pcm/Order: new library to convert from FLAC to ALSA channel order

This new library is integrated in the PcmExport class and (if enabled)
converts MPD's channel order (= FLAC channel order) to ALSA channel
order.

This fixes:
 http://bugs.musicpd.org/view.php?id=3147
and
 http://bugs.musicpd.org/view.php?id=3255

4 files changed, 199 insertions, 0 deletions

diff --git a/src/pcm/Order.cxx b/src/pcm/Order.cxx
new file mode 100644
index 000000000..470f9d119
--- /dev/null
+++ b/src/pcm/Order.cxx
@@ -0,0 +1,135 @@
+/*
+ * 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
+ * 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 "Order.hxx"
+#include "PcmBuffer.hxx"
+#include "util/ConstBuffer.hxx"
+
+template<typename V>
+struct TwoPointers {
+	V *dest;
+	const V *src;
+
+	TwoPointers<V> &CopyOne() {
+		*dest++ = *src++;
+		return *this;
+	}
+
+	TwoPointers<V> &CopyTwo() {
+		return CopyOne().CopyOne();
+	}
+
+	TwoPointers<V> &SwapTwoPairs() {
+		*dest++ = src[2];
+		*dest++ = src[3];
+		*dest++ = src[0];
+		*dest++ = src[1];
+		src += 4;
+		return *this;
+	}
+
+	TwoPointers<V> &ToAlsa51() {
+		return CopyTwo() // left+right
+			.SwapTwoPairs(); // center, LFE, surround left+right
+	}
+
+	TwoPointers<V> &ToAlsa71() {
+		return ToAlsa51()
+			.CopyTwo(); // side left+right
+	}
+};
+
+template<typename V>
+static void
+ToAlsaChannelOrder51(V *dest, const V *src, size_t n)
+{
+	TwoPointers<V> p{dest, src};
+	for (size_t i = 0; i != n; ++i)
+		p.ToAlsa51();
+}
+
+template<typename V>
+static inline ConstBuffer<V>
+ToAlsaChannelOrder51(PcmBuffer &buffer, ConstBuffer<V> src)
+{
+	auto dest = buffer.GetT<V>(src.size);
+	ToAlsaChannelOrder51(dest, src.data, src.size / 6);
+	return { dest, src.size };
+}
+
+template<typename V>
+static void
+ToAlsaChannelOrder71(V *dest, const V *src, size_t n)
+{
+	TwoPointers<V> p{dest, src};
+	for (size_t i = 0; i != n; ++i)
+		p.ToAlsa71();
+}
+
+template<typename V>
+static inline ConstBuffer<V>
+ToAlsaChannelOrder71(PcmBuffer &buffer, ConstBuffer<V> src)
+{
+	auto dest = buffer.GetT<V>(src.size);
+	ToAlsaChannelOrder71(dest, src.data, src.size / 6);
+	return { dest, src.size };
+}
+
+template<typename V>
+static ConstBuffer<V>
+ToAlsaChannelOrderT(PcmBuffer &buffer, ConstBuffer<V> src, unsigned channels)
+{
+	switch (channels) {
+	case 6: // 5.1
+		return ToAlsaChannelOrder51(buffer, src);
+
+	case 8: // 7.1
+		return ToAlsaChannelOrder71(buffer, src);
+
+	default:
+		return src;
+	}
+}
+
+ConstBuffer<void>
+ToAlsaChannelOrder(PcmBuffer &buffer, ConstBuffer<void> src,
+		   SampleFormat sample_format, unsigned channels)
+{
+	switch (sample_format) {
+	case SampleFormat::UNDEFINED:
+	case SampleFormat::S8:
+	case SampleFormat::DSD:
+		return src;
+
+	case SampleFormat::S16:
+		return ToAlsaChannelOrderT(buffer,
+					   ConstBuffer<int16_t>::FromVoid(src),
+					   channels).ToVoid();
+
+	case SampleFormat::S24_P32:
+	case SampleFormat::S32:
+	case SampleFormat::FLOAT:
+		return ToAlsaChannelOrderT(buffer,
+					   ConstBuffer<int32_t>::FromVoid(src),
+					   channels).ToVoid();
+	}
+
+	gcc_unreachable();
+}
diff --git a/src/pcm/Order.hxx b/src/pcm/Order.hxx
new file mode 100644
index 000000000..14bfa2382
--- /dev/null
+++ b/src/pcm/Order.hxx
@@ -0,0 +1,37 @@
+/*
+ * 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
+ * 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_ORDER_HXX
+#define MPD_PCM_ORDER_HXX
+
+#include "check.h"
+#include "AudioFormat.hxx"
+
+class PcmBuffer;
+template<typename T> struct ConstBuffer;
+
+/**
+ * Convert the given buffer from FLAC channel order
+ * (https://xiph.org/flac/format.html) to ALSA channel order.
+ */
+ConstBuffer<void>
+ToAlsaChannelOrder(PcmBuffer &buffer, ConstBuffer<void> src,
+		   SampleFormat sample_format, unsigned channels);
+
+#endif
diff --git a/src/pcm/PcmExport.cxx b/src/pcm/PcmExport.cxx
index af2eb7d9f..0bf4f8be8 100644
--- a/src/pcm/PcmExport.cxx
+++ b/src/pcm/PcmExport.cxx
@@ -19,6 +19,7 @@
 
