pcm_*: move to src/pcm/

src/pcm/PcmChannels.cxx

@@ -0,0 +1,290 @@
+/*
+ * Copyright (C) 2003-2013 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 "PcmChannels.hxx"
+#include "pcm_buffer.h"
+#include "PcmUtils.hxx"
+
+#include <assert.h>
+
+template<typename D, typename S>
+static void
+MonoToStereo(D dest, S src, S end)
+{
+	while (src != end) {
+		const auto value = *src++;
+
+		*dest++ = value;
+		*dest++ = value;
+	}
+
+}
+
+static void
+pcm_convert_channels_16_2_to_1(int16_t *restrict dest,
+			       const int16_t *restrict src,
+			       const int16_t *restrict src_end)
+{
+	while (src < src_end) {
+		int32_t a = *src++, b = *src++;
+
+		*dest++ = (a + b) / 2;
+	}
+}
+
+static void
+pcm_convert_channels_16_n_to_2(int16_t *restrict dest,
+			       unsigned src_channels,
+			       const int16_t *restrict src,
+			       const int16_t *restrict src_end)
+{
+	unsigned c;
+
+	assert(src_channels > 0);
+
+	while (src < src_end) {
+		int32_t sum = 0;
+		int16_t value;
+
+		for (c = 0; c < src_channels; ++c)
+			sum += *src++;
+		value = sum / (int)src_channels;
+
+		/* XXX this is actually only mono ... */
+		*dest++ = value;
+		*dest++ = value;
+	}
+}
+
+const int16_t *
+pcm_convert_channels_16(struct pcm_buffer *buffer,
+			unsigned dest_channels,
+			unsigned src_channels, const int16_t *src,
+			size_t src_size, size_t *dest_size_r)
+{
+	assert(src_size % (sizeof(*src) * src_channels) == 0);
+
+	size_t dest_size = src_size / src_channels * dest_channels;
+	*dest_size_r = dest_size;
+
+	int16_t *dest = (int16_t *)pcm_buffer_get(buffer, dest_size);
+	const int16_t *src_end = pcm_end_pointer(src, src_size);
+
+	if (src_channels == 1 && dest_channels == 2)
+		MonoToStereo(dest, src, src_end);
+	else if (src_channels == 2 && dest_channels == 1)
+		pcm_convert_channels_16_2_to_1(dest, src, src_end);
+	else if (dest_channels == 2)
+		pcm_convert_channels_16_n_to_2(dest, src_channels, src,
+					       src_end);
+	else
+		return NULL;
+
+	return dest;
+}
+
+static void
+pcm_convert_channels_24_2_to_1(int32_t *restrict dest,
+			       const int32_t *restrict src,
+			       const int32_t *restrict src_end)
+{
+	while (src < src_end) {
+		int32_t a = *src++, b = *src++;
+
+		*dest++ = (a + b) / 2;
+	}
+}
+
+static void
+pcm_convert_channels_24_n_to_2(int32_t *restrict dest,
+			       unsigned src_channels,
+			       const int32_t *restrict src,
+			       const int32_t *restrict src_end)
+{
+	unsigned c;
+
+	assert(src_channels > 0);
+
+	while (src < src_end) {
+		int32_t sum = 0;
+		int32_t value;
+
+		for (c = 0; c < src_channels; ++c)
+			sum += *src++;
+		value = sum / (int)src_channels;
+
+		/* XXX this is actually only mono ... */
+		*dest++ = value;
+		*dest++ = value;
+	}
+}
+
+const int32_t *
+pcm_convert_channels_24(struct pcm_buffer *buffer,
+			unsigned dest_channels,
+			unsigned src_channels, const int32_t *src,
+			size_t src_size, size_t *dest_size_r)
+{
+	assert(src_size % (sizeof(*src) * src_channels) == 0);
+
+	size_t dest_size = src_size / src_channels * dest_channels;
+	*dest_size_r = dest_size;
+
+	int32_t *dest = (int32_t *)pcm_buffer_get(buffer, dest_size);
+	const int32_t *src_end = (const int32_t *)
+		pcm_end_pointer(src, src_size);
+
+	if (src_channels == 1 && dest_channels == 2)
+		MonoToStereo(dest, src, src_end);
+	else if (src_channels == 2 && dest_channels == 1)
+		pcm_convert_channels_24_2_to_1(dest, src, src_end);
+	else if (dest_channels == 2)
+		pcm_convert_channels_24_n_to_2(dest, src_channels, src,
+					       src_end);
+	else
+		return NULL;
+
+	return dest;
+}
+
+static void
+pcm_convert_channels_32_2_to_1(int32_t *restrict dest,
+			       const int32_t *restrict src,
+			       const int32_t *restrict src_end)
+{
+	while (src < src_end) {
+		int64_t a = *src++, b = *src++;
+
+		*dest++ = (a + b) / 2;
+	}
+}
+
+static void
+pcm_convert_channels_32_n_to_2(int32_t *dest,
+			       unsigned src_channels, const int32_t *src,
+			       const int32_t *src_end)
+{
+	unsigned c;
+
+	assert(src_channels > 0);
+
+	while (src < src_end) {
+		int64_t sum = 0;
+		int32_t value;
+
+		for (c = 0; c < src_channels; ++c)
+			sum += *src++;
+		value = sum / (int64_t)src_channels;
+
+		/* XXX this is actually only mono ... */
+		*dest++ = value;
+		*dest++ = value;
+	}
+}
+
+const int32_t *
+pcm_convert_channels_32(struct pcm_buffer *buffer,
+			unsigned dest_channels,
+			unsigned src_channels, const int32_t *src,
+			size_t src_size, size_t *dest_size_r)
+{
+	assert(src_size % (sizeof(*src) * src_channels) == 0);
+
+	size_t dest_size = src_size / src_channels * dest_channels;
+	*dest_size_r = dest_size;
+
+	int32_t *dest = (int32_t *)pcm_buffer_get(buffer, dest_size);
+	const int32_t *src_end = (const int32_t *)
+		pcm_end_pointer(src, src_size);
+
+	if (src_channels == 1 && dest_channels == 2)
+		MonoToStereo(dest, src, src_end);
+	else if (src_channels == 2 && dest_channels == 1)
+		pcm_convert_channels_32_2_to_1(dest, src, src_end);
+	else if (dest_channels == 2)
+		pcm_convert_channels_32_n_to_2(dest, src_channels, src,
+					       src_end);
+	else
+		return NULL;
+
+	return dest;
+}
+
+static void
+pcm_convert_channels_float_2_to_1(float *restrict dest,
+				  const float *restrict src,
+				  const float *restrict src_end)
+{
+	while (src < src_end) {
+		double a = *src++, b = *src++;
+
+		*dest++ = (a + b) / 2;
+	}
+}
+
+static void
+pcm_convert_channels_float_n_to_2(float *dest,
+				  unsigned src_channels, const float *src,
+				  const float *src_end)
+{
+	unsigned c;
+
+	assert(src_channels > 0);
+
+	while (src < src_end) {
+		double sum = 0;
+		float value;
+
+		for (c = 0; c < src_channels; ++c)
+			sum += *src++;
+		value = sum / (double)src_channels;
+
+		/* XXX this is actually only mono ... */
+		*dest++ = value;
+		*dest++ = value;
+	}
+}
+
+const float *
+pcm_convert_channels_float(struct pcm_buffer *buffer,
+			   unsigned dest_channels,
+			   unsigned src_channels, const float *src,
+			   size_t src_size, size_t *dest_size_r)
+{
+	assert(src_size % (sizeof(*src) * src_channels) == 0);
+
+	size_t dest_size = src_size / src_channels * dest_channels;
+	*dest_size_r = dest_size;
+
+	float *dest = (float *)pcm_buffer_get(buffer, dest_size);
+	const float *src_end = (const float *)pcm_end_pointer(src, src_size);
+
+	if (src_channels == 1 && dest_channels == 2)
+		MonoToStereo(dest, src, src_end);
+	else if (src_channels == 2 && dest_channels == 1)
+		pcm_convert_channels_float_2_to_1(dest, src, src_end);
+	else if (dest_channels == 2)
+		pcm_convert_channels_float_n_to_2(dest, src_channels, src,
+						  src_end);
+	else
+		return NULL;
+
+	return dest;
+}

src/pcm/PcmChannels.hxx

@@ -0,0 +1,97 @@
+/*
+ * Copyright (C) 2003-2013 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_CHANNELS_HXX
+#define MPD_PCM_CHANNELS_HXX
+
+#include <stdint.h>
+#include <stddef.h>
+
+struct pcm_buffer;
+
+/**
+ * Changes the number of channels in 16 bit PCM data.
+ *
+ * @param buffer the destination pcm_buffer object
+ * @param dest_channels the number of channels requested
+ * @param src_channels the number of channels in the source buffer
+ * @param src the source PCM buffer
+ * @param src_size the number of bytes in #src
+ * @param dest_size_r returns the number of bytes of the destination buffer
+ * @return the destination buffer
+ */
+const int16_t *
+pcm_convert_channels_16(struct pcm_buffer *buffer,
+			unsigned dest_channels,
+			unsigned src_channels, const int16_t *src,
+			size_t src_size, size_t *dest_size_r);
+
+/**
+ * Changes the number of channels in 24 bit PCM data (aligned at 32
+ * bit boundaries).
+ *
+ * @param buffer the destination pcm_buffer object
+ * @param dest_channels the number of channels requested
+ * @param src_channels the number of channels in the source buffer
+ * @param src the source PCM buffer
+ * @param src_size the number of bytes in #src
+ * @param dest_size_r returns the number of bytes of the destination buffer
+ * @return the destination buffer
+ */
+const int32_t *
+pcm_convert_channels_24(struct pcm_buffer *buffer,
+			unsigned dest_channels,
+			unsigned src_channels, const int32_t *src,
+			size_t src_size, size_t *dest_size_r);
+
+/**
+ * Changes the number of channels in 32 bit PCM data.
+ *
+ * @param buffer the destination pcm_buffer object
+ * @param dest_channels the number of channels requested
+ * @param src_channels the number of channels in the source buffer
+ * @param src the source PCM buffer
+ * @param src_size the number of bytes in #src
+ * @param dest_size_r returns the number of bytes of the destination buffer
+ * @return the destination buffer
+ */
+const int32_t *
+pcm_convert_channels_32(struct pcm_buffer *buffer,
+			unsigned dest_channels,
+			unsigned src_channels, const int32_t *src,
+			size_t src_size, size_t *dest_size_r);
+
+/**
+ * Changes the number of channels in 32 bit float PCM data.
+ *
+ * @param buffer the destination pcm_buffer object
+ * @param dest_channels the number of channels requested
+ * @param src_channels the number of channels in the source buffer
+ * @param src the source PCM buffer
+ * @param src_size the number of bytes in #src
+ * @param dest_size_r returns the number of bytes of the destination buffer
+ * @return the destination buffer
+ */
+const float *
+pcm_convert_channels_float(struct pcm_buffer *buffer,
+			   unsigned dest_channels,
+			   unsigned src_channels, const float *src,
+			   size_t src_size, size_t *dest_size_r);
+
+#endif

src/pcm/PcmConvert.cxx

@@ -0,0 +1,326 @@
+/*
+ * 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 "PcmConvert.hxx"
+#include "PcmChannels.hxx"
+#include "PcmFormat.hxx"
+#include "pcm_pack.h"
+#include "audio_format.h"
+
+#include <assert.h>
+#include <string.h>
+#include <math.h>
+#include <glib.h>
+
+#undef G_LOG_DOMAIN
+#define G_LOG_DOMAIN "pcm"
+
+PcmConvert::PcmConvert()
+{
+	memset(this, 0, sizeof(*this));
+
+	pcm_dsd_init(&dsd);
+	pcm_resample_init(&resample);
+
+	pcm_buffer_init(&format_buffer);
+	pcm_buffer_init(&channels_buffer);
+}
+
+PcmConvert::~PcmConvert()
+{
+	pcm_dsd_deinit(&dsd);
+	pcm_resample_deinit(&resample);
+
+	pcm_buffer_deinit(&format_buffer);
+	pcm_buffer_deinit(&channels_buffer);
+}
+
+void
+PcmConvert::Reset()
+{
+	pcm_dsd_reset(&dsd);
+	pcm_resample_reset(&resample);
+}
+
+inline const int16_t *
+PcmConvert::Convert16(const audio_format *src_format,
+		      const void *src_buffer, size_t src_size,
+		      const audio_format *dest_format, size_t *dest_size_r,
+		      GError **error_r)
+{
+	const int16_t *buf;
+	size_t len;
+
+	assert(dest_format->format == SAMPLE_FORMAT_S16);
+
+	buf = pcm_convert_to_16(&format_buffer, dither,
+				sample_format(src_format->format),
+				src_buffer, src_size,
+				&len);
+	if (buf == NULL) {
+		g_set_error(error_r, pcm_convert_quark(), 0,
+			    "Conversion from %s to 16 bit is not implemented",
+			    sample_format_to_string(sample_format(src_format->format)));
+		return NULL;
+	}
+
+	if (src_format->channels != dest_format->channels) {
+		buf = pcm_convert_channels_16(&channels_buffer,
+					      dest_format->channels,
+					      src_format->channels,
+					      buf, len, &len);
+		if (buf == NULL) {
+			g_set_error(error_r, pcm_convert_quark(), 0,
+				    "Conversion from %u to %u channels "
+				    "is not implemented",
+				    src_format->channels,
+				    dest_format->channels);
+			return NULL;
+		}
+	}
+
+	if (src_format->sample_rate != dest_format->sample_rate) {
+		buf = pcm_resample_16(&resample,
+				      dest_format->channels,
+				      src_format->sample_rate, buf, len,
+				      dest_format->sample_rate, &len,
+				      error_r);
+		if (buf == NULL)
+			return NULL;
+	}
+
+	*dest_size_r = len;
+	return buf;
+}
+
+inline const int32_t *
+PcmConvert::Convert24(const audio_format *src_format,
+		      const void *src_buffer, size_t src_size,
+		      const audio_format *dest_format, size_t *dest_size_r,
+		      GError **error_r)
+{
+	const int32_t *buf;
+	size_t len;
+
+	assert(dest_format->format == SAMPLE_FORMAT_S24_P32);
+
+	buf = pcm_convert_to_24(&format_buffer,
+				sample_format(src_format->format),
+				src_buffer, src_size, &len);
+	if (buf == NULL) {
+		g_set_error(error_r, pcm_convert_quark(), 0,
+			    "Conversion from %s to 24 bit is not implemented",
+			    sample_format_to_string(sample_format(src_format->format)));
+		return NULL;
+	}
+
+	if (src_format->channels != dest_format->channels) {
+		buf = pcm_convert_channels_24(&channels_buffer,
+					      dest_format->channels,
+					      src_format->channels,
+					      buf, len, &len);
+		if (buf == NULL) {
+			g_set_error(error_r, pcm_convert_quark(), 0,
+				    "Conversion from %u to %u channels "
+				    "is not implemented",
+				    src_format->channels,
+				    dest_format->channels);
+			return NULL;
+		}
+	}
+
+	if (src_format->sample_rate != dest_format->sample_rate) {
+		buf = pcm_resample_24(&resample,
+				      dest_format->channels,
+				      src_format->sample_rate, buf, len,
+				      dest_format->sample_rate, &len,
+				      error_r);
+		if (buf == NULL)
+			return NULL;
+	}
+
+	*dest_size_r = len;
+	return buf;
+}
+
+inline const int32_t *
+PcmConvert::Convert32(const audio_format *src_format,
+		      const void *src_buffer, size_t src_size,
+		      const audio_format *dest_format, size_t *dest_size_r,
+		      GError **error_r)
+{
+	const int32_t *buf;
+	size_t len;
+
+	assert(dest_format->format == SAMPLE_FORMAT_S32);
+
+	buf = pcm_convert_to_32(&format_buffer,
+				sample_format(src_format->format),
+				src_buffer, src_size, &len);
+	if (buf == NULL) {
+		g_set_error(error_r, pcm_convert_quark(), 0,
+			    "Conversion from %s to 32 bit is not implemented",
+			    sample_format_to_string(sample_format(src_format->format)));
+		return NULL;
+	}
+
+	if (src_format->channels != dest_format->channels) {
+		buf = pcm_convert_channels_32(&channels_buffer,
+					      dest_format->channels,
+					      src_format->channels,
+					      buf, len, &len);
+		if (buf == NULL) {
+			g_set_error(error_r, pcm_convert_quark(), 0,
+				    "Conversion from %u to %u channels "
+				    "is not implemented",
+				    src_format->channels,
+				    dest_format->channels);
+			return NULL;
+		}
+	}
+
+	if (src_format->sample_rate != dest_format->sample_rate) {
+		buf = pcm_resample_32(&resample,
+				      dest_format->channels,
+				      src_format->sample_rate, buf, len,
+				      dest_format->sample_rate, &len,
+				      error_r);
+		if (buf == NULL)
+			return buf;
+	}
+
+	*dest_size_r = len;
+	return buf;
+}
+
+inline const float *
+PcmConvert::ConvertFloat(const audio_format *src_format,
+			 const void *src_buffer, size_t src_size,
+			 const audio_format *dest_format, size_t *dest_size_r,
+			 GError **error_r)
+{
+	const float *buffer = (const float *)src_buffer;
+	size_t size = src_size;
+
+	assert(dest_format->format == SAMPLE_FORMAT_FLOAT);
+
+	/* convert to float now */
+
+	buffer = pcm_convert_to_float(&format_buffer,
+				      sample_format(src_format->format),
+				      buffer, size, &size);
+	if (buffer == NULL) {
+		g_set_error(error_r, pcm_convert_quark(), 0,
+			    "Conversion from %s to float is not implemented",
+			    sample_format_to_string(sample_format(src_format->format)));
+		return NULL;
+	}
+
+	/* convert channels */
+
+	if (src_format->channels != dest_format->channels) {
+		buffer = pcm_convert_channels_float(&channels_buffer,
+						    dest_format->channels,
+						    src_format->channels,
+						    buffer, size, &size);
+		if (buffer == NULL) {
+			g_set_error(error_r, pcm_convert_quark(), 0,
+				    "Conversion from %u to %u channels "
+				    "is not implemented",
+				    src_format->channels,
+				    dest_format->channels);
+			return NULL;
+		}
+	}
+
+	/* resample with float, because this is the best format for
+	   libsamplerate */
+
+	if (src_format->sample_rate != dest_format->sample_rate) {
+		buffer = pcm_resample_float(&resample,
+					    dest_format->channels,
+					    src_format->sample_rate,
+					    buffer, size,
+					    dest_format->sample_rate, &size,
+					    error_r);
+		if (buffer == NULL)
+			return NULL;
+	}
+
+	*dest_size_r = size;
+	return buffer;
+}
+
+const void *
+PcmConvert::Convert(const audio_format *src_format,
+		    const void *src, size_t src_size,
+		    const audio_format *dest_format,
+		    size_t *dest_size_r,
+		    GError **error_r)
+{
+	struct audio_format float_format;
+	if (src_format->format == SAMPLE_FORMAT_DSD) {
+		size_t f_size;
+		const float *f = pcm_dsd_to_float(&dsd,
+						  src_format->channels,
+						  false, (const uint8_t *)src,
+						  src_size, &f_size);
+		if (f == NULL) {
+			g_set_error_literal(error_r, pcm_convert_quark(), 0,
+					    "DSD to PCM conversion failed");
+			return NULL;
+		}
+
+		float_format = *src_format;
+		float_format.format = SAMPLE_FORMAT_FLOAT;
+
+		src_format = &float_format;
+		src = f;
+		src_size = f_size;
+	}
+
+	switch (sample_format(dest_format->format)) {
+	case SAMPLE_FORMAT_S16:
+		return Convert16(src_format, src, src_size,
+				 dest_format, dest_size_r,
+				 error_r);
+
+	case SAMPLE_FORMAT_S24_P32:
+		return Convert24(src_format, src, src_size,
+				 dest_format, dest_size_r,
+				 error_r);
+
+	case SAMPLE_FORMAT_S32:
+		return Convert32(src_format, src, src_size,
+				 dest_format, dest_size_r,
+				 error_r);
+
+	case SAMPLE_FORMAT_FLOAT:
+		return ConvertFloat(src_format, src, src_size,
+				    dest_format, dest_size_r,
+				    error_r);
+
+	default:
+		g_set_error(error_r, pcm_convert_quark(), 0,
+			    "PCM conversion to %s is not implemented",
+			    sample_format_to_string(sample_format(dest_format->format)));
+		return NULL;
+	}
+}

