pcm/PcmResampler: convert to abstract interface

The PcmResampler interface is implemented by the two classes
FallbackPcmResampler and LibsampleratePcmResampler.  This prepares for
adding more resampler libraries.

Makefile.am

@@ -339,10 +339,10 @@ libpcm_a_SOURCES = \
 	src/pcm/PcmFormat.cxx src/pcm/PcmFormat.hxx \
 	src/pcm/FormatConverter.cxx src/pcm/FormatConverter.hxx \
 	src/pcm/ChannelsConverter.cxx src/pcm/ChannelsConverter.hxx \
+	src/pcm/Resampler.hxx \
 	src/pcm/GlueResampler.cxx src/pcm/GlueResampler.hxx \
-	src/pcm/PcmResample.cxx src/pcm/PcmResample.hxx \
+	src/pcm/FallbackResampler.cxx src/pcm/FallbackResampler.hxx \
-	src/pcm/PcmResampleFallback.cxx \
+	src/pcm/ConfiguredResampler.cxx src/pcm/ConfiguredResampler.hxx \
-	src/pcm/PcmResampleInternal.hxx \
 	src/pcm/PcmDither.cxx src/pcm/PcmDither.hxx \
 	src/pcm/PcmPrng.hxx \
 	src/pcm/PcmUtils.hxx
@@ -354,7 +354,8 @@ PCM_LIBS = \
 	$(SAMPLERATE_LIBS)
 
 if HAVE_LIBSAMPLERATE
-libpcm_a_SOURCES += src/pcm/PcmResampleLibsamplerate.cxx
+libpcm_a_SOURCES += \
+	src/pcm/LibsamplerateResampler.cxx src/pcm/LibsamplerateResampler.hxx
 endif
 
 # File system library

src/pcm/ConfiguredResampler.cxx

@@ -0,0 +1,63 @@
+/*
+ * 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 "ConfiguredResampler.hxx"
+#include "FallbackResampler.hxx"
+
+#ifdef HAVE_LIBSAMPLERATE
+#include "LibsamplerateResampler.hxx"
+#include "ConfigGlobal.hxx"
+#include "ConfigOption.hxx"
+#endif
+
+#include <string.h>
+
+#ifdef HAVE_LIBSAMPLERATE
+static bool lsr_enabled;
+#endif
+
+bool
+pcm_resampler_global_init(Error &error)
+{
+#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);
+	else
+		return true;
+#else
+	(void)error;
+	return true;
+#endif
+}
+
+PcmResampler *
+pcm_resampler_create()
+{
+#ifdef HAVE_LIBSAMPLERATE
+	if (lsr_enabled)
+		return new LibsampleratePcmResampler();
+#endif
+
+	return new FallbackPcmResampler();
+}

src/pcm/ConfiguredResampler.hxx

@@ -0,0 +1,38 @@
+/*
+ * 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_CONFIGURED_RESAMPLER_HXX
+#define MPD_CONFIGURED_RESAMPLER_HXX
+
+#include "check.h"
+
+class Error;
+class PcmResampler;
+
+bool
+pcm_resampler_global_init(Error &error);
+
+/**
+ * Create a #PcmResampler instance from the implementation class
+ * configured in mpd.conf.
+ */
+PcmResampler *
+pcm_resampler_create();
+
+#endif