 #include "config.h"
 #include "PcmExport.hxx"
+#include "Order.hxx"
 #include "PcmDop.hxx"
 #include "PcmPack.hxx"
 #include "util/ByteReverse.hxx"
@@ -28,12 +29,16 @@
 
 void
 PcmExport::Open(SampleFormat sample_format, unsigned _channels,
+		bool _alsa_channel_order,
 		bool _dop, bool _shift8, bool _pack, bool _reverse_endian)
 {
 	assert(audio_valid_sample_format(sample_format));
 	assert(!_dop || audio_valid_channel_count(_channels));
 
 	channels = _channels;
+	alsa_channel_order = _alsa_channel_order
+		? sample_format
+		: SampleFormat::UNDEFINED;
 	dop = _dop && sample_format == SampleFormat::DSD;
 	if (dop)
 		/* after the conversion to DoP, the DSD
@@ -77,6 +82,10 @@ PcmExport::GetFrameSize(const AudioFormat &audio_format) const
 ConstBuffer<void>
 PcmExport::Export(ConstBuffer<void> data)
 {
+	if (alsa_channel_order != SampleFormat::UNDEFINED)
+		data = ToAlsaChannelOrder(order_buffer, data,
+					  alsa_channel_order, channels);
+
 	if (dop)
 		data = pcm_dsd_to_dop(dop_buffer, channels,
 				      ConstBuffer<uint8_t>::FromVoid(data))
diff --git a/src/pcm/PcmExport.hxx b/src/pcm/PcmExport.hxx
index 7265ca07d..aafa1cea0 100644
--- a/src/pcm/PcmExport.hxx
+++ b/src/pcm/PcmExport.hxx
@@ -34,6 +34,13 @@ template<typename T> struct ConstBuffer;
  */
 struct PcmExport {
 	/**
+	 * This buffer is used to reorder channels.
+	 *
+	 * @see #alsa_channel_order
+	 */
+	PcmBuffer order_buffer;
+
+	/**
 	 * The buffer is used to convert DSD samples to the
 	 * DoP format.
 	 *
@@ -61,6 +68,16 @@ struct PcmExport {
 	uint8_t channels;
 
 	/**
+	 * Convert the given buffer from FLAC channel order to ALSA
+	 * channel order using ToAlsaChannelOrder()?
+	 *
+	 * If this value is SampleFormat::UNDEFINED, then no channel
+	 * reordering is applied, otherwise this is the input sample
+	 * format.
+	 */
+	SampleFormat alsa_channel_order;
+
+	/**
 	 * Convert DSD to DSD-over-PCM (DoP)?  Input format must be
 	 * SampleFormat::DSD and output format must be
 	 * SampleFormat::S24_P32.
@@ -96,6 +113,7 @@ struct PcmExport {
 	 * @param channels the number of channels; ignored unless dop is set
 	 */
 	void Open(SampleFormat sample_format, unsigned channels,
+		  bool _alsa_channel_order,
 		  bool dop, bool shift8, bool pack, bool reverse_endian);
 
 	/**