src/pcm/PcmConvert.hxx

@@ -0,0 +1,115 @@
+/*
+ * Copyright (C) 2003-2013 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 PCM_CONVERT_HXX
+#define PCM_CONVERT_HXX
+
+#include "PcmDither.hxx"
+
+extern "C" {
+#include "pcm_dsd.h"
+#include "pcm_resample.h"
+#include "pcm_buffer.h"
+}
+
+#include <glib.h>
+
+struct audio_format;
+
+/**
+ * This object is statically allocated (within another struct), and
+ * holds buffer allocations and the state for all kinds of PCM
+ * conversions.
+ */
+class PcmConvert {
+	struct pcm_dsd dsd;
+
+	struct pcm_resample_state resample;
+
+	PcmDither dither;
+
+	/** the buffer for converting the sample format */
+	struct pcm_buffer format_buffer;
+
+	/** the buffer for converting the channel count */
+	struct pcm_buffer channels_buffer;
+
+public:
+	PcmConvert();
+	~PcmConvert();
+
+
+	/**
+	 * Reset the pcm_convert_state object.  Use this at the
+	 * boundary between two distinct songs and each time the
+	 * format changes.
+	 */
+	void Reset();
+
+	/**
+	 * Converts PCM data between two audio formats.
+	 *
+	 * @param src_format the source audio format
+	 * @param src the source PCM buffer
+	 * @param src_size the size of #src in bytes
+	 * @param dest_format the requested destination audio format
+	 * @param dest_size_r returns the number of bytes of the destination buffer
+	 * @param error_r location to store the error occurring, or NULL to
+	 * ignore errors
+	 * @return the destination buffer, or NULL on error
+	 */
+	const void *Convert(const audio_format *src_format,
+			    const void *src, size_t src_size,
+			    const audio_format *dest_format,
+			    size_t *dest_size_r,
+			    GError **error_r);
+
+private:
+	const int16_t *Convert16(const audio_format *src_format,
+				 const void *src_buffer, size_t src_size,
+				 const audio_format *dest_format,
+				 size_t *dest_size_r,
+				 GError **error_r);
+
+	const int32_t *Convert24(const audio_format *src_format,
+				 const void *src_buffer, size_t src_size,
+				 const audio_format *dest_format,
+				 size_t *dest_size_r,
+				 GError **error_r);
+
+	const int32_t *Convert32(const audio_format *src_format,
+				 const void *src_buffer, size_t src_size,
+				 const audio_format *dest_format,
+				 size_t *dest_size_r,
+				 GError **error_r);
+
+	const float *ConvertFloat(const audio_format *src_format,
+				  const void *src_buffer, size_t src_size,
+				  const audio_format *dest_format,
+				  size_t *dest_size_r,
+				  GError **error_r);
+};
+
+static inline GQuark
+pcm_convert_quark(void)
+{
+	return g_quark_from_static_string("pcm_convert");
+}
+
+#endif

src/pcm/PcmDither.cxx

@@ -0,0 +1,89 @@
+/*
+ * Copyright (C) 2003-2013 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 "PcmDither.hxx"
+#include "PcmPrng.hxx"
+
+inline int16_t
+PcmDither::Dither24To16(int_fast32_t sample)
+{
+	constexpr unsigned from_bits = 24;
+	constexpr unsigned to_bits = 16;
+	constexpr unsigned scale_bits = from_bits - to_bits;
+	constexpr int_fast32_t round = 1 << (scale_bits - 1);
+	constexpr int_fast32_t mask = (1 << scale_bits) - 1;
+	constexpr int_fast32_t ONE = 1 << (from_bits - 1);
+	constexpr int_fast32_t MIN = -ONE;
+	constexpr int_fast32_t MAX = ONE - 1;
+
+	sample += error[0] - error[1] + error[2];
+
+	error[2] = error[1];
+	error[1] = error[0] / 2;
+
+	/* round */
+	int_fast32_t output = sample + round;
+
+	int_fast32_t rnd = pcm_prng(random);
+	output += (rnd & mask) - (random & mask);
+
+	random = rnd;
+
+	/* clip */
+	if (output > MAX) {
+		output = MAX;
+
+		if (sample > MAX)
+			sample = MAX;
+	} else if (output < MIN) {
+		output = MIN;
+
+		if (sample < MIN)
+			sample = MIN;
+	}
+
+	output &= ~mask;
+
+	error[0] = sample - output;
+
+	return (int16_t)(output >> scale_bits);
+}
+
+void
+PcmDither::Dither24To16(int16_t *dest, const int32_t *src,
+			const int32_t *src_end)
+{
+	while (src < src_end)
+		*dest++ = Dither24To16(*src++);
+}
+
+inline int16_t
+PcmDither::Dither32To16(int_fast32_t sample)
+{
+	return Dither24To16(sample >> 8);
+}
+
+void
+PcmDither::Dither32To16(int16_t *dest, const int32_t *src,
+			const int32_t *src_end)
+{
+	while (src < src_end)
+		*dest++ = Dither32To16(*src++);
+}

src/pcm/PcmDither.hxx

@@ -0,0 +1,44 @@
+/*
+ * Copyright (C) 2003-2013 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_DITHER_HXX
+#define MPD_PCM_DITHER_HXX
+
+#include <stdint.h>
+
+class PcmDither {
+	int32_t error[3];
+	int32_t random;
+
+public:
+	constexpr PcmDither()
+		:error{0, 0, 0}, random(0) {}
+
+	void Dither24To16(int16_t *dest, const int32_t *src,
+			  const int32_t *src_end);
+
+	void Dither32To16(int16_t *dest, const int32_t *src,
+			  const int32_t *src_end);
+
+private:
+	int16_t Dither24To16(int_fast32_t sample);
+	int16_t Dither32To16(int_fast32_t sample);
+};
+
+#endif

