<void> src, gcc_unused Error &error)
 	case SampleFormat::UNDEFINED:
 	case SampleFormat::S8:
 	case SampleFormat::DSD:
+	case SampleFormat::DSD_U8:
+	case SampleFormat::DSD_U32:
 		assert(false);
 		gcc_unreachable();
 
diff --git a/src/pcm/FormatConverter.cxx b/src/pcm/FormatConverter.cxx
index 8874e1b..5238e1c 100644
--- a/src/pcm/FormatConverter.cxx
+++ b/src/pcm/FormatConverter.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * Copyright (C) 2003-2015 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -40,6 +40,8 @@ PcmFormatConverter::Open(SampleFormat _src_format, SampleFormat _dest_format,
 
 	case SampleFormat::S8:
 	case SampleFormat::DSD:
+	case SampleFormat::DSD_U8:
+	case SampleFormat::DSD_U32:
 		error.Format(pcm_domain,
 			     "PCM conversion from %s to %s is not implemented",
 			     sample_format_to_string(_src_format),
@@ -74,6 +76,8 @@ PcmFormatConverter::Convert(ConstBuffer<void> src, gcc_unused Error &error)
 	case SampleFormat::UNDEFINED:
 	case SampleFormat::S8:
 	case SampleFormat::DSD:
+	case SampleFormat::DSD_U8:
+	case SampleFormat::DSD_U32:
 		assert(false);
 		gcc_unreachable();
 
diff --git a/src/pcm/PcmDsdNative.cxx b/src/pcm/PcmDsdNative.cxx
new file mode 100644
index 0000000..2e0e7a1
--- /dev/null
+++ b/src/pcm/PcmDsdNative.cxx
@@ -0,0 +1,69 @@
+/*
+ * Copyright (C) 2014-2015 Jurgen Kramer
+ *
+ * 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 "PcmDsdNative.hxx"
+#include "PcmBuffer.hxx"
+#include "AudioFormat.hxx"
+#include "util/ConstBuffer.hxx"
+
+/* JK For debug info */
+#include "util/Domain.hxx"
+#include "Log.hxx"
+
+static constexpr Domain dsdn_dom("dsd_native");
+
+constexpr
+static inline uint32_t
+pcm_two_dsd_native(uint8_t a, uint8_t b, uint8_t c, uint8_t d)
+{
+	return 0x00000000 | (a << 24 ) | (b << 16) | (c << 8) | d;
+}
+
+ConstBuffer<uint32_t>
+pcm_dsd_native(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;
+
+	const unsigned num_frames = num_src_frames / 2;
+	unsigned num_samples = num_frames * channels;
+
+	uint32_t *const dest0 = (uint32_t *)buffer.GetT<uint32_t>(num_samples / 2),
+		*dest = dest0;
+
+	auto src = _src.data;
+
+	for (unsigned i = num_frames / 2; i > 0 ; --i) {
+
+		/* Left channel */
+		*dest++ = pcm_two_dsd_native(src[6], src[4], src[2], src[0]);
+		/* Right channel */
+		*dest++ = pcm_two_dsd_native(src[7], src[5], src[3], src[1]);
+
+		src += 8;
+	}
+
+	return { dest0, num_samples / 2 };
+}
diff --git a/src/pcm/PcmDsdNative.hxx b/src/pcm/PcmDsdNative.hxx
new file mode 100644
index 0000000..e904949
--- /dev/null
+++ b/src/pcm/PcmDsdNative.hxx
@@ -0,0 +1,38 @@
+/*
+ * Copyright (C) 2014-2015 Jurgen Kramer
+ *
+ * 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.
+ */
+
+#ifndef MPD_PCM_DSD_NATIVE_HXX
+#define MPD_PCM_DSD_NATIVE_HXX
+
+#include "check.h"
+
+#include <stdint.h>
+#include <stddef.h>
+
+class PcmBuffer;
+template<typename T> struct ConstBuffer;
+
+/**
+ * Pack DSD 1 bit samples into DSD_U32_LE samples for
+ * native DSD playback
+ */
+ConstBuffer<uint32_t>
+pcm_dsd_native(PcmBuffer &buffer, unsigned channels,
+	       ConstBuffer<uint8_t> src);
+
+#endif
diff --git a/src/pcm/PcmExport.cxx b/src/pcm/PcmExport.cxx
index ef099ba..0bcb9d7 100644
--- a/src/pcm/PcmExport.cxx
+++ b/src/pcm/PcmExport.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * Copyright (C) 2003-2015 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -20,6 +20,7 @@
 #include "config.h"
 #include "PcmExport.hxx"
 #include "PcmDop.hxx"
+#include "PcmDsdNative.hxx"
 #include "PcmPack.hxx"
 #include "util/ByteReverse.hxx"
 #include "util/ConstBuffer.hxx"
@@ -28,18 +29,26 @@
 