src/pcm/FallbackResampler.cxx

@@ -0,0 +1,147 @@
+/*
+ * 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 "FallbackResampler.hxx"
+
+#include <assert.h>
+
+AudioFormat
+FallbackPcmResampler::Open(AudioFormat &af, unsigned new_sample_rate,
+			   gcc_unused Error &error)
+{
+	assert(af.IsValid());
+	assert(audio_valid_sample_rate(new_sample_rate));
+
+	switch (af.format) {
+	case SampleFormat::UNDEFINED:
+		assert(false);
+		gcc_unreachable();
+
+	case SampleFormat::S8:
+		af.format = SampleFormat::S16;
+		break;
+
+	case SampleFormat::S16:
+	case SampleFormat::FLOAT:
+	case SampleFormat::S24_P32:
+	case SampleFormat::S32:
+		break;
+
+	case SampleFormat::DSD:
+		af.format = SampleFormat::FLOAT;
+		break;
+	}
+
+	format = af;
+	out_rate = new_sample_rate;
+
+	AudioFormat result = af;
+	result.sample_rate = new_sample_rate;
+	return result;
+}
+
+void
+FallbackPcmResampler::Close()
+{
+}
+
+template<typename T>
+static ConstBuffer<T>
+pcm_resample_fallback(PcmBuffer &buffer,
+		      unsigned channels,
+		      unsigned src_rate,
+		      ConstBuffer<T> src,
+		      unsigned dest_rate)
+{
+	unsigned dest_pos = 0;
+	unsigned src_frames = src.size / channels;
+	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.data);
+	T *dest_buffer = (T *)buffer.Get(dest_size);
+
+	assert((src.size % channels) == 0);
+
+	switch (channels) {
+	case 1:
+		while (dest_pos < dest_samples) {
+			unsigned src_pos = dest_pos * src_rate / dest_rate;
+
+			dest_buffer[dest_pos++] = src.data[src_pos];
+		}
+		break;
+	case 2:
+		while (dest_pos < dest_samples) {
+			unsigned src_pos = dest_pos * src_rate / dest_rate;
+			src_pos &= ~1;
+
+			dest_buffer[dest_pos++] = src.data[src_pos];
+			dest_buffer[dest_pos++] = src.data[src_pos + 1];
+		}
+		break;
+	}
+
+	return { dest_buffer, dest_samples };
+}
+
+template<typename T>
+static ConstBuffer<void>
+pcm_resample_fallback_void(PcmBuffer &buffer,
+			   unsigned channels,
+			   unsigned src_rate,
+			   ConstBuffer<void> src,
+			   unsigned dest_rate)
+{
+	const auto typed_src = ConstBuffer<T>::FromVoid(src);
+	return pcm_resample_fallback(buffer, channels, src_rate, typed_src,
+				     dest_rate);
+}
+
+ConstBuffer<void>
+FallbackPcmResampler::Resample(ConstBuffer<void> src, gcc_unused Error &error)
+{
+	switch (format.format) {
+	case SampleFormat::UNDEFINED:
+	case SampleFormat::S8:
+	case SampleFormat::DSD:
+		assert(false);
+		gcc_unreachable();
+
+	case SampleFormat::S16:
+		return pcm_resample_fallback_void<int16_t>(buffer,
+							   format.channels,
+							   format.sample_rate,
+							   src,
+							   out_rate);
+
+	case SampleFormat::FLOAT:
+	case SampleFormat::S24_P32:
+	case SampleFormat::S32:
+		return pcm_resample_fallback_void<int32_t>(buffer,
+							   format.channels,
+							   format.sample_rate,
+							   src,
+							   out_rate);
+	}
+
+	assert(false);
+	gcc_unreachable();
+}

src/pcm/FallbackResampler.hxx

@@ -0,0 +1,45 @@
+/*
+ * 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_FALLBACK_RESAMPLER_HXX
+#define MPD_PCM_FALLBACK_RESAMPLER_HXX
+
+#include "Resampler.hxx"
+#include "PcmBuffer.hxx"
+#include "AudioFormat.hxx"
+
+/**
+ * A naive resampler that is used when no external library was found
+ * (or when the user explicitly asks for bad quality).
+ */
+class FallbackPcmResampler final : public PcmResampler {
+	AudioFormat format;
+	unsigned out_rate;
+
+	PcmBuffer buffer;
+
+public:
+	virtual AudioFormat Open(AudioFormat &af, unsigned new_sample_rate,
+				 Error &error) override;
+	virtual void Close() override;
+	virtual ConstBuffer<void> Resample(ConstBuffer<void> src,
+					   Error &error) override;
+};
+
+#endif

src/pcm/GlueResampler.cxx

@@ -19,76 +19,67 @@
 
 #include "config.h"
 #include "GlueResampler.hxx"
-#include "PcmConvert.hxx"
+#include "ConfiguredResampler.hxx"
-#include "PcmFormat.hxx"
+#include "Resampler.hxx"
-#include "util/ConstBuffer.hxx"
+
-#include "util/Error.hxx"
+#include <assert.h>
+
+GluePcmResampler::GluePcmResampler()
+	:resampler(pcm_resampler_create()) {}
+
+GluePcmResampler::~GluePcmResampler()
+{
+	delete resampler;
+}
 
 bool
-GluePcmResampler::Open(AudioFormat _src_format, unsigned _new_sample_rate,
+GluePcmResampler::Open(AudioFormat src_format, unsigned new_sample_rate,
-		       gcc_unused Error &error)
+		       Error &error)
 {
-	src_format = _src_format;
+	assert(src_format.IsValid());
-	new_sample_rate = _new_sample_rate;
+	assert(audio_valid_sample_rate(new_sample_rate));
 