src/pcm/PcmFormat.cxx

@@ -0,0 +1,500 @@
+/*
+ * 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 "PcmFormat.hxx"
+#include "PcmDither.hxx"
+#include "pcm_buffer.h"
+#include "pcm_pack.h"
+#include "PcmUtils.hxx"
+
+#include <type_traits>
+
+template<typename S>
+struct DefaultSampleBits {
+	typedef decltype(*S()) T;
+	typedef typename std::remove_reference<T>::type U;
+
+	static constexpr auto value = sizeof(U) * 8;
+};
+
+static void
+pcm_convert_8_to_16(int16_t *out, const int8_t *in, const int8_t *in_end)
+{
+	while (in < in_end) {
+		*out++ = *in++ << 8;
+	}
+}
+
+static void
+pcm_convert_24_to_16(PcmDither &dither,
+		     int16_t *out, const int32_t *in, const int32_t *in_end)
+{
+	dither.Dither24To16(out, in, in_end);
+}
+
+static void
+pcm_convert_32_to_16(PcmDither &dither,
+		     int16_t *out, const int32_t *in, const int32_t *in_end)
+{
+	dither.Dither32To16(out, in, in_end);
+}
+
+template<typename S, unsigned bits=DefaultSampleBits<S>::value>
+static void
+ConvertFromFloat(S dest, const float *src, const float *end)
+{
+	typedef decltype(*S()) T;
+	typedef typename std::remove_reference<T>::type U;
+
+	const float factor = 1 << (bits - 1);
+
+	while (src != end) {
+		int sample(*src++ * factor);
+		*dest++ = PcmClamp<U, int, bits>(sample);
+	}
+}
+
+template<typename S, unsigned bits=DefaultSampleBits<S>::value>
+static void
+ConvertFromFloat(S dest, const float *src, size_t size)
+{
+	ConvertFromFloat<S, bits>(dest, src, pcm_end_pointer(src, size));
+}
+
+template<typename S, unsigned bits=sizeof(S)*8>
+static S *
+AllocateFromFloat(pcm_buffer &buffer, const float *src, size_t src_size,
+		  size_t *dest_size_r)
+{
+	constexpr size_t src_sample_size = sizeof(*src);
+	assert(src_size % src_sample_size == 0);
+
+	const size_t num_samples = src_size / src_sample_size;
+	*dest_size_r = num_samples * sizeof(S);
+	S *dest = (S *)pcm_buffer_get(&buffer, *dest_size_r);
+	ConvertFromFloat<S *, bits>(dest, src, src_size);
+	return dest;
+}
+
+static int16_t *
+pcm_allocate_8_to_16(struct pcm_buffer *buffer,
+		     const int8_t *src, size_t src_size, size_t *dest_size_r)
+{
+	int16_t *dest;
+	*dest_size_r = src_size / sizeof(*src) * sizeof(*dest);
+	dest = (int16_t *)pcm_buffer_get(buffer, *dest_size_r);
+	pcm_convert_8_to_16(dest, src, pcm_end_pointer(src, src_size));
+	return dest;
+}
+
+static int16_t *
+pcm_allocate_24p32_to_16(struct pcm_buffer *buffer, PcmDither &dither,
+			 const int32_t *src, size_t src_size,
+			 size_t *dest_size_r)
+{
+	int16_t *dest;
+	*dest_size_r = src_size / 2;
+	assert(*dest_size_r == src_size / sizeof(*src) * sizeof(*dest));
+	dest = (int16_t *)pcm_buffer_get(buffer, *dest_size_r);
+	pcm_convert_24_to_16(dither, dest, src,
+			     pcm_end_pointer(src, src_size));
+	return dest;
+}
+
+static int16_t *
+pcm_allocate_32_to_16(struct pcm_buffer *buffer, PcmDither &dither,
+		      const int32_t *src, size_t src_size,
+		      size_t *dest_size_r)
+{
+	int16_t *dest;
+	*dest_size_r = src_size / 2;
+	assert(*dest_size_r == src_size / sizeof(*src) * sizeof(*dest));
+	dest = (int16_t *)pcm_buffer_get(buffer, *dest_size_r);
+	pcm_convert_32_to_16(dither, dest, src,
+			     pcm_end_pointer(src, src_size));
+	return dest;
+}
+
+static int16_t *
+pcm_allocate_float_to_16(struct pcm_buffer *buffer,
+			 const float *src, size_t src_size,
+			 size_t *dest_size_r)
+{
+	return AllocateFromFloat<int16_t>(*buffer, src, src_size, dest_size_r);
+}
+
+const int16_t *
+pcm_convert_to_16(struct pcm_buffer *buffer, PcmDither &dither,
+		  enum sample_format src_format, const void *src,
+		  size_t src_size, size_t *dest_size_r)
+{
+	assert(src_size % sample_format_size(src_format) == 0);
+
+	switch (src_format) {
+	case SAMPLE_FORMAT_UNDEFINED:
+	case SAMPLE_FORMAT_DSD:
+		break;
+
+	case SAMPLE_FORMAT_S8:
+		return pcm_allocate_8_to_16(buffer,
+					    (const int8_t *)src, src_size,
+					    dest_size_r);
+
+	case SAMPLE_FORMAT_S16:
+		*dest_size_r = src_size;
+		return (const int16_t *)src;
+
+	case SAMPLE_FORMAT_S24_P32:
+		return pcm_allocate_24p32_to_16(buffer, dither,
+						(const int32_t *)src, src_size,
+						dest_size_r);
+
+	case SAMPLE_FORMAT_S32:
+		return pcm_allocate_32_to_16(buffer, dither,
+					     (const int32_t *)src, src_size,
+					     dest_size_r);
+
+	case SAMPLE_FORMAT_FLOAT:
+		return pcm_allocate_float_to_16(buffer,
+						(const float *)src, src_size,
+						dest_size_r);
+	}
+
+	return NULL;
+}
+
+static void
+pcm_convert_8_to_24(int32_t *out, const int8_t *in, const int8_t *in_end)
+{
+	while (in < in_end)
+		*out++ = *in++ << 16;
+}
+
+static void
+pcm_convert_16_to_24(int32_t *out, const int16_t *in, const int16_t *in_end)
+{
+	while (in < in_end)
+		*out++ = *in++ << 8;
+}
+
+static void
+pcm_convert_32_to_24(int32_t *restrict out,
+		     const int32_t *restrict in,
+		     const int32_t *restrict in_end)
+{
+	while (in < in_end)
+		*out++ = *in++ >> 8;
+}
+
+static int32_t *
+pcm_allocate_8_to_24(struct pcm_buffer *buffer,
+		     const int8_t *src, size_t src_size, size_t *dest_size_r)
+{
+	int32_t *dest;
+	*dest_size_r = src_size / sizeof(*src) * sizeof(*dest);
+	dest = (int32_t *)pcm_buffer_get(buffer, *dest_size_r);
+	pcm_convert_8_to_24(dest, src, pcm_end_pointer(src, src_size));
+	return dest;
+}
+
+static int32_t *
+pcm_allocate_16_to_24(struct pcm_buffer *buffer,
+		      const int16_t *src, size_t src_size, size_t *dest_size_r)
+{
+	int32_t *dest;
+	*dest_size_r = src_size * 2;
+	assert(*dest_size_r == src_size / sizeof(*src) * sizeof(*dest));
+	dest = (int32_t *)pcm_buffer_get(buffer, *dest_size_r);
+	pcm_convert_16_to_24(dest, src, pcm_end_pointer(src, src_size));
+	return dest;
+}
+
+static int32_t *
+pcm_allocate_32_to_24(struct pcm_buffer *buffer,
+		      const int32_t *src, size_t src_size, size_t *dest_size_r)
+{
+	*dest_size_r = src_size;
+	int32_t *dest = (int32_t *)pcm_buffer_get(buffer, *dest_size_r);
+	pcm_convert_32_to_24(dest, src, pcm_end_pointer(src, src_size));
+	return dest;
+}
+
+static int32_t *
+pcm_allocate_float_to_24(struct pcm_buffer *buffer,
+			 const float *src, size_t src_size,
+			 size_t *dest_size_r)
+{
+	return AllocateFromFloat<int32_t, 24>(*buffer, src, src_size,
+					      dest_size_r);
+}
+
+const int32_t *
+pcm_convert_to_24(struct pcm_buffer *buffer,
+		  enum sample_format src_format, const void *src,
+		  size_t src_size, size_t *dest_size_r)
+{
+	assert(src_size % sample_format_size(src_format) == 0);
+
+	switch (src_format) {
+	case SAMPLE_FORMAT_UNDEFINED:
+	case SAMPLE_FORMAT_DSD:
+		break;
+
+	case SAMPLE_FORMAT_S8:
+		return pcm_allocate_8_to_24(buffer,
+					    (const int8_t *)src, src_size,
+					    dest_size_r);
+
+	case SAMPLE_FORMAT_S16:
+		return pcm_allocate_16_to_24(buffer,
+					     (const int16_t *)src, src_size,
+					     dest_size_r);
+
+	case SAMPLE_FORMAT_S24_P32:
+		*dest_size_r = src_size;
+		return (const int32_t *)src;
+
+	case SAMPLE_FORMAT_S32:
+		return pcm_allocate_32_to_24(buffer,
+					     (const int32_t *)src, src_size,
+					     dest_size_r);
+
+	case SAMPLE_FORMAT_FLOAT:
+		return pcm_allocate_float_to_24(buffer,
+						(const float *)src, src_size,
+						dest_size_r);
+	}
+
+	return NULL;
+}
+
+static void
+pcm_convert_8_to_32(int32_t *out, const int8_t *in, const int8_t *in_end)
+{
+	while (in < in_end)
+		*out++ = *in++ << 24;
+}
+
+static void
+pcm_convert_16_to_32(int32_t *out, const int16_t *in, const int16_t *in_end)
+{
+	while (in < in_end)
+		*out++ = *in++ << 16;
+}
+
+static void
+pcm_convert_24_to_32(int32_t *restrict out,
+		     const int32_t *restrict in,
+		     const int32_t *restrict in_end)
+{
+	while (in < in_end)
+		*out++ = *in++ << 8;
+}
+
+static int32_t *
+pcm_allocate_8_to_32(struct pcm_buffer *buffer,
+		     const int8_t *src, size_t src_size, size_t *dest_size_r)
+{
+	int32_t *dest;
+	*dest_size_r = src_size / sizeof(*src) * sizeof(*dest);
+	dest = (int32_t *)pcm_buffer_get(buffer, *dest_size_r);
+	pcm_convert_8_to_32(dest, src, pcm_end_pointer(src, src_size));
+	return dest;
+}
+
+static int32_t *
+pcm_allocate_16_to_32(struct pcm_buffer *buffer,
+		      const int16_t *src, size_t src_size, size_t *dest_size_r)
+{
+	int32_t *dest;
+	*dest_size_r = src_size * 2;
+	assert(*dest_size_r == src_size / sizeof(*src) * sizeof(*dest));
+	dest = (int32_t *)pcm_buffer_get(buffer, *dest_size_r);
+	pcm_convert_16_to_32(dest, src, pcm_end_pointer(src, src_size));
+	return dest;
+}
+
+static int32_t *
+pcm_allocate_24p32_to_32(struct pcm_buffer *buffer,
+			 const int32_t *src, size_t src_size,
+			 size_t *dest_size_r)
+{
+	*dest_size_r = src_size;
+	int32_t *dest = (int32_t *)pcm_buffer_get(buffer, *dest_size_r);
+	pcm_convert_24_to_32(dest, src, pcm_end_pointer(src, src_size));
+	return dest;
+}
+
+static int32_t *
+pcm_allocate_float_to_32(struct pcm_buffer *buffer,
+			 const float *src, size_t src_size,
+			 size_t *dest_size_r)
+{
+	/* convert to S24_P32 first */
+	int32_t *dest = pcm_allocate_float_to_24(buffer, src, src_size,
+						 dest_size_r);
+
+	/* convert to 32 bit in-place */
+	pcm_convert_24_to_32(dest, dest, pcm_end_pointer(dest, *dest_size_r));
+	return dest;
+}
+
+const int32_t *
+pcm_convert_to_32(struct pcm_buffer *buffer,
+		  enum sample_format src_format, const void *src,
+		  size_t src_size, size_t *dest_size_r)
+{
+	assert(src_size % sample_format_size(src_format) == 0);
+
+	switch (src_format) {
+	case SAMPLE_FORMAT_UNDEFINED:
+	case SAMPLE_FORMAT_DSD:
+		break;
+
+	case SAMPLE_FORMAT_S8:
+		return pcm_allocate_8_to_32(buffer,
+					    (const int8_t *)src, src_size,
+					    dest_size_r);
+
+	case SAMPLE_FORMAT_S16:
+		return pcm_allocate_16_to_32(buffer,
+					     (const int16_t *)src, src_size,
+					     dest_size_r);
+
+	case SAMPLE_FORMAT_S24_P32:
+		return pcm_allocate_24p32_to_32(buffer,
+						(const int32_t *)src, src_size,
+						dest_size_r);
+
+	case SAMPLE_FORMAT_S32:
+		*dest_size_r = src_size;
+		return (const int32_t *)src;
+
+	case SAMPLE_FORMAT_FLOAT:
+		return pcm_allocate_float_to_32(buffer,
+						(const float *)src, src_size,
+						dest_size_r);
+	}
+
+	return NULL;
+}
+
+template<typename S, unsigned bits=DefaultSampleBits<S>::value>
+static void
+ConvertToFloat(float *dest, S src, S end)
+{
+	constexpr float factor = 0.5 / (1 << (bits - 2));
+	while (src != end)
+		*dest++ = float(*src++) * factor;
+
+}
+
+template<typename S, unsigned bits=DefaultSampleBits<S>::value>
+static void
+ConvertToFloat(float *dest, S src, size_t size)
+{
+	ConvertToFloat<S, bits>(dest, src, pcm_end_pointer(src, size));
+}
+
+template<typename S, unsigned bits=DefaultSampleBits<S>::value>
+static float *
+AllocateToFloat(pcm_buffer &buffer, S src, size_t src_size,
+		size_t *dest_size_r)
+{
+	constexpr size_t src_sample_size = sizeof(*S());
+	assert(src_size % src_sample_size == 0);
+
+	const size_t num_samples = src_size / src_sample_size;
+	*dest_size_r = num_samples * sizeof(float);
+	float *dest = (float *)pcm_buffer_get(&buffer, *dest_size_r);
+	ConvertToFloat<S, bits>(dest, src, src_size);
+	return dest;
+}
+
+static float *
+pcm_allocate_8_to_float(struct pcm_buffer *buffer,
+			const int8_t *src, size_t src_size,
+			size_t *dest_size_r)
+{
+	return AllocateToFloat(*buffer, src, src_size, dest_size_r);
+}
+
+static float *
+pcm_allocate_16_to_float(struct pcm_buffer *buffer,
+			 const int16_t *src, size_t src_size,
+			 size_t *dest_size_r)
+{
+	return AllocateToFloat(*buffer, src, src_size, dest_size_r);
+}
+
+static float *
+pcm_allocate_24p32_to_float(struct pcm_buffer *buffer,
+			    const int32_t *src, size_t src_size,
+			    size_t *dest_size_r)
+{
+	return AllocateToFloat<decltype(src), 24>
+		(*buffer, src, src_size, dest_size_r);
+}
+
+static float *
+pcm_allocate_32_to_float(struct pcm_buffer *buffer,
+			 const int32_t *src, size_t src_size,
+			 size_t *dest_size_r)
+{
+	return AllocateToFloat(*buffer, src, src_size, dest_size_r);
+}
+
+const float *
+pcm_convert_to_float(struct pcm_buffer *buffer,
+		     enum sample_format src_format, const void *src,
+		     size_t src_size, size_t *dest_size_r)
+{
+	switch (src_format) {
+	case SAMPLE_FORMAT_UNDEFINED:
+	case SAMPLE_FORMAT_DSD:
+		break;
+
+	case SAMPLE_FORMAT_S8:
+		return pcm_allocate_8_to_float(buffer,
+					       (const int8_t *)src, src_size,
+					       dest_size_r);
+
+	case SAMPLE_FORMAT_S16:
+		return pcm_allocate_16_to_float(buffer,
+						(const int16_t *)src, src_size,
+						dest_size_r);
+
+	case SAMPLE_FORMAT_S24_P32:
+		return pcm_allocate_24p32_to_float(buffer,
+						   (const int32_t *)src, src_size,
+						   dest_size_r);
+
+	case SAMPLE_FORMAT_S32:
+		return pcm_allocate_32_to_float(buffer,
+						(const int32_t *)src, src_size,
+						dest_size_r);
+
+	case SAMPLE_FORMAT_FLOAT:
+		*dest_size_r = src_size;
+		return (const float *)src;
+	}
+
+	return NULL;
+}

src/pcm/PcmFormat.hxx

@@ -0,0 +1,93 @@
+/*
+ * Copyright (C) 2003-2013 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_FORMAT_HXX
+#define MPD_PCM_FORMAT_HXX
+
+#include "audio_format.h"
+
+#include <stdint.h>
+#include <stddef.h>
+
+struct pcm_buffer;
+class PcmDither;
+
+/**
+ * Converts PCM samples to 16 bit.  If the source format is 24 bit,
+ * then dithering is applied.
+ *
+ * @param buffer a pcm_buffer object
+ * @param dither a pcm_dither object for 24-to-16 conversion
+ * @param bits the number of in the source buffer
+ * @param src the source PCM buffer
+ * @param src_size the size of #src in bytes
+ * @param dest_size_r returns the number of bytes of the destination buffer
+ * @return the destination buffer
+ */
+const int16_t *
+pcm_convert_to_16(struct pcm_buffer *buffer, PcmDither &dither,
+		  enum sample_format src_format, const void *src,
+		  size_t src_size, size_t *dest_size_r);
+
+/**
+ * Converts PCM samples to 24 bit (32 bit alignment).
+ *
+ * @param buffer a pcm_buffer object
+ * @param bits the number of in the source buffer
+ * @param src the source PCM buffer
+ * @param src_size the size of #src in bytes
+ * @param dest_size_r returns the number of bytes of the destination buffer
+ * @return the destination buffer
+ */
+const int32_t *
+pcm_convert_to_24(struct pcm_buffer *buffer,
+		  enum sample_format src_format, const void *src,
+		  size_t src_size, size_t *dest_size_r);
+
+/**
+ * Converts PCM samples to 32 bit.
+ *
+ * @param buffer a pcm_buffer object
+ * @param bits the number of in the source buffer
+ * @param src the source PCM buffer
+ * @param src_size the size of #src in bytes
+ * @param dest_size_r returns the number of bytes of the destination buffer
+ * @return the destination buffer
+ */
+const int32_t *
+pcm_convert_to_32(struct pcm_buffer *buffer,
+		  enum sample_format src_format, const void *src,
+		  size_t src_size, size_t *dest_size_r);
+
+/**
+ * Converts PCM samples to 32 bit floating point.
+ *
+ * @param buffer a pcm_buffer object
+ * @param bits the number of in the source buffer
+ * @param src the source PCM buffer
+ * @param src_size the size of #src in bytes
+ * @param dest_size_r returns the number of bytes of the destination buffer
+ * @return the destination buffer
+ */
+const float *
+pcm_convert_to_float(struct pcm_buffer *buffer,
+		     enum sample_format src_format, const void *src,
+		     size_t src_size, size_t *dest_size_r);
+
+#endif