 void
 PcmExport::Open(SampleFormat sample_format, unsigned _channels,
-		bool _dop, bool _shift8, bool _pack, bool _reverse_endian)
+		bool _dop, bool _shift8, bool _pack, bool _reverse_endian,
+		bool _dsd_native, unsigned _dsd_native_type)
 {
 	assert(audio_valid_sample_format(sample_format));
 	assert(!_dop || audio_valid_channel_count(_channels));
 
 	channels = _channels;
 	dop = _dop && sample_format == SampleFormat::DSD;
+
+	dsd_native = _dsd_native;
+	dsd_native_type = _dsd_native_type;
+
 	if (dop)
 		/* after the conversion to DoP, the DSD
 		   samples are stuffed inside fake 24 bit samples */
 		sample_format = SampleFormat::S24_P32;
 
+	if (dsd_native && dsd_native_type == 2)
+		sample_format = SampleFormat::S32;
+
 	shift8 = _shift8 && sample_format == SampleFormat::S24_P32;
 	pack24 = _pack && sample_format == SampleFormat::S24_P32;
 
@@ -71,6 +80,9 @@ PcmExport::GetFrameSize(const AudioFormat &audio_format) const
 		   bytes per sample) */
 		return channels * 4;
 
+	if (dsd_native && dsd_native_type == 2)
+		return channels * 4;
+
 	return audio_format.GetFrameSize();
 }
 
@@ -82,6 +94,11 @@ PcmExport::Export(ConstBuffer<void> data)
 				      ConstBuffer<uint8_t>::FromVoid(data))
 			.ToVoid();
 
+	if (dsd_native && dsd_native_type == 2)
+		data = pcm_dsd_native(dop_buffer, channels,
+				      ConstBuffer<uint8_t>::FromVoid(data))
+			.ToVoid();
+
 	if (pack24) {
 		const auto src = ConstBuffer<int32_t>::FromVoid(data);
 		const size_t num_samples = src.size;
diff --git a/src/pcm/PcmExport.hxx b/src/pcm/PcmExport.hxx
index b99a358..6048505 100644
--- a/src/pcm/PcmExport.hxx
+++ b/src/pcm/PcmExport.hxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * Copyright (C) 2003-2015 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -68,6 +68,22 @@ struct PcmExport {
 	bool dop;
 
 	/**
+	 * Output native DSD?
+	 * dsd_native_type contains requested output type
+	 */
+
+	bool dsd_native;
+
+	/**
+	 * DSD native output type
+	 * 0 = DSD_U8, no export needed
+	 * 1 = DSD_U16_BE, reserved. Not supported yet
+	 * 2 = DSD_U32_BE, e.g. XMOS based USB DACs
+	 */
+
+	unsigned dsd_native_type;
+
+	/**
 	 * Convert (padded) 24 bit samples to 32 bit by shifting 8
 	 * bits to the left?
 	 */
@@ -96,7 +112,8 @@ struct PcmExport {
 	 * @param channels the number of channels; ignored unless dop is set
 	 */
 	void Open(SampleFormat sample_format, unsigned channels,
-		  bool dop, bool shift8, bool pack, bool reverse_endian);
+		  bool dop, bool shift8, bool pack, bool reverse_endian,
+		  bool dsd_native, unsigned dsd_native_type);
 