src/pcm/PcmMix.cxx

@@ -0,0 +1,211 @@
+/*
+ * 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 "PcmMix.hxx"
+#include "PcmVolume.hxx"
+#include "PcmUtils.hxx"
+#include "audio_format.h"
+
+#include <math.h>
+
+template<typename T, typename U, unsigned bits>
+static T
+PcmAddVolume(T _a, T _b, int volume1, int volume2)
+{
+	U a(_a), b(_b);
+
+	U c = ((a * volume1 + b * volume2) +
+	       pcm_volume_dither() + PCM_VOLUME_1 / 2)
+		/ PCM_VOLUME_1;
+
+	return PcmClamp<T, U, bits>(c);
+}
+
+template<typename T, typename U, unsigned bits>
+static void
+PcmAddVolume(T *a, const T *b, unsigned n, int volume1, int volume2)
+{
+	for (size_t i = 0; i != n; ++i)
+		a[i] = PcmAddVolume<T, U, bits>(a[i], b[i], volume1, volume2);
+}
+
+template<typename T, typename U, unsigned bits>
+static void
+PcmAddVolumeVoid(void *a, const void *b, size_t size, int volume1, int volume2)
+{
+	constexpr size_t sample_size = sizeof(T);
+	assert(size % sample_size == 0);
+
+	PcmAddVolume<T, U, bits>((T *)a, (const T *)b, size / sample_size,
+				 volume1, volume2);
+}
+
+static void
+pcm_add_vol_float(float *buffer1, const float *buffer2,
+		  unsigned num_samples, float volume1, float volume2)
+{
+	while (num_samples > 0) {
+		float sample1 = *buffer1;
+		float sample2 = *buffer2++;
+
+		sample1 = (sample1 * volume1 + sample2 * volume2);
+		*buffer1++ = sample1;
+		--num_samples;
+	}
+}
+
+static bool
+pcm_add_vol(void *buffer1, const void *buffer2, size_t size,
+	    int vol1, int vol2,
+	    enum sample_format format)
+{
+	switch (format) {
+	case SAMPLE_FORMAT_UNDEFINED:
+	case SAMPLE_FORMAT_DSD:
+		/* not implemented */
+		return false;
+
+	case SAMPLE_FORMAT_S8:
+		PcmAddVolumeVoid<int8_t, int32_t, 8>(buffer1, buffer2, size,
+						     vol1, vol2);
+		return true;
+
+	case SAMPLE_FORMAT_S16:
+		PcmAddVolumeVoid<int16_t, int32_t, 16>(buffer1, buffer2, size,
+						       vol1, vol2);
+		return true;
+
+	case SAMPLE_FORMAT_S24_P32:
+		PcmAddVolumeVoid<int32_t, int64_t, 24>(buffer1, buffer2, size,
+						       vol1, vol2);
+		return true;
+
+	case SAMPLE_FORMAT_S32:
+		PcmAddVolumeVoid<int32_t, int64_t, 32>(buffer1, buffer2, size,
+						       vol1, vol2);
+		return true;
+
+	case SAMPLE_FORMAT_FLOAT:
+		pcm_add_vol_float((float *)buffer1, (const float *)buffer2,
+				  size / 4,
+				  pcm_volume_to_float(vol1),
+				  pcm_volume_to_float(vol2));
+		return true;
+	}
+
+	/* unreachable */
+	assert(false);
+	return false;
+}
+
+template<typename T, typename U, unsigned bits>
+static T
+PcmAdd(T _a, T _b)
+{
+	U a(_a), b(_b);
+	return PcmClamp<T, U, bits>(a + b);
+}
+
+template<typename T, typename U, unsigned bits>
+static void
+PcmAdd(T *a, const T *b, unsigned n)
+{
+	for (size_t i = 0; i != n; ++i)
+		a[i] = PcmAdd<T, U, bits>(a[i], b[i]);
+}
+
+template<typename T, typename U, unsigned bits>
+static void
+PcmAddVoid(void *a, const void *b, size_t size)
+{
+	constexpr size_t sample_size = sizeof(T);
+	assert(size % sample_size == 0);
+
+	PcmAdd<T, U, bits>((T *)a, (const T *)b, size / sample_size);
+}
+
+static void
+pcm_add_float(float *buffer1, const float *buffer2, unsigned num_samples)
+{
+	while (num_samples > 0) {
+		float sample1 = *buffer1;
+		float sample2 = *buffer2++;
+		*buffer1++ = sample1 + sample2;
+		--num_samples;
+	}
+}
+
+static bool
+pcm_add(void *buffer1, const void *buffer2, size_t size,
+	enum sample_format format)
+{
+	switch (format) {
+	case SAMPLE_FORMAT_UNDEFINED:
+	case SAMPLE_FORMAT_DSD:
+		/* not implemented */
+		return false;
+
+	case SAMPLE_FORMAT_S8:
+		PcmAddVoid<int8_t, int32_t, 8>(buffer1, buffer2, size);
+		return true;
+
+	case SAMPLE_FORMAT_S16:
+		PcmAddVoid<int16_t, int32_t, 16>(buffer1, buffer2, size);
+		return true;
+
+	case SAMPLE_FORMAT_S24_P32:
+		PcmAddVoid<int32_t, int64_t, 24>(buffer1, buffer2, size);
+		return true;
+
+	case SAMPLE_FORMAT_S32:
+		PcmAddVoid<int32_t, int64_t, 32>(buffer1, buffer2, size);
+		return true;
+
+	case SAMPLE_FORMAT_FLOAT:
+		pcm_add_float((float *)buffer1, (const float *)buffer2,
+			      size / 4);
+		return true;
+	}
+
+	/* unreachable */
+	assert(false);
+	return false;
+}
+
+bool
+pcm_mix(void *buffer1, const void *buffer2, size_t size,
+	enum sample_format format, float portion1)
+{
+	int vol1;
+	float s;
+
+	/* portion1 is between 0.0 and 1.0 for crossfading, MixRamp uses NaN
+	 * to signal mixing rather than fading */
+	if (isnan(portion1))
+		return pcm_add(buffer1, buffer2, size, format);
+
+	s = sin(M_PI_2 * portion1);
+	s *= s;
+
+	vol1 = s * PCM_VOLUME_1 + 0.5;
+	vol1 = vol1 > PCM_VOLUME_1 ? PCM_VOLUME_1 : (vol1 < 0 ? 0 : vol1);
+
+	return pcm_add_vol(buffer1, buffer2, size, vol1, PCM_VOLUME_1 - vol1, format);
+}

src/pcm/PcmMix.hxx

@@ -0,0 +1,49 @@
+/*
+ * Copyright (C) 2003-2013 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_MIX_HXX
+#define MPD_PCM_MIX_HXX
+
+#include "audio_format.h"
+#include "gcc.h"
+
+#include <stddef.h>
+
+/*
+ * Linearly mixes two PCM buffers.  Both must have the same length and
+ * the same audio format.  The formula is:
+ *
+ *   s1 := s1 * portion1 + s2 * (1 - portion1)
+ *
+ * @param buffer1 the first PCM buffer, and the destination buffer
+ * @param buffer2 the second PCM buffer
+ * @param size the size of both buffers in bytes
+ * @param format the sample format of both buffers
+ * @param portion1 a number between 0.0 and 1.0 specifying the portion
+ * of the first buffer in the mix; portion2 = (1.0 - portion1). The value
+ * NaN is used by the MixRamp code to specify that simple addition is required.
+ *
+ * @return true on success, false if the format is not supported
+ */
+gcc_warn_unused_result
+bool
+pcm_mix(void *buffer1, const void *buffer2, size_t size,
+	enum sample_format format, float portion1);
+
+#endif

src/pcm/PcmPrng.hxx

@@ -0,0 +1,33 @@
+/*
+ * Copyright (C) 2003-2013 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_PRNG_HXX
+#define MPD_PCM_PRNG_HXX
+
+/**
+ * A very simple linear congruential PRNG.  It's good enough for PCM
+ * dithering.
+ */
+static unsigned long
+pcm_prng(unsigned long state)
+{
+	return (state * 0x0019660dL + 0x3c6ef35fL) & 0xffffffffL;
+}
+
+#endif

src/pcm/PcmUtils.hxx

@@ -0,0 +1,66 @@
+/*
+ * 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.
+ */
+
+#ifndef MPD_PCM_UTILS_H
+#define MPD_PCM_UTILS_H
+
+#include "gcc.h"
+
+#include <limits>
+
+#include <stdint.h>
+
+/**
+ * Add a byte count to the specified pointer.  This is a utility
+ * function to convert a source pointer and a byte count to an "end"
+ * pointer for use in loops.
+ */
+template<typename T>
+static inline const T *
+pcm_end_pointer(const T *p, size_t size)
+{
+	return (const T *)((const uint8_t *)p + size);
+}
+
+/**
+ * Check if the value is within the range of the provided bit size,
+ * and caps it if necessary.
+ */
+template<typename T, typename U, unsigned bits>
+gcc_const
+static inline T
+PcmClamp(U x)
+{
+	constexpr U MIN_VALUE = -(U(1) << (bits - 1));
+	constexpr U MAX_VALUE = (U(1) << (bits - 1)) - 1;
+
+	typedef std::numeric_limits<T> limits;
+	static_assert(MIN_VALUE >= limits::min(), "out of range");
+	static_assert(MAX_VALUE <= limits::max(), "out of range");
+
+	if (gcc_unlikely(x < MIN_VALUE))
+		return T(MIN_VALUE);
+
+	if (gcc_unlikely(x > MAX_VALUE))
+		return T(MAX_VALUE);
+
+	return T(x);
+}
+
+#endif

src/pcm/PcmVolume.cxx

@@ -0,0 +1,194 @@
+/*
+ * 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 "PcmVolume.hxx"
+#include "PcmUtils.hxx"
+#include "audio_format.h"
+
+#include <glib.h>
+
+#include <stdint.h>
+#include <string.h>
+
+#undef G_LOG_DOMAIN
+#define G_LOG_DOMAIN "pcm_volume"
+
+static void
+pcm_volume_change_8(int8_t *buffer, const int8_t *end, int volume)
+{
+	while (buffer < end) {
+		int32_t sample = *buffer;
+
+		sample = (sample * volume + pcm_volume_dither() +
+			  PCM_VOLUME_1 / 2)
+			/ PCM_VOLUME_1;
+
+		*buffer++ = PcmClamp<int8_t, int16_t, 8>(sample);
+	}
+}
+
+static void
+pcm_volume_change_16(int16_t *buffer, const int16_t *end, int volume)
+{
+	while (buffer < end) {
+		int32_t sample = *buffer;
+
+		sample = (sample * volume + pcm_volume_dither() +
+			  PCM_VOLUME_1 / 2)
+			/ PCM_VOLUME_1;
+
+		*buffer++ = PcmClamp<int16_t, int32_t, 16>(sample);
+	}
+}
+
+#ifdef __i386__
+/**
+ * Optimized volume function for i386.  Use the EDX:EAX 2*32 bit
+ * multiplication result instead of emulating 64 bit multiplication.
+ */
+static inline int32_t
+pcm_volume_sample_24(int32_t sample, int32_t volume, G_GNUC_UNUSED int32_t dither)
+{
+	int32_t result;
+
+	asm(/* edx:eax = sample * volume */
+	    "imul %2\n"
+
+	    /* "add %3, %1\n"  dithering disabled for now, because we
+	       have no overflow check - is dithering really important
+	       here? */
+
+	    /* eax = edx:eax / PCM_VOLUME_1 */
+	    "sal $22, %%edx\n"
+	    "shr $10, %1\n"
+	    "or %%edx, %1\n"
+
+	    : "=a"(result)
+	    : "0"(sample), "r"(volume) /* , "r"(dither) */
+	    : "edx"
+	    );
+
+	return result;
+}
+#endif
+
+static void
+pcm_volume_change_24(int32_t *buffer, const int32_t *end, int volume)
+{
+	while (buffer < end) {
+#ifdef __i386__
+		/* assembly version for i386 */
+		int32_t sample = *buffer;
+
+		sample = pcm_volume_sample_24(sample, volume,
+					      pcm_volume_dither());
+#else
+		/* portable version */
+		int64_t sample = *buffer;
+
+		sample = (sample * volume + pcm_volume_dither() +
+			  PCM_VOLUME_1 / 2)
+			/ PCM_VOLUME_1;
+#endif
+		*buffer++ = PcmClamp<int32_t, int32_t, 24>(sample);
+	}
+}
+
+static void
+pcm_volume_change_32(int32_t *buffer, const int32_t *end, int volume)
+{
+	while (buffer < end) {
+#ifdef __i386__
+		/* assembly version for i386 */
+		int32_t sample = *buffer;
+
+		*buffer++ = pcm_volume_sample_24(sample, volume, 0);
+#else
+		/* portable version */
+		int64_t sample = *buffer;
+
+		sample = (sample * volume + pcm_volume_dither() +
+			  PCM_VOLUME_1 / 2)
+			/ PCM_VOLUME_1;
+		*buffer++ = PcmClamp<int32_t, int64_t, 32>(sample);
+#endif
+	}
+}
+
+static void
+pcm_volume_change_float(float *buffer, const float *end, float volume)
+{
+	while (buffer < end) {
+		float sample = *buffer;
+		sample *= volume;
+		*buffer++ = sample;
+	}
+}
+
+bool
+pcm_volume(void *buffer, size_t length,
+	   enum sample_format format,
+	   int volume)
+{
+	if (volume == PCM_VOLUME_1)
+		return true;
+
+	if (volume <= 0) {
+		memset(buffer, 0, length);
+		return true;
+	}
+
+	const void *end = pcm_end_pointer(buffer, length);
+	switch (format) {
+	case SAMPLE_FORMAT_UNDEFINED:
+	case SAMPLE_FORMAT_DSD:
+		/* not implemented */
+		return false;
+
+	case SAMPLE_FORMAT_S8:
+		pcm_volume_change_8((int8_t *)buffer, (const int8_t *)end,
+				    volume);
+		return true;
+
+	case SAMPLE_FORMAT_S16:
+		pcm_volume_change_16((int16_t *)buffer, (const int16_t *)end,
+				     volume);
+		return true;
+
+	case SAMPLE_FORMAT_S24_P32:
+		pcm_volume_change_24((int32_t *)buffer, (const int32_t *)end,
+				     volume);
+		return true;
+
+	case SAMPLE_FORMAT_S32:
+		pcm_volume_change_32((int32_t *)buffer, (const int32_t *)end,
+				     volume);
+		return true;
+
+	case SAMPLE_FORMAT_FLOAT:
+		pcm_volume_change_float((float *)buffer, (const float *)end,
+					pcm_volume_to_float(volume));
+		return true;
+	}
+
+	/* unreachable */
+	assert(false);
+	return false;
+}

src/pcm/PcmVolume.hxx

@@ -0,0 +1,82 @@
+/*
+ * Copyright (C) 2003-2013 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_VOLUME_HXX
+#define MPD_PCM_VOLUME_HXX
+
+#include "PcmPrng.hxx"
+#include "audio_format.h"
+
+#include <stdint.h>
+#include <stdbool.h>
+#include <stddef.h>
+
+enum {
+	/** this value means "100% volume" */
+	PCM_VOLUME_1 = 1024,
+};
+
+struct audio_format;
+
+/**
+ * Converts a float value (0.0 = silence, 1.0 = 100% volume) to an
+ * integer volume value (1000 = 100%).
+ */
+static inline int
+pcm_float_to_volume(float volume)
+{
+	return volume * PCM_VOLUME_1 + 0.5;
+}
+
+static inline float
+pcm_volume_to_float(int volume)
+{
+	return (float)volume / (float)PCM_VOLUME_1;
+}
+
+/**
+ * Returns the next volume dithering number, between -511 and +511.
+ * This number is taken from a global PRNG, see pcm_prng().
+ */
+static inline int
+pcm_volume_dither(void)
+{
+	static unsigned long state;
+	uint32_t r;
+
+	r = state = pcm_prng(state);
+
+	return (r & 511) - ((r >> 9) & 511);
+}
+
+/**
+ * Adjust the volume of the specified PCM buffer.
+ *
+ * @param buffer the PCM buffer
+ * @param length the length of the PCM buffer
+ * @param format the sample format of the PCM buffer
+ * @param volume the volume between 0 and #PCM_VOLUME_1
+ * @return true on success, false if the audio format is not supported
+ */
+bool
+pcm_volume(void *buffer, size_t length,
+	   enum sample_format format,
+	   int volume);
+
+#endif

src/pcm/dsd2pcm/dsd2pcm.c

@@ -0,0 +1,184 @@
+#include "util/bit_reverse.h"
+
+#include <stdlib.h>
+#include <string.h>
+
+#include "dsd2pcm.h"
+
+#define HTAPS    48             /* number of FIR constants */
+#define FIFOSIZE 16             /* must be a power of two */
+#define FIFOMASK (FIFOSIZE-1)   /* bit mask for FIFO offsets */
+#define CTABLES ((HTAPS+7)/8)   /* number of "8 MACs" lookup tables */
+
+#if FIFOSIZE*8 < HTAPS*2
+#error "FIFOSIZE too small"
+#endif
+
+/*
+ * Properties of this 96-tap lowpass filter when applied on a signal
+ * with sampling rate of 44100*64 Hz:
+ *
+ * () has a delay of 17 microseconds.
+ *
+ * () flat response up to 48 kHz
+ *
+ * () if you downsample afterwards by a factor of 8, the
+ *    spectrum below 70 kHz is practically alias-free.
+ *
+ * () stopband rejection is about 160 dB
+ *
+ * The coefficient tables ("ctables") take only 6 Kibi Bytes and
+ * should fit into a modern processor's fast cache.
+ */
+
+/*
+ * The 2nd half (48 coeffs) of a 96-tap symmetric lowpass filter
+ */
+static const double htaps[HTAPS] = {
+  0.09950731974056658,
+  0.09562845727714668,
+  0.08819647126516944,
+  0.07782552527068175,
+  0.06534876523171299,
+  0.05172629311427257,
+  0.0379429484910187,
+  0.02490921351762261,
+  0.0133774746265897,
+  0.003883043418804416,
+ -0.003284703416210726,
+ -0.008080250212687497,
+ -0.01067241812471033,
+ -0.01139427235000863,
+ -0.0106813877974587,
+ -0.009007905078766049,
+ -0.006828859761015335,
+ -0.004535184322001496,
+ -0.002425035959059578,
+ -0.0006922187080790708,
+  0.0005700762133516592,
+  0.001353838005269448,
+  0.001713709169690937,
+  0.001742046839472948,
+  0.001545601648013235,
+  0.001226696225277855,
+  0.0008704322683580222,
+  0.0005381636200535649,
+  0.000266446345425276,
+  7.002968738383528e-05,
+ -5.279407053811266e-05,
+ -0.0001140625650874684,
+ -0.0001304796361231895,
+ -0.0001189970287491285,
+ -9.396247155265073e-05,
+ -6.577634378272832e-05,
+ -4.07492895872535e-05,
+ -2.17407957554587e-05,
+ -9.163058931391722e-06,
+ -2.017460145032201e-06,
+  1.249721855219005e-06,
+  2.166655190537392e-06,
+  1.930520892991082e-06,
+  1.319400334374195e-06,
+  7.410039764949091e-07,
+  3.423230509967409e-07,
+  1.244182214744588e-07,
+  3.130441005359396e-08
+};
+
+static float ctables[CTABLES][256];
+static int precalculated = 0;
+
+static void precalc(void)
+{
+	int t, e, m, k;
+	double acc;
+	if (precalculated) return;
+	for (t=0; t<CTABLES; ++t) {
+		k = HTAPS - t*8;
+		if (k>8) k=8;
+		for (e=0; e<256; ++e) {
+			acc = 0.0;
+			for (m=0; m<k; ++m) {
+				acc += (((e >> (7-m)) & 1)*2-1) * htaps[t*8+m];
+			}
+			ctables[CTABLES-1-t][e] = (float)acc;
+		}
+	}
+	precalculated = 1;
+}
+
+struct dsd2pcm_ctx_s
+{
+	unsigned char fifo[FIFOSIZE];
+	unsigned fifopos;
+};
+
+extern dsd2pcm_ctx* dsd2pcm_init(void)
+{
+	dsd2pcm_ctx* ptr;
+	if (!precalculated) precalc();
+	ptr = (dsd2pcm_ctx*) malloc(sizeof(dsd2pcm_ctx));
+	if (ptr) dsd2pcm_reset(ptr);
+	return ptr;
+}
+
+extern void dsd2pcm_destroy(dsd2pcm_ctx* ptr)
+{
+	free(ptr);
+}
+
+extern dsd2pcm_ctx* dsd2pcm_clone(dsd2pcm_ctx* ptr)
+{
+	dsd2pcm_ctx* p2;
+	p2 = (dsd2pcm_ctx*) malloc(sizeof(dsd2pcm_ctx));
+	if (p2) {
+		memcpy(p2,ptr,sizeof(dsd2pcm_ctx));
+	}
+	return p2;
+}
+
+extern void dsd2pcm_reset(dsd2pcm_ctx* ptr)
+{
+	int i;
+	for (i=0; i<FIFOSIZE; ++i)
+		ptr->fifo[i] = 0x69; /* my favorite silence pattern */
+	ptr->fifopos = 0;
+	/* 0x69 = 01101001
+	 * This pattern "on repeat" makes a low energy 352.8 kHz tone
+	 * and a high energy 1.0584 MHz tone which should be filtered
+	 * out completely by any playback system --> silence
+	 */
+}
+
+extern void dsd2pcm_translate(
+	dsd2pcm_ctx* ptr,
+	size_t samples,
+	const unsigned char *src, ptrdiff_t src_stride,
+	int lsbf,
+	float *dst, ptrdiff_t dst_stride)
+{
+	unsigned ffp;
+	unsigned i;
+	unsigned bite1, bite2;
+	unsigned char* p;
+	double acc;
+	ffp = ptr->fifopos;
+	lsbf = lsbf ? 1 : 0;
+	while (samples-- > 0) {
+		bite1 = *src & 0xFFu;
+		if (lsbf) bite1 = bit_reverse(bite1);
+		ptr->fifo[ffp] = bite1; src += src_stride;
+		p = ptr->fifo + ((ffp-CTABLES) & FIFOMASK);
+		*p = bit_reverse(*p);
+		acc = 0;
+		for (i=0; i<CTABLES; ++i) {
+			bite1 = ptr->fifo[(ffp              -i) & FIFOMASK] & 0xFF;
+			bite2 = ptr->fifo[(ffp-(CTABLES*2-1)+i) & FIFOMASK] & 0xFF;
+			acc += ctables[i][bite1] + ctables[i][bite2];
+		}
+		*dst = (float)acc; dst += dst_stride;
+		ffp = (ffp + 1) & FIFOMASK;
+	}
+	ptr->fifopos = ffp;
+}
+

src/pcm/dsd2pcm/dsd2pcm.h

@@ -0,0 +1,64 @@
+#ifndef DSD2PCM_H_INCLUDED
+#define DSD2PCM_H_INCLUDED
+
+#include <stddef.h>
+#include <string.h>
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+struct dsd2pcm_ctx_s;
+
+typedef struct dsd2pcm_ctx_s dsd2pcm_ctx;
+
+/**
+ * initializes a "dsd2pcm engine" for one channel
+ * (precomputes tables and allocates memory)
+ *
+ * This is the only function that is not thread-safe in terms of the
+ * POSIX thread-safety definition because it modifies global state
+ * (lookup tables are computed during the first call)
+ */
+extern dsd2pcm_ctx* dsd2pcm_init(void);
+
+/**
+ * deinitializes a "dsd2pcm engine"
+ * (releases memory, don't forget!)
+ */
+extern void dsd2pcm_destroy(dsd2pcm_ctx *ctx);
+
+/**
+ * clones the context and returns a pointer to the
+ * newly allocated copy
+ */
+extern dsd2pcm_ctx* dsd2pcm_clone(dsd2pcm_ctx *ctx);
+
+/**
+ * resets the internal state for a fresh new stream
+ */
+extern void dsd2pcm_reset(dsd2pcm_ctx *ctx);
+
+/**
+ * "translates" a stream of octets to a stream of floats
+ * (8:1 decimation)
+ * @param ctx -- pointer to abstract context (buffers)
+ * @param samples -- number of octets/samples to "translate"
+ * @param src -- pointer to first octet (input)
+ * @param src_stride -- src pointer increment
+ * @param lsbitfirst -- bitorder, 0=msb first, 1=lsbfirst
+ * @param dst -- pointer to first float (output)
+ * @param dst_stride -- dst pointer increment
+ */
+extern void dsd2pcm_translate(dsd2pcm_ctx *ctx,
+	size_t samples,
+	const unsigned char *src, ptrdiff_t src_stride,
+	int lsbitfirst,
+	float *dst, ptrdiff_t dst_stride);
+
+#ifdef __cplusplus
+} /* extern "C" */
+#endif
+
+#endif /* include guard DSD2PCM_H_INCLUDED */
+

src/pcm/dsd2pcm/dsd2pcm.hpp

@@ -0,0 +1,39 @@
+#ifndef DSD2PCM_HXX_INCLUDED
+#define DSD2PCM_HXX_INCLUDED
+
+#include <algorithm>
+#include <stdexcept>
+#include "dsd2pcm.h"
+
+/**
+ * C++ PImpl Wrapper for the dsd2pcm C library
+ */
+
+class dxd
+{
+	dsd2pcm_ctx *handle;
+public:
+	dxd() : handle(dsd2pcm_init()) {}
+
+	dxd(dxd const& x) : handle(dsd2pcm_clone(x.handle)) {}
+
+	~dxd() { dsd2pcm_destroy(handle); }
+
+	friend void swap(dxd & a, dxd & b)
+	{ std::swap(a.handle,b.handle); }
+
+	dxd& operator=(dxd x)
+	{ swap(*this,x); return *this; }
+
+	void translate(size_t samples,
+		const unsigned char *src, ptrdiff_t src_stride,
+		bool lsbitfirst,
+		float *dst, ptrdiff_t dst_stride)
+	{
+		dsd2pcm_translate(handle,samples,src,src_stride,
+			lsbitfirst,dst,dst_stride);
+	}
+};
+
+#endif // DSD2PCM_HXX_INCLUDED
+

src/pcm/dsd2pcm/info.txt

@@ -0,0 +1,38 @@
+You downloaded the source code for "dsd2pcm" which is a simple little
+"filter" program, that takes a DSD data stream on stdin and converts
+it to a PCM stream (352.8 kHz, either 16 or 24 bits) and writes it to
+stdout. The code is split into three modules:
+
+  (1) dsd2pcm
+
+      This is where the 8:1 decimation magic happens. It's an
+      implementation of a symmetric 96-taps FIR lowpass filter
+      optimized for DSD inputs. If you feed this converter with
+      DSD64 you get a PCM stream at 352.8 kHz and floating point
+      samples. This module is independent and can be reused. 
+
+  (2) noiseshape
+
+      A module for applying generic noise shaping filters. It's
+      used for the 16-bit output mode in "main" to preserve the
+      dynamic range. This module is independent and can be reused.
+
+  (3) main.cpp (file contains the main function and handles I/O)
+
+The first two modules are pure C for maximum portability. In addition,
+there are C++ wrapper headers for convenient use of these modules in
+C++. The main application is a C++ application and makes use of the
+C++ headers to access the functionality of the first two modules.
+
+
+Under Linux this program is easily compiled by typing
+
+  g++ *.c *.cpp -O3 -o dsd2pcm
+
+provided you have GCC installed. That's why I didn't bother writing
+any makefiles. :-p
+
+
+Cheers!
+SG
+

src/pcm/dsd2pcm/main.cpp

@@ -0,0 +1,120 @@
+#include <iostream>
+#include <vector>
+#include <cstring>
+
+#include "dsd2pcm.hpp"
+#include "noiseshape.hpp"
+
+namespace {
+
+const float my_ns_coeffs[] = {
+//     b1           b2           a1           a2
+  -1.62666423,  0.79410094,  0.61367127,  0.23311013,  // section 1
+  -1.44870017,  0.54196219,  0.03373857,  0.70316556   // section 2
+};
+
+const int my_ns_soscount = sizeof(my_ns_coeffs)/(sizeof(my_ns_coeffs[0])*4);
+
+inline long myround(float x)
+{
+	return static_cast<long>(x + (x>=0 ? 0.5f : -0.5f));
+}
+
+template<typename T>
+struct id { typedef T type; };
+
+template<typename T>
+inline T clip(
+	typename id<T>::type min,
+	T v,
+	typename id<T>::type max)
+{
+	if (v<min) return min;
+	if (v>max) return max;
+	return v;
+}
+
+inline void write_intel16(unsigned char * ptr, unsigned word)
+{
+	ptr[0] =  word       & 0xFF;
+	ptr[1] = (word >> 8) & 0xFF;
+}
+
+inline void write_intel24(unsigned char * ptr, unsigned long word)
+{
+	ptr[0] =  word        & 0xFF;
+	ptr[1] = (word >>  8) & 0xFF;
+	ptr[2] = (word >> 16) & 0xFF;
+}
+
+} // anonymous namespace
+
+using std::vector;
+using std::cin;
+using std::cout;
+using std::cerr;
+
+int main(int argc, char *argv[])
+{
+	const int block = 16384;
+	int channels   = -1;
+	int lsbitfirst = -1;
+	int bits       = -1;
+	if (argc==4) {
+		if ('1'<=argv[1][0] && argv[1][0]<='9') channels = 1 + (argv[1][0]-'1');
+		if (argv[2][0]=='m' || argv[2][0]=='M') lsbitfirst=0;
+		if (argv[2][0]=='l' || argv[2][0]=='L') lsbitfirst=1;
+		if (!strcmp(argv[3],"16")) bits = 16;
+		if (!strcmp(argv[3],"24")) bits = 24;
+	}
+	if (channels<1 || lsbitfirst<0 || bits<0) {
+		cerr << "\n"
+			"DSD2PCM filter (raw DSD64 --> 352 kHz raw PCM)\n"
+			"(c) 2009 Sebastian Gesemann\n\n"
+			"(filter as in \"reads data from stdin and writes to stdout\")\n\n"
+			"Syntax: dsd2pcm <channels> <bitorder> <bitdepth>\n"
+			"channels = 1,2,3,...,9 (number of channels in DSD stream)\n"
+			"bitorder = L (lsb first), M (msb first) (DSD stream option)\n"
+			"bitdepth = 16 or 24 (intel byte order, output option)\n\n"
+			"Note: At 16 bits/sample a noise shaper kicks in that can preserve\n"
+			"a dynamic range of 135 dB below 30 kHz.\n\n";
+		return 1;
+	}
+	int bytespersample = bits/8;
+	vector<dxd> dxds (channels);
+	vector<noise_shaper> ns;
+	if (bits==16) {
+		ns.resize(channels, noise_shaper(my_ns_soscount, my_ns_coeffs) );
+	}
+	vector<unsigned char> dsd_data (block * channels);
+	vector<float> float_data (block);
+	vector<unsigned char> pcm_data (block * channels * bytespersample);
+	char * const dsd_in  = reinterpret_cast<char*>(&dsd_data[0]);
+	char * const pcm_out = reinterpret_cast<char*>(&pcm_data[0]);
+	while (cin.read(dsd_in,block * channels)) {
+		for (int c=0; c<channels; ++c) {
+			dxds[c].translate(block,&dsd_data[0]+c,channels,
+				lsbitfirst,
+				&float_data[0],1);
+			unsigned char * out = &pcm_data[0] + c*bytespersample;
+			if (bits==16) {
+				for (int s=0; s<block; ++s) {
+					float r = float_data[s]*32768 + ns[c].get();
+					long smp = clip(-32768,myround(r),32767);
+					ns[c].update( clip(-1,smp-r,1) );
+					write_intel16(out,smp);
+					out += channels*bytespersample;
+				}
+			} else {
+				for (int s=0; s<block; ++s) {
+					float r = float_data[s]*8388608;
+					long smp = clip(-8388608,myround(r),8388607);
+					write_intel24(out,smp);
+					out += channels*bytespersample;
+				}
+			}
+		}
+		cout.write(pcm_out,block*channels*bytespersample);
+	}
+}
+

src/pcm/dsd2pcm/noiseshape.c

@@ -0,0 +1,83 @@
+#include <stdlib.h>
+#include <string.h>
+
+#include "noiseshape.h"
+
+extern int noise_shape_init(
+	noise_shape_ctx *ctx,
+	int sos_count,
+	const float *coeffs)
+{
+	int i;
+	ctx->sos_count = sos_count;
+	ctx->bbaa = coeffs;
+	ctx->t1 = (float*) malloc(sizeof(float)*sos_count);
+	if (!ctx->t1) goto escape1;
+	ctx->t2 = (float*) malloc(sizeof(float)*sos_count);
+	if (!ctx->t2) goto escape2;
+	for (i=0; i<sos_count; ++i) {
+		ctx->t1[i] = 0.f;
+		ctx->t2[i] = 0.f;
+	}
+	return 0;
+escape2:
+	free(ctx->t1);
+escape1:
+	return -1;
+}
+
+extern void noise_shape_destroy(
+	noise_shape_ctx *ctx)
+{
+	free(ctx->t1);
+	free(ctx->t2);
+}
+
+extern int noise_shape_clone(
+	const noise_shape_ctx *from,
+	noise_shape_ctx *to)
+{
+	to->sos_count = from->sos_count;
+	to->bbaa = from->bbaa;
+	to->t1 = (float*) malloc(sizeof(float)*to->sos_count);
+	if (!to->t1) goto error1;
+	to->t2 = (float*) malloc(sizeof(float)*to->sos_count);
+	if (!to->t2) goto error2;
+	memcpy(to->t1,from->t1,sizeof(float)*to->sos_count);
+	memcpy(to->t2,from->t2,sizeof(float)*to->sos_count);
+	return 0;
+error2:
+	free(to->t1);
+error1:
+	return -1;
+}
+
+extern float noise_shape_get(noise_shape_ctx *ctx)
+{
+	int i;
+	float acc;
+	const float *c;
+	acc = 0.0;
+	c = ctx->bbaa;
+	for (i=0; i<ctx->sos_count; ++i) {
+		float t1i = ctx->t1[i];
+		float t2i = ctx->t2[i];
+		ctx->t2[i] = acc -= t1i * c[2] + t2i * c[3];
+		acc += t1i * c[0] + t2i * c[1];
+		c += 4;
+	}
+	return acc;
+}
+
+extern void noise_shape_update(noise_shape_ctx *ctx, float qerror)
+{
+	float *p;
+	int i;
+	for (i=0; i<ctx->sos_count; ++i) {
+		ctx->t2[i] += qerror;
+	}
+	p = ctx->t1;
+	ctx->t1 = ctx->t2;
+	ctx->t2 = p;
+}
+