 	/**
 	 * Calculate the size of one output frame.
diff --git a/src/pcm/PcmFormat.cxx b/src/pcm/PcmFormat.cxx
index 4cabc05..775a9c7 100644
--- a/src/pcm/PcmFormat.cxx
+++ b/src/pcm/PcmFormat.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * Copyright (C) 2003-2015 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -166,6 +166,8 @@ pcm_convert_to_16(PcmBuffer &buffer, PcmDither &dither,
 	switch (src_format) {
 	case SampleFormat::UNDEFINED:
 	case SampleFormat::DSD:
+	case SampleFormat::DSD_U8:
+	case SampleFormat::DSD_U32:
 		break;
 
 	case SampleFormat::S8:
@@ -234,6 +236,8 @@ pcm_convert_to_24(PcmBuffer &buffer,
 	switch (src_format) {
 	case SampleFormat::UNDEFINED:
 	case SampleFormat::DSD:
+	case SampleFormat::DSD_U8:
+	case SampleFormat::DSD_U32:
 		break;
 
 	case SampleFormat::S8:
@@ -302,6 +306,8 @@ pcm_convert_to_32(PcmBuffer &buffer,
 	switch (src_format) {
 	case SampleFormat::UNDEFINED:
 	case SampleFormat::DSD:
+	case SampleFormat::DSD_U8:
+	case SampleFormat::DSD_U32:
 		break;
 
 	case SampleFormat::S8:
@@ -370,6 +376,8 @@ pcm_convert_to_float(PcmBuffer &buffer,
 	switch (src_format) {
 	case SampleFormat::UNDEFINED:
 	case SampleFormat::DSD:
+	case SampleFormat::DSD_U8:
+	case SampleFormat::DSD_U32:
 		break;
 
 	case SampleFormat::S8:
diff --git a/src/pcm/PcmMix.cxx b/src/pcm/PcmMix.cxx
index d21b5f0..c72afdb 100644
--- a/src/pcm/PcmMix.cxx
+++ b/src/pcm/PcmMix.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * Copyright (C) 2003-2015 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -93,6 +93,8 @@ pcm_add_vol(PcmDither &dither, void *buffer1, const void *buffer2, size_t size,
 	switch (format) {
 	case SampleFormat::UNDEFINED:
 	case SampleFormat::DSD:
+	case SampleFormat::DSD_U8:
+	case SampleFormat::DSD_U32:
 		/* not implemented */
 		return false;
 
@@ -181,6 +183,8 @@ pcm_add(void *buffer1, const void *buffer2, size_t size,
 	switch (format) {
 	case SampleFormat::UNDEFINED:
 	case SampleFormat::DSD:
+	case SampleFormat::DSD_U8:
+	case SampleFormat::DSD_U32:
 		/* not implemented */
 		return false;
 
diff --git a/src/pcm/Volume.cxx b/src/pcm/Volume.cxx
index b12d8fd..ec718be 100644
--- a/src/pcm/Volume.cxx
+++ b/src/pcm/Volume.cxx
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2003-2014 The Music Player Daemon Project
+ * Copyright (C) 2003-2015 The Music Player Daemon Project
  * http://www.musicpd.org
  *
  * This program is free software; you can redistribute it and/or modify
@@ -116,6 +116,8 @@ PcmVolume::Open(SampleFormat _format, Error &error)
 		break;
 
 	case SampleFormat::DSD:
+	case SampleFormat::DSD_U8:
+	case SampleFormat::DSD_U32:
 		// TODO: implement this; currently, it's a no-op
 		break;
 	}
@@ -181,6 +183,8 @@ PcmVolume::Apply(ConstBuffer<void> src)
 		break;
 
 	case SampleFormat::DSD:
+	case SampleFormat::DSD_U8:
+	case SampleFormat::DSD_U32:
 		// TODO: implement this; currently, it's a no-op
 		return src;
 	}
-- 
2.1.0