src/pcm/dsd2pcm/noiseshape.h

@@ -0,0 +1,57 @@
+#ifndef NOISE_SHAPE_H_INCLUDED
+#define NOISE_SHAPE_H_INCLUDED
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+typedef struct noise_shape_ctx_s {
+	int sos_count;      /* number of second order sections */
+	const float *bbaa;  /* filter coefficients, owned by user */
+	float *t1, *t2;     /* filter state, owned by ns library */
+} noise_shape_ctx;
+
+/**
+ * initializes a noise_shaper context
+ * returns an error code or 0
+ */
+extern int noise_shape_init(
+	noise_shape_ctx *ctx,
+	int sos_count,
+	const float *coeffs);
+
+/**
+ * destroys a noise_shaper context
+ */
+extern void noise_shape_destroy(
+	noise_shape_ctx *ctx);
+
+/**
+ * initializes a noise_shaper context so that its state
+ * is a copy of a given context
+ * returns an error code or 0
+ */
+extern int noise_shape_clone(
+	const noise_shape_ctx *from, noise_shape_ctx *to);
+
+/**
+ * computes the next "noise shaping sample". Note: This call
+ * alters the internal state. xxx_get and xxx_update must be
+ * called in an alternating manner.
+ */
+extern float noise_shape_get(
+	noise_shape_ctx *ctx);
+
+/**
+ * updates the noise shaper's state with the
+ * last quantization error
+ */
+extern void noise_shape_update(
+	noise_shape_ctx *ctx, float qerror);
+
+#ifdef __cplusplus
+} /* extern "C" */
+#endif
+
+#endif /* NOISE_SHAPE_H_INCLUDED */
+

src/pcm/dsd2pcm/noiseshape.hpp

@@ -0,0 +1,43 @@
+#ifndef NOISE_SHAPE_HXX_INCLUDED
+#define NOISE_SHAPE_HXX_INCLUDED
+
+#include <stdexcept>
+#include "noiseshape.h"
+
+/**
+ * C++ wrapper for the noiseshape C library
+ */
+
+class noise_shaper
+{
+	noise_shape_ctx ctx;
+public:
+	noise_shaper(int sos_count, const float *bbaa)
+	{
+		noise_shape_init(&ctx, sos_count, bbaa);
+	}
+
+	noise_shaper(noise_shaper const& x)
+	{
+		noise_shape_clone(&x.ctx,&ctx);
+	}
+
+	~noise_shaper()
+	{ noise_shape_destroy(&ctx); }
+
+	noise_shaper& operator=(noise_shaper const& x)
+	{
+		if (this != &x) {
+			noise_shape_destroy(&ctx);
+			noise_shape_clone(&x.ctx,&ctx);
+		}
+		return *this;
+	}
+
+	float get() { return noise_shape_get(&ctx); }
+
+	void update(float error) { noise_shape_update(&ctx,error); }
+};
+
+#endif /* NOISE_SHAPE_HXX_INCLUDED */
+

src/pcm/pcm_buffer.c

@@ -0,0 +1,58 @@
+/*
+ * 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 "pcm_buffer.h"
+#include "poison.h"
+
+/**
+ * Align the specified size to the next 8k boundary.
+ */
+G_GNUC_CONST
+static size_t
+align_8k(size_t size)
+{
+	return ((size - 1) | 0x1fff) + 1;
+}
+
+void *
+pcm_buffer_get(struct pcm_buffer *buffer, size_t size)
+{
+	assert(buffer != NULL);
+
+	if (size == 0)
+		/* never return NULL, because NULL would be assumed to
+		   be an error condition */
+		size = 1;
+
+	if (buffer->size < size) {
+		/* free the old buffer */
+		g_free(buffer->buffer);
+
+		buffer->size = align_8k(size);
+		buffer->buffer = g_malloc(buffer->size);
+	} else {
+		/* discard old buffer contents */
+		poison_undefined(buffer->buffer, buffer->size);
+	}
+
+	assert(buffer->size >= size);
+
+	return buffer->buffer;
+}

src/pcm/pcm_buffer.h

@@ -0,0 +1,85 @@
+/*
+ * 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.
+ */
+
+#ifndef PCM_BUFFER_H
+#define PCM_BUFFER_H
+
+#include "check.h"
+
+#include <glib.h>
+
+#include <assert.h>
+
+/**
+ * Manager for a temporary buffer which grows as needed.  We could
+ * allocate a new buffer every time pcm_convert() is called, but that
+ * would put too much stress on the allocator.
+ */
+struct pcm_buffer {
+	void *buffer;
+
+	size_t size;
+};
+
+/**
+ * Initialize the buffer, but don't allocate anything yet.
+ */
+static inline void
+pcm_buffer_init(struct pcm_buffer *buffer)
+{
+	assert(buffer != NULL);
+
+	buffer->buffer = NULL;
+	buffer->size = 0;
+}
+
+/**
+ * Free resources.  This function may be called more than once.
+ */
+static inline void
+pcm_buffer_deinit(struct pcm_buffer *buffer)
+{
+	assert(buffer != NULL);
+
+	g_free(buffer->buffer);
+
+	buffer->buffer = NULL;
+}
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/**
+ * Get the buffer, and guarantee a minimum size.  This buffer becomes
+ * invalid with the next pcm_buffer_get() call.
+ *
+ * This function will never return NULL, even if size is zero, because
+ * the PCM library uses the NULL return value to signal "error".  An
+ * empty destination buffer is not always an error.
+ */
+G_GNUC_MALLOC
+void *
+pcm_buffer_get(struct pcm_buffer *buffer, size_t size);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif

src/pcm/pcm_dsd.c

@@ -0,0 +1,85 @@
+/*
+ * 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 "pcm_dsd.h"
+#include "dsd2pcm/dsd2pcm.h"
+
+#include <glib.h>
+#include <string.h>
+
+void
+pcm_dsd_init(struct pcm_dsd *dsd)
+{
+	pcm_buffer_init(&dsd->buffer);
+
+	memset(dsd->dsd2pcm, 0, sizeof(dsd->dsd2pcm));
+}
+
+void
+pcm_dsd_deinit(struct pcm_dsd *dsd)
+{
+	pcm_buffer_deinit(&dsd->buffer);
+
+	for (unsigned i = 0; i < G_N_ELEMENTS(dsd->dsd2pcm); ++i)
+		if (dsd->dsd2pcm[i] != NULL)
+			dsd2pcm_destroy(dsd->dsd2pcm[i]);
+}
+
+void
+pcm_dsd_reset(struct pcm_dsd *dsd)
+{
+	for (unsigned i = 0; i < G_N_ELEMENTS(dsd->dsd2pcm); ++i)
+		if (dsd->dsd2pcm[i] != NULL)
+			dsd2pcm_reset(dsd->dsd2pcm[i]);
+}
+
+const float *
+pcm_dsd_to_float(struct pcm_dsd *dsd, unsigned channels, bool lsbfirst,
+		 const uint8_t *src, size_t src_size,
+		 size_t *dest_size_r)
+{
+	assert(dsd != NULL);
+	assert(src != NULL);
+	assert(src_size > 0);
+	assert(src_size % channels == 0);
+	assert(channels <= G_N_ELEMENTS(dsd->dsd2pcm));
+
+	const unsigned num_samples = src_size;
+	const unsigned num_frames = src_size / channels;
+
+	float *dest;
+	const size_t dest_size = num_samples * sizeof(*dest);
+	*dest_size_r = dest_size;
+	dest = pcm_buffer_get(&dsd->buffer, dest_size);
+
+	for (unsigned c = 0; c < channels; ++c) {
+		if (dsd->dsd2pcm[c] == NULL) {
+			dsd->dsd2pcm[c] = dsd2pcm_init();
+			if (dsd->dsd2pcm[c] == NULL)
+				return NULL;
+		}
+
+		dsd2pcm_translate(dsd->dsd2pcm[c], num_frames,
+				  src + c, channels,
+				  lsbfirst, dest + c, channels);
+	}
+
+	return dest;
+}

src/pcm/pcm_dsd.h

@@ -0,0 +1,52 @@
+/*
+ * 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.
+ */
+
+#ifndef MPD_PCM_DSD_H
+#define MPD_PCM_DSD_H
+
+#include "check.h"
+#include "pcm_buffer.h"
+
+#include <stdbool.h>
+#include <stdint.h>
+
+/**
+ * Wrapper for the dsd2pcm library.
+ */
+struct pcm_dsd {
+	struct pcm_buffer buffer;
+
+	struct dsd2pcm_ctx_s *dsd2pcm[32];
+};
+
+void
+pcm_dsd_init(struct pcm_dsd *dsd);
+
+void
+pcm_dsd_deinit(struct pcm_dsd *dsd);
+
+void
+pcm_dsd_reset(struct pcm_dsd *dsd);
+
+const float *
+pcm_dsd_to_float(struct pcm_dsd *dsd, unsigned channels, bool lsbfirst,
+		 const uint8_t *src, size_t src_size,
+		 size_t *dest_size_r);
+
+#endif

src/pcm/pcm_dsd_usb.c

@@ -0,0 +1,97 @@
+/*
+ * 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 "pcm_dsd_usb.h"
+#include "pcm_buffer.h"
+#include "audio_format.h"
+
+G_GNUC_CONST
+static inline uint32_t
+pcm_two_dsd_to_usb_marker1(uint8_t a, uint8_t b)
+{
+	return 0xff050000 | (a << 8) | b;
+}
+
+G_GNUC_CONST
+static inline uint32_t
+pcm_two_dsd_to_usb_marker2(uint8_t a, uint8_t b)
+{
+	return 0xfffa0000 | (a << 8) | b;
+}
+
+
+const uint32_t *
+pcm_dsd_to_usb(struct pcm_buffer *buffer, unsigned channels,
+	       const uint8_t *src, size_t src_size,
+	       size_t *dest_size_r)
+{
+	assert(buffer != NULL);
+	assert(audio_valid_channel_count(channels));
+	assert(src != NULL);
+	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;
+
+	const size_t dest_size = num_samples * 4;
+	*dest_size_r = dest_size;
+	uint32_t *const dest0 = pcm_buffer_get(buffer, dest_size),
+		*dest = dest0;
+
+	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_usb_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_usb_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;
+}

src/pcm/pcm_dsd_usb.h

@@ -0,0 +1,42 @@
+/*
+ * 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.
+ */
+
+#ifndef MPD_PCM_DSD_USB_H
+#define MPD_PCM_DSD_USB_H
+
+#include "check.h"
+
+#include <stdbool.h>
+#include <stdint.h>
+#include <stddef.h>
+
+struct pcm_buffer;
+
+/**
+ * Pack DSD 1 bit samples into (padded) 24 bit PCM samples for
+ * playback over USB, according to the proposed standard by 
+ * dCS and others:
+ * http://www.sonore.us/DoP_openStandard_1v1.pdf
+ */
+const uint32_t *
+pcm_dsd_to_usb(struct pcm_buffer *buffer, unsigned channels,
+	       const uint8_t *src, size_t src_size,
+	       size_t *dest_size_r);
+
+#endif

src/pcm/pcm_export.c

@@ -0,0 +1,160 @@
+/*
+ * 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 "pcm_export.h"
+#include "pcm_dsd_usb.h"
+#include "pcm_pack.h"
+#include "util/byte_reverse.h"
+
+void
+pcm_export_init(struct pcm_export_state *state)
+{
+	pcm_buffer_init(&state->reverse_buffer);
+	pcm_buffer_init(&state->pack_buffer);
+	pcm_buffer_init(&state->dsd_buffer);
+}
+
+void pcm_export_deinit(struct pcm_export_state *state)
+{
+	pcm_buffer_deinit(&state->reverse_buffer);
+	pcm_buffer_deinit(&state->pack_buffer);
+	pcm_buffer_deinit(&state->dsd_buffer);
+}
+
+void
+pcm_export_open(struct pcm_export_state *state,
+		enum sample_format sample_format, unsigned channels,
+		bool dsd_usb, bool shift8, bool pack, bool reverse_endian)
+{
+	assert(audio_valid_sample_format(sample_format));
+	assert(!dsd_usb || audio_valid_channel_count(channels));
+
+	state->channels = channels;
+	state->dsd_usb = dsd_usb && sample_format == SAMPLE_FORMAT_DSD;
+	if (state->dsd_usb)
+		/* after the conversion to DSD-over-USB, the DSD
+		   samples are stuffed inside fake 24 bit samples */
+		sample_format = SAMPLE_FORMAT_S24_P32;
+
+	state->shift8 = shift8 && sample_format == SAMPLE_FORMAT_S24_P32;
+	state->pack24 = pack && sample_format == SAMPLE_FORMAT_S24_P32;
+
+	assert(!state->shift8 || !state->pack24);
+
+	state->reverse_endian = 0;
+	if (reverse_endian) {
+		size_t sample_size = state->pack24
+			? 3
+			: sample_format_size(sample_format);
+		assert(sample_size <= 0xff);
+
+		if (sample_size > 1)
+			state->reverse_endian = sample_size;
+	}
+}
+
+size_t
+pcm_export_frame_size(const struct pcm_export_state *state,
+		      const struct audio_format *audio_format)
+{
+	assert(state != NULL);
+	assert(audio_format != NULL);
+
+	if (state->pack24)
+		/* packed 24 bit samples (3 bytes per sample) */
+		return audio_format->channels * 3;
+
+	if (state->dsd_usb)
+		/* the DSD-over-USB draft says that DSD 1-bit samples
+		   are enclosed within 24 bit samples, and MPD's
+		   representation of 24 bit is padded to 32 bit (4
+		   bytes per sample) */
+		return audio_format->channels * 4;
+
+	return audio_format_frame_size(audio_format);
+}
+
+const void *
+pcm_export(struct pcm_export_state *state, const void *data, size_t size,
+	   size_t *dest_size_r)
+{
+	if (state->dsd_usb)
+		data = pcm_dsd_to_usb(&state->dsd_buffer, state->channels,
+				      data, size, &size);
+
+	if (state->pack24) {
+		assert(size % 4 == 0);
+
+		const size_t num_samples = size / 4;
+		const size_t dest_size = num_samples * 3;
+
+		const uint8_t *src8 = data, *src_end8 = src8 + size;
+		uint8_t *dest = pcm_buffer_get(&state->pack_buffer, dest_size);
+		assert(dest != NULL);
+
+		pcm_pack_24(dest, (const int32_t *)src8,
+			    (const int32_t *)src_end8);
+
+		data = dest;
+		size = dest_size;
+	} else if (state->shift8) {
+		assert(size % 4 == 0);
+
+		const uint8_t *src8 = data, *src_end8 = src8 + size;
+		const uint32_t *src = (const uint32_t *)src8;
+		const uint32_t *const src_end = (const uint32_t *)src_end8;
+
+		uint32_t *dest = pcm_buffer_get(&state->pack_buffer, size);
+		data = dest;
+
+		while (src < src_end)
+			*dest++ = *src++ << 8;
+	}
+
+
+	if (state->reverse_endian > 0) {
+		assert(state->reverse_endian >= 2);
+
+		void *dest = pcm_buffer_get(&state->reverse_buffer, size);
+		assert(dest != NULL);
+
+		const uint8_t *src = data, *src_end = src + size;
+		reverse_bytes(dest, src, src_end, state->reverse_endian);
+
+		data = dest;
+	}
+
+	*dest_size_r = size;
+	return data;
+}
+
+size_t
+pcm_export_source_size(const struct pcm_export_state *state, size_t size)
+{
+	if (state->pack24)
+		/* 32 bit to 24 bit conversion (4 to 3 bytes) */
+		size = (size / 3) * 4;
+
+	if (state->dsd_usb)
+		/* DSD over USB doubles the transport size */
+		size /= 2;
+
+	return size;
+}

src/pcm/pcm_export.h

@@ -0,0 +1,155 @@
+/*
+ * 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.
+ */
+
+#ifndef PCM_EXPORT_H
+#define PCM_EXPORT_H
+
+#include "check.h"
+#include "pcm_buffer.h"
+#include "audio_format.h"
+
+#include <stdbool.h>
+
+struct audio_format;
+
+/**
+ * An object that handles export of PCM samples to some instance
+ * outside of MPD.  It has a few more options to tweak the binary
+ * representation which are not supported by the pcm_convert library.
+ */
+struct pcm_export_state {
+	/**
+	 * The buffer is used to convert DSD samples to the
+	 * DSD-over-USB format.
+	 *
+	 * @see #dsd_usb
+	 */
+	struct pcm_buffer dsd_buffer;
+
+	/**
+	 * The buffer is used to pack samples, removing padding.
+	 *
+	 * @see #pack24
+	 */
+	struct pcm_buffer pack_buffer;
+
+	/**
+	 * The buffer is used to reverse the byte order.
+	 *
+	 * @see #reverse_endian
+	 */
+	struct pcm_buffer reverse_buffer;
+
+	/**
+	 * The number of channels.
+	 */
+	uint8_t channels;
+
+	/**
+	 * Convert DSD to DSD-over-USB?  Input format must be
+	 * SAMPLE_FORMAT_DSD and output format must be
+	 * SAMPLE_FORMAT_S24_P32.
+	 */
+	bool dsd_usb;
+
+	/**
+	 * Convert (padded) 24 bit samples to 32 bit by shifting 8
+	 * bits to the left?
+	 */
+	bool shift8;
+
+	/**
+	 * Pack 24 bit samples?
+	 */
+	bool pack24;
+
+	/**
+	 * Export the samples in reverse byte order?  A non-zero value
+	 * means the option is enabled and represents the size of each
+	 * sample (2 or bigger).
+	 */
+	uint8_t reverse_endian;
+};
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/**
+ * Initialize a #pcm_export_state object.
+ */
+void
+pcm_export_init(struct pcm_export_state *state);
+
+/**
+ * Deinitialize a #pcm_export_state object and free allocated memory.
+ */
+void
+pcm_export_deinit(struct pcm_export_state *state);
+
+/**
+ * Open the #pcm_export_state object.
+ *
+ * There is no "close" method.  This function may be called multiple
+ * times to reuse the object, until pcm_export_deinit() is called.
+ *
+ * This function cannot fail.
+ *
+ * @param channels the number of channels; ignored unless dsd_usb is set
+ */
+void
+pcm_export_open(struct pcm_export_state *state,
+		enum sample_format sample_format, unsigned channels,
+		bool dsd_usb, bool shift8, bool pack, bool reverse_endian);
+
+/**
+ * Calculate the size of one output frame.
+ */
+G_GNUC_PURE
+size_t
+pcm_export_frame_size(const struct pcm_export_state *state,
+		      const struct audio_format *audio_format);
+
+/**
+ * Export a PCM buffer.
+ *
+ * @param state an initialized and open pcm_export_state object
+ * @param src the source PCM buffer
+ * @param src_size the size of #src in bytes
+ * @param dest_size_r returns the number of bytes of the destination buffer
+ * @return the destination buffer (may be a pointer to the source buffer)
+ */
+const void *
+pcm_export(struct pcm_export_state *state, const void *src, size_t src_size,
+	   size_t *dest_size_r);
+
+/**
+ * Converts the number of consumed bytes from the pcm_export()
+ * destination buffer to the according number of bytes from the
+ * pcm_export() source buffer.
+ */
+G_GNUC_PURE
+size_t
+pcm_export_source_size(const struct pcm_export_state *state, size_t dest_size);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif

src/pcm/pcm_pack.c

@@ -0,0 +1,77 @@
+/*
+ * 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 "pcm_pack.h"
+
+#include <glib.h>
+
+static void
+pack_sample(uint8_t *dest, const int32_t *src0)
+{
+	const uint8_t *src = (const uint8_t *)src0;
+
+	if (G_BYTE_ORDER == G_BIG_ENDIAN)
+		++src;
+
+	*dest++ = *src++;
+	*dest++ = *src++;
+	*dest++ = *src++;
+}
+
+void
+pcm_pack_24(uint8_t *dest, const int32_t *src, const int32_t *src_end)
+{
+	/* duplicate loop to help the compiler's optimizer (constant
+	   parameter to the pack_sample() inline function) */
+
+	while (src < src_end) {
+		pack_sample(dest, src++);
+		dest += 3;
+	}
+}
+
+static void
+unpack_sample(int32_t *dest0, const uint8_t *src)
+{
+	uint8_t *dest = (uint8_t *)dest0;
+
+	if (G_BYTE_ORDER == G_BIG_ENDIAN)
+		/* extend the sign bit to the most fourth byte */
+		*dest++ = *src & 0x80 ? 0xff : 0x00;
+
+	*dest++ = *src++;
+	*dest++ = *src++;
+	*dest++ = *src;
+
+	if (G_BYTE_ORDER == G_LITTLE_ENDIAN)
+		/* extend the sign bit to the most fourth byte */
+		*dest++ = *src & 0x80 ? 0xff : 0x00;
+}
+
+void
+pcm_unpack_24(int32_t *dest, const uint8_t *src, const uint8_t *src_end)
+{
+	/* duplicate loop to help the compiler's optimizer (constant
+	   parameter to the unpack_sample() inline function) */
+
+	while (src < src_end) {
+		unpack_sample(dest++, src);
+		src += 3;
+	}
+}

src/pcm/pcm_pack.h

@@ -0,0 +1,55 @@
+/*
+ * 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.
+ */
+
+/** \file
+ *
+ * Library for working with packed 24 bit samples.
+ */
+
+#ifndef PCM_PACK_H
+#define PCM_PACK_H
+
+#include <stdbool.h>
+#include <stdint.h>
+
+/**
+ * Converts padded 24 bit samples (4 bytes per sample) to packed 24
+ * bit samples (3 bytes per sample).
+ *
+ * This function can be used to convert a buffer in-place.
+ *
+ * @param dest the destination buffer (array of triples)
+ * @param src the source buffer
+ * @param num_samples the number of samples to convert
+ */
+void
+pcm_pack_24(uint8_t *dest, const int32_t *src, const int32_t *src_end);
+
+/**
+ * Converts packed 24 bit samples (3 bytes per sample) to padded 24
+ * bit samples (4 bytes per sample).
+ *
+ * @param dest the destination buffer
+ * @param src the source buffer (array of triples)
+ * @param num_samples the number of samples to convert
+ */
+void
+pcm_unpack_24(int32_t *dest, const uint8_t *src, const uint8_t *src_end);
+
+#endif

src/pcm/pcm_resample.c

@@ -0,0 +1,159 @@
+/*
+ * 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 "pcm_resample_internal.h"
+
+#ifdef HAVE_LIBSAMPLERATE
+#include "conf.h"
+#endif
+
+#include <string.h>
+
+#ifdef HAVE_LIBSAMPLERATE
+static bool lsr_enabled;
+#endif
+
+#ifdef HAVE_LIBSAMPLERATE
+static bool
+pcm_resample_lsr_enabled(void)
+{
+	return lsr_enabled;
+}
+#endif
+
+bool
+pcm_resample_global_init(GError **error_r)
+{
+#ifdef HAVE_LIBSAMPLERATE
+	const char *converter =
+		config_get_string(CONF_SAMPLERATE_CONVERTER, "");
+
+	lsr_enabled = strcmp(converter, "internal") != 0;
+	if (lsr_enabled)
+		return pcm_resample_lsr_global_init(converter, error_r);
+	else
+		return true;
+#else
+	(void)error_r;
+	return true;
+#endif
+}
+
+void pcm_resample_init(struct pcm_resample_state *state)
+{
+#ifdef HAVE_LIBSAMPLERATE
+	if (pcm_resample_lsr_enabled())
+		pcm_resample_lsr_init(state);
+	else
+#endif
+		pcm_resample_fallback_init(state);
+}
+
+void pcm_resample_deinit(struct pcm_resample_state *state)
+{
+#ifdef HAVE_LIBSAMPLERATE
+	if (pcm_resample_lsr_enabled())
+		pcm_resample_lsr_deinit(state);
+	else
+#endif
+		pcm_resample_fallback_deinit(state);
+}
+
+void
+pcm_resample_reset(struct pcm_resample_state *state)
+{
+#ifdef HAVE_LIBSAMPLERATE
+	pcm_resample_lsr_reset(state);
+#else
+	(void)state;
+#endif
+}
+
+const float *
+pcm_resample_float(struct pcm_resample_state *state,
+		   unsigned channels,
+		   unsigned src_rate,
+		   const float *src_buffer, size_t src_size,
+		   unsigned dest_rate, size_t *dest_size_r,
+		   GError **error_r)
+{
+#ifdef HAVE_LIBSAMPLERATE
+	if (pcm_resample_lsr_enabled())
+		return pcm_resample_lsr_float(state, channels,
+					      src_rate, src_buffer, src_size,
+					      dest_rate, dest_size_r,
+					      error_r);
+#else
+	(void)error_r;
+#endif
+
+	/* sizeof(float)==sizeof(int32_t); the fallback resampler does
+	   not do any math on the sample values, so this hack is
+	   possible: */
+	return (const float *)
+		pcm_resample_fallback_32(state, channels,
+					 src_rate, (const int32_t *)src_buffer,
+					 src_size,
+					 dest_rate, dest_size_r);
+}
+
+const int16_t *
+pcm_resample_16(struct pcm_resample_state *state,
+		unsigned channels,
+		unsigned src_rate, const int16_t *src_buffer, size_t src_size,
+		unsigned dest_rate, size_t *dest_size_r,
+		GError **error_r)
+{
+#ifdef HAVE_LIBSAMPLERATE
+	if (pcm_resample_lsr_enabled())
+		return pcm_resample_lsr_16(state, channels,
+					   src_rate, src_buffer, src_size,
+					   dest_rate, dest_size_r,
+					   error_r);
+#else
+	(void)error_r;
+#endif
+
+	return pcm_resample_fallback_16(state, channels,
+					src_rate, src_buffer, src_size,
+					dest_rate, dest_size_r);
+}
+
+const int32_t *
+pcm_resample_32(struct pcm_resample_state *state,
+		unsigned channels,
+		unsigned src_rate, const int32_t *src_buffer, size_t src_size,
+		unsigned dest_rate, size_t *dest_size_r,
+		GError **error_r)
+{
+#ifdef HAVE_LIBSAMPLERATE
+	if (pcm_resample_lsr_enabled())
+		return pcm_resample_lsr_32(state, channels,
+					   src_rate, src_buffer, src_size,
+					   dest_rate, dest_size_r,
+					   error_r);
+#else
+	(void)error_r;
+#endif
+
+	return pcm_resample_fallback_32(state, channels,
+					src_rate, src_buffer, src_size,
+					dest_rate, dest_size_r);
+}

src/pcm/pcm_resample.h

@@ -0,0 +1,164 @@
+/*
+ * 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.
+ */
+
+#ifndef MPD_PCM_RESAMPLE_H
+#define MPD_PCM_RESAMPLE_H
+
+#include "check.h"
+#include "pcm_buffer.h"
+
+#include <stdint.h>
+#include <stddef.h>
+#include <stdbool.h>
+
+#ifdef HAVE_LIBSAMPLERATE
+#include <samplerate.h>
+#endif
+
+/**
+ * This object is statically allocated (within another struct), and
+ * holds buffer allocations and the state for the resampler.
+ */
+struct pcm_resample_state {
+#ifdef HAVE_LIBSAMPLERATE
+	SRC_STATE *state;
+	SRC_DATA data;
+
+	struct pcm_buffer in, out;
+
+	struct {
+		unsigned src_rate;
+		unsigned dest_rate;
+		unsigned channels;
+	} prev;
+
+	int error;
+#endif
+
+	struct pcm_buffer buffer;
+};
+
+bool
+pcm_resample_global_init(GError **error_r);
+
+/**
+ * Initializes a pcm_resample_state object.
+ */
+void pcm_resample_init(struct pcm_resample_state *state);
+
+/**
+ * Deinitializes a pcm_resample_state object and frees allocated
+ * memory.
+ */
+void pcm_resample_deinit(struct pcm_resample_state *state);
+
+/**
+ * @see pcm_convert_reset()
+ */
+void
+pcm_resample_reset(struct pcm_resample_state *state);
+
+/**
+ * Resamples 32 bit float data.
+ *
+ * @param state an initialized pcm_resample_state object
+ * @param channels the number of channels
+ * @param src_rate the source sample rate
+ * @param src the source PCM buffer
+ * @param src_size the size of #src in bytes
+ * @param dest_rate the requested destination sample rate
+ * @param dest_size_r returns the number of bytes of the destination buffer
+ * @return the destination buffer
+ */
+const float *
+pcm_resample_float(struct pcm_resample_state *state,
+		   unsigned channels,
+		   unsigned src_rate,
+		   const float *src_buffer, size_t src_size,
+		   unsigned dest_rate, size_t *dest_size_r,
+		   GError **error_r);
+
+/**
+ * Resamples 16 bit PCM data.
+ *
+ * @param state an initialized pcm_resample_state object
+ * @param channels the number of channels
+ * @param src_rate the source sample rate
+ * @param src the source PCM buffer
+ * @param src_size the size of #src in bytes
+ * @param dest_rate the requested destination sample rate
+ * @param dest_size_r returns the number of bytes of the destination buffer
+ * @return the destination buffer
+ */
+const int16_t *
+pcm_resample_16(struct pcm_resample_state *state,
+		unsigned channels,
+		unsigned src_rate,
+		const int16_t *src_buffer, size_t src_size,
+		unsigned dest_rate, size_t *dest_size_r,
+		GError **error_r);
+
+/**
+ * Resamples 32 bit PCM data.
+ *
+ * @param state an initialized pcm_resample_state object
+ * @param channels the number of channels
+ * @param src_rate the source sample rate
+ * @param src the source PCM buffer
+ * @param src_size the size of #src in bytes
+ * @param dest_rate the requested destination sample rate
+ * @param dest_size_r returns the number of bytes of the destination buffer
+ * @return the destination buffer
+ */
+const int32_t *
+pcm_resample_32(struct pcm_resample_state *state,
+		unsigned channels,
+		unsigned src_rate,
+		const int32_t *src_buffer, size_t src_size,
+		unsigned dest_rate, size_t *dest_size_r,
+		GError **error_r);
+
+/**
+ * Resamples 24 bit PCM data.
+ *
+ * @param state an initialized pcm_resample_state object
+ * @param channels the number of channels
+ * @param src_rate the source sample rate
+ * @param src the source PCM buffer
+ * @param src_size the size of #src in bytes
+ * @param dest_rate the requested destination sample rate
+ * @param dest_size_r returns the number of bytes of the destination buffer
+ * @return the destination buffer
+ */
+static inline const int32_t *
+pcm_resample_24(struct pcm_resample_state *state,
+		unsigned channels,
+		unsigned src_rate,
+		const int32_t *src_buffer, size_t src_size,
+		unsigned dest_rate, size_t *dest_size_r,
+		GError **error_r)
+{
+	/* reuse the 32 bit code - the resampler code doesn't care if
+	   the upper 8 bits are actually used */
+	return pcm_resample_32(state, channels,
+			       src_rate, src_buffer, src_size,
+			       dest_rate, dest_size_r, error_r);
+}
+
+#endif

src/pcm/pcm_resample_fallback.c

@@ -0,0 +1,118 @@
+/*
+ * 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 "pcm_resample_internal.h"
+
+#include <assert.h>
+
+void
+pcm_resample_fallback_init(struct pcm_resample_state *state)
+{
+	pcm_buffer_init(&state->buffer);
+}
+
+void
+pcm_resample_fallback_deinit(struct pcm_resample_state *state)
+{
+	pcm_buffer_deinit(&state->buffer);
+}
+
+/* resampling code blatantly ripped from ESD */
+const int16_t *
+pcm_resample_fallback_16(struct pcm_resample_state *state,
+			 unsigned channels,
+			 unsigned src_rate,
+			 const int16_t *src_buffer, size_t src_size,
+			 unsigned dest_rate,
+			 size_t *dest_size_r)
+{
+	unsigned src_pos, dest_pos = 0;
+	unsigned src_frames = src_size / channels / sizeof(*src_buffer);
+	unsigned dest_frames =
+		(src_frames * dest_rate + src_rate - 1) / src_rate;
+	unsigned dest_samples = dest_frames * channels;
+	size_t dest_size = dest_samples * sizeof(*src_buffer);
+	int16_t *dest_buffer = pcm_buffer_get(&state->buffer, dest_size);
+
+	assert((src_size % (sizeof(*src_buffer) * channels)) == 0);
+
+	switch (channels) {
+	case 1:
+		while (dest_pos < dest_samples) {
+			src_pos = dest_pos * src_rate / dest_rate;
+
+			dest_buffer[dest_pos++] = src_buffer[src_pos];
+		}
+		break;
+	case 2:
+		while (dest_pos < dest_samples) {
+			src_pos = dest_pos * src_rate / dest_rate;
+			src_pos &= ~1;
+
+			dest_buffer[dest_pos++] = src_buffer[src_pos];
+			dest_buffer[dest_pos++] = src_buffer[src_pos + 1];
+		}
+		break;
+	}
+
+	*dest_size_r = dest_size;
+	return dest_buffer;
+}
+
+const int32_t *
+pcm_resample_fallback_32(struct pcm_resample_state *state,
+			 unsigned channels,
+			 unsigned src_rate,
+			 const int32_t *src_buffer, size_t src_size,
+			 unsigned dest_rate,
+			 size_t *dest_size_r)
+{
+	unsigned src_pos, dest_pos = 0;
+	unsigned src_frames = src_size / channels / sizeof(*src_buffer);
+	unsigned dest_frames =
+		(src_frames * dest_rate + src_rate - 1) / src_rate;
+	unsigned dest_samples = dest_frames * channels;
+	size_t dest_size = dest_samples * sizeof(*src_buffer);
+	int32_t *dest_buffer = pcm_buffer_get(&state->buffer, dest_size);
+
+	assert((src_size % (sizeof(*src_buffer) * channels)) == 0);
+
+	switch (channels) {
+	case 1:
+		while (dest_pos < dest_samples) {
+			src_pos = dest_pos * src_rate / dest_rate;
+
+			dest_buffer[dest_pos++] = src_buffer[src_pos];
+		}
+		break;
+	case 2:
+		while (dest_pos < dest_samples) {
+			src_pos = dest_pos * src_rate / dest_rate;
+			src_pos &= ~1;
+
+			dest_buffer[dest_pos++] = src_buffer[src_pos];
+			dest_buffer[dest_pos++] = src_buffer[src_pos + 1];
+		}
+		break;
+	}
+
+	*dest_size_r = dest_size;
+	return dest_buffer;
+}

src/pcm/pcm_resample_internal.h

@@ -0,0 +1,97 @@
+/*
+ * 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.
+ */
+
+/** \file
+ *
+ * Internal declarations for the pcm_resample library.  The "internal"
+ * resampler is called "fallback" in the MPD source, so the file name
+ * of this header is somewhat unrelated to it.
+ */
+
+#ifndef MPD_PCM_RESAMPLE_INTERNAL_H
+#define MPD_PCM_RESAMPLE_INTERNAL_H
+
+#include "check.h"
+#include "pcm_resample.h"
+
+#ifdef HAVE_LIBSAMPLERATE
+
+bool
+pcm_resample_lsr_global_init(const char *converter, GError **error_r);
+
+void
+pcm_resample_lsr_init(struct pcm_resample_state *state);
+
+void
+pcm_resample_lsr_deinit(struct pcm_resample_state *state);
+
+void
+pcm_resample_lsr_reset(struct pcm_resample_state *state);
+
+const float *
+pcm_resample_lsr_float(struct pcm_resample_state *state,
+		       unsigned channels,
+		       unsigned src_rate,
+		       const float *src_buffer, size_t src_size,
+		       unsigned dest_rate, size_t *dest_size_r,
+		       GError **error_r);
+
+const int16_t *
+pcm_resample_lsr_16(struct pcm_resample_state *state,
+		    unsigned channels,
+		    unsigned src_rate,
+		    const int16_t *src_buffer, size_t src_size,
+		    unsigned dest_rate, size_t *dest_size_r,
+		    GError **error_r);
+
+const int32_t *
+pcm_resample_lsr_32(struct pcm_resample_state *state,
+		    unsigned channels,
+		    unsigned src_rate,
+		    const int32_t *src_buffer,
+		    G_GNUC_UNUSED size_t src_size,
+		    unsigned dest_rate, size_t *dest_size_r,
+		    GError **error_r);
+
+#endif
+
+void
+pcm_resample_fallback_init(struct pcm_resample_state *state);
+
+void
+pcm_resample_fallback_deinit(struct pcm_resample_state *state);
+
+const int16_t *
+pcm_resample_fallback_16(struct pcm_resample_state *state,
+			 unsigned channels,
+			 unsigned src_rate,
+			 const int16_t *src_buffer, size_t src_size,
+			 unsigned dest_rate,
+			 size_t *dest_size_r);
+
+const int32_t *
+pcm_resample_fallback_32(struct pcm_resample_state *state,
+			 unsigned channels,
+			 unsigned src_rate,
+			 const int32_t *src_buffer,
+			 G_GNUC_UNUSED size_t src_size,
+			 unsigned dest_rate,
+			 size_t *dest_size_r);
+
+#endif

src/pcm/pcm_resample_libsamplerate.c

@@ -0,0 +1,312 @@
+/*
+ * 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 "pcm_resample_internal.h"
+#include "conf.h"
+
+#include <glib.h>
+
+#include <assert.h>
+#include <stdlib.h>
+#include <string.h>
+
+#undef G_LOG_DOMAIN
+#define G_LOG_DOMAIN "pcm"
+
+static int lsr_converter = SRC_SINC_FASTEST;
+
+static inline GQuark
+libsamplerate_quark(void)
+{
+	return g_quark_from_static_string("libsamplerate");
+}
+
+static bool
+lsr_parse_converter(const char *s)
+{
+	assert(s != NULL);
+
+	if (*s == 0)
+		return true;
+
+	char *endptr;
+	long l = strtol(s, &endptr, 10);
+	if (*endptr == 0 && src_get_name(l) != NULL) {
+		lsr_converter = l;
+		return true;
+	}
+
+	size_t length = strlen(s);
+	for (int i = 0;; ++i) {
+		const char *name = src_get_name(i);
+		if (name == NULL)
+			break;
+
+		if (g_ascii_strncasecmp(s, name, length) == 0) {
+			lsr_converter = i;
+			return true;
+		}
+	}
+
+	return false;
+}
+
+bool
+pcm_resample_lsr_global_init(const char *converter, GError **error_r)
+{
+	if (!lsr_parse_converter(converter)) {
+		g_set_error(error_r, libsamplerate_quark(), 0,
+			    "unknown samplerate converter '%s'", converter);
+		return false;
+	}
+
+	g_debug("libsamplerate converter '%s'",
+		src_get_name(lsr_converter));
+
+	return true;
+}
+
+void
+pcm_resample_lsr_init(struct pcm_resample_state *state)
+{
+	memset(state, 0, sizeof(*state));
+
+	pcm_buffer_init(&state->in);
+	pcm_buffer_init(&state->out);
+	pcm_buffer_init(&state->buffer);
+}
+
+void
+pcm_resample_lsr_deinit(struct pcm_resample_state *state)
+{
+	if (state->state != NULL)
+		state->state = src_delete(state->state);
+
+	pcm_buffer_deinit(&state->in);
+	pcm_buffer_deinit(&state->out);
+	pcm_buffer_deinit(&state->buffer);
+}
+
+void
+pcm_resample_lsr_reset(struct pcm_resample_state *state)
+{
+	if (state->state != NULL)
+		src_reset(state->state);
+}
+
+static bool
+pcm_resample_set(struct pcm_resample_state *state,
+		 unsigned channels, unsigned src_rate, unsigned dest_rate,
+		 GError **error_r)
+{
+	int error;
+	SRC_DATA *data = &state->data;
+
+	/* (re)set the state/ratio if the in or out format changed */
+	if (channels == state->prev.channels &&
+	    src_rate == state->prev.src_rate &&
+	    dest_rate == state->prev.dest_rate)
+		return true;
+
+	state->error = 0;
+	state->prev.channels = channels;
+	state->prev.src_rate = src_rate;
+	state->prev.dest_rate = dest_rate;
+
+	if (state->state)
+		state->state = src_delete(state->state);
+
+	state->state = src_new(lsr_converter, channels, &error);
+	if (!state->state) {
+		g_set_error(error_r, libsamplerate_quark(), state->error,
+			    "libsamplerate initialization has failed: %s",
+			    src_strerror(error));
+		return false;
+	}
+
+	data->src_ratio = (double)dest_rate / (double)src_rate;
+	g_debug("setting samplerate conversion ratio to %.2lf",
+		data->src_ratio);
+	src_set_ratio(state->state, data->src_ratio);
+
+	return true;
+}
+
+static bool
+lsr_process(struct pcm_resample_state *state, GError **error_r)
+{
+	if (state->error == 0)
+		state->error = src_process(state->state, &state->data);
+	if (state->error) {
+		g_set_error(error_r, libsamplerate_quark(), state->error,
+			    "libsamplerate has failed: %s",
+			    src_strerror(state->error));
+		return false;
+	}
+
+	return true;
+}
+
+static float *
+deconst_float_buffer(const float *in)
+{
+	union {
+		const float *in;
+		float *out;
+	} u = { .in = in };
+	return u.out;
+}
+
+const float *
+pcm_resample_lsr_float(struct pcm_resample_state *state,
+		       unsigned channels,
+		       unsigned src_rate,
+		       const float *src_buffer, size_t src_size,
+		       unsigned dest_rate, size_t *dest_size_r,
+		       GError **error_r)
+{
+	assert((src_size % (sizeof(*src_buffer) * channels)) == 0);
+
+	if (!pcm_resample_set(state, channels, src_rate, dest_rate, error_r))
+		return NULL;
+
+	SRC_DATA *data = &state->data;
+	data->input_frames = src_size / sizeof(*src_buffer) / channels;
+	data->data_in = deconst_float_buffer(src_buffer);
+
+	data->output_frames = (src_size * dest_rate + src_rate - 1) / src_rate;
+	size_t data_out_size = data->output_frames * sizeof(float) * channels;
+	data->data_out = pcm_buffer_get(&state->out, data_out_size);
+
+	if (!lsr_process(state, error_r))
+		return NULL;
+
+	*dest_size_r = data->output_frames_gen *
+		sizeof(*data->data_out) * channels;
+	return data->data_out;
+}
+
+const int16_t *
+pcm_resample_lsr_16(struct pcm_resample_state *state,
+		    unsigned channels,
+		    unsigned src_rate,
+		    const int16_t *src_buffer, size_t src_size,
+		    unsigned dest_rate, size_t *dest_size_r,
+		    GError **error_r)
+{
+	bool success;
+	SRC_DATA *data = &state->data;
+	size_t data_in_size;
+	size_t data_out_size;
+	int16_t *dest_buffer;
+
+	assert((src_size % (sizeof(*src_buffer) * channels)) == 0);
+
+	success = pcm_resample_set(state, channels, src_rate, dest_rate,
+				   error_r);
+	if (!success)
+		return NULL;
+
+	data->input_frames = src_size / sizeof(*src_buffer) / channels;
+	data_in_size = data->input_frames * sizeof(float) * channels;
+	data->data_in = pcm_buffer_get(&state->in, data_in_size);
+
+	data->output_frames = (src_size * dest_rate + src_rate - 1) / src_rate;
+	data_out_size = data->output_frames * sizeof(float) * channels;
+	data->data_out = pcm_buffer_get(&state->out, data_out_size);
+
+	src_short_to_float_array(src_buffer, data->data_in,
+				 data->input_frames * channels);
+
+	if (!lsr_process(state, error_r))
+		return NULL;
+
+	*dest_size_r = data->output_frames_gen *
+		sizeof(*dest_buffer) * channels;
+	dest_buffer = pcm_buffer_get(&state->buffer, *dest_size_r);
+	src_float_to_short_array(data->data_out, dest_buffer,
+				 data->output_frames_gen * channels);
+
+	return dest_buffer;
+}
+
+#ifdef HAVE_LIBSAMPLERATE_NOINT
+
+/* libsamplerate introduced these functions in v0.1.3 */
+
+static void
+src_int_to_float_array(const int *in, float *out, int len)
+{
+	while (len-- > 0)
+		*out++ = *in++ / (float)(1 << (24 - 1));
+}
+
+static void
+src_float_to_int_array (const float *in, int *out, int len)
+{
+	while (len-- > 0)
+		*out++ = *in++ * (float)(1 << (24 - 1));
+}
+
+#endif
+
+const int32_t *
+pcm_resample_lsr_32(struct pcm_resample_state *state,
+		    unsigned channels,
+		    unsigned src_rate,
+		    const int32_t *src_buffer, size_t src_size,
+		    unsigned dest_rate, size_t *dest_size_r,
+		    GError **error_r)
+{
+	bool success;
+	SRC_DATA *data = &state->data;
+	size_t data_in_size;
+	size_t data_out_size;
+	int32_t *dest_buffer;
+
+	assert((src_size % (sizeof(*src_buffer) * channels)) == 0);
+
+	success = pcm_resample_set(state, channels, src_rate, dest_rate,
+				   error_r);
+	if (!success)
+		return NULL;
+
+	data->input_frames = src_size / sizeof(*src_buffer) / channels;
+	data_in_size = data->input_frames * sizeof(float) * channels;
+	data->data_in = pcm_buffer_get(&state->in, data_in_size);
+
+	data->output_frames = (src_size * dest_rate + src_rate - 1) / src_rate;
+	data_out_size = data->output_frames * sizeof(float) * channels;
+	data->data_out = pcm_buffer_get(&state->out, data_out_size);
+
+	src_int_to_float_array(src_buffer, data->data_in,
+			       data->input_frames * channels);
+
+	if (!lsr_process(state, error_r))
+		return NULL;
+
+	*dest_size_r = data->output_frames_gen *
+		sizeof(*dest_buffer) * channels;
+	dest_buffer = pcm_buffer_get(&state->buffer, *dest_size_r);
+	src_float_to_int_array(data->data_out, dest_buffer,
+			       data->output_frames_gen * channels);
+
+	return dest_buffer;
+}