+	AudioFormat requested_format = src_format;
+	AudioFormat dest_format = resampler->Open(requested_format,
+						  new_sample_rate,
+						  error);
+	if (!dest_format.IsValid())
+		return false;
+
+	assert(requested_format.channels == src_format.channels);
+	assert(dest_format.channels == src_format.channels);
+	assert(dest_format.sample_rate == new_sample_rate);
+
+	if (requested_format.format != src_format.format &&
+	    !format_converter.Open(src_format.format, requested_format.format,
+				   error))
+		return false;
+
+	src_sample_format = src_format.format;
+	requested_sample_format = requested_format.format;
+	output_sample_format = dest_format.format;
 	return true;
 }
 
 void
 GluePcmResampler::Close()
 {
-	resampler.Reset();
+	if (requested_sample_format != src_sample_format)
+		format_converter.Close();
+
+	resampler->Close();
 }
 
 ConstBuffer<void>
 GluePcmResampler::Resample(ConstBuffer<void> src, Error &error)
 {
-	const void *result;
+	assert(!src.IsNull());
-	size_t size;
 
-	switch (src_format.format) {
+	if (requested_sample_format != src_sample_format) {
-	case SampleFormat::S16:
+		src = format_converter.Convert(src, error);
-		result = resampler.Resample16(src_format.channels,
+		if (src.IsNull())
-					      src_format.sample_rate,
+			return nullptr;
-					      (const int16_t *)src.data,
-					      src.size,
-					      new_sample_rate, &size,
-					      error);
-		break;
-
-	case SampleFormat::S24_P32:
-		result = resampler.Resample24(src_format.channels,
-					      src_format.sample_rate,
-					      (const int32_t *)src.data,
-					      src.size,
-					      new_sample_rate, &size,
-					      error);
-		break;
-
-	case SampleFormat::S32:
-		result = resampler.Resample24(src_format.channels,
-					      src_format.sample_rate,
-					      (const int32_t *)src.data,
-					      src.size,
-					      new_sample_rate, &size,
-					      error);
-		break;
-
-	case SampleFormat::FLOAT:
-		result = resampler.ResampleFloat(src_format.channels,
-						 src_format.sample_rate,
-						 (const float *)src.data,
-						 src.size,
-						 new_sample_rate, &size,
-						 error);
-		break;
-
-	default:
-		error.Format(pcm_convert_domain,
-			     "Resampling %s is not implemented",
-			     sample_format_to_string(src_format.format));
-		return nullptr;
 	}
 
-	return { result, size };
+	return resampler->Resample(src, error);
 }

src/pcm/GlueResampler.hxx

@@ -22,22 +22,41 @@
 
 #include "check.h"
 #include "AudioFormat.hxx"
-#include "PcmResample.hxx"
+#include "FormatConverter.hxx"
 
 class Error;
+class PcmResampler;
 template<typename T> struct ConstBuffer;
 
+/**
+ * A glue class that integrates a #PcmResampler and automatically
+ * converts source data to the sample format required by the
+ * #PcmResampler instance.
+ */
 class GluePcmResampler {
-	PcmResampler resampler;
+	PcmResampler *const resampler;
 
-	AudioFormat src_format;
+	SampleFormat src_sample_format, requested_sample_format;
-	unsigned new_sample_rate;
+	SampleFormat output_sample_format;
+
+	/**
+	 * This object converts input data to the sample format
+	 * requested by the #PcmResampler.
+	 */
+	PcmFormatConverter format_converter;
 
 public:
+	GluePcmResampler();
+	~GluePcmResampler();
+
 	bool Open(AudioFormat src_format, unsigned new_sample_rate,
 		  Error &error);
 	void Close();
 
+	SampleFormat GetOutputSampleFormat() const {
+		return output_sample_format;
+	}
+
 	ConstBuffer<void> Resample(ConstBuffer<void> src, Error &error);
 };
 

src/pcm/LibsamplerateResampler.cxx

@@ -0,0 +1,163 @@
+/*
+ * 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 "LibsamplerateResampler.hxx"
+#include "util/ASCII.hxx"
+#include "util/Error.hxx"
+#include "util/Domain.hxx"
+#include "Log.hxx"
+
+#include <assert.h>
+#include <stdlib.h>
+#include <string.h>
+
+static constexpr Domain libsamplerate_domain("libsamplerate");
+
+static int lsr_converter = SRC_SINC_FASTEST;
+
+static bool
+lsr_parse_converter(const char *s)
+{
+	assert(s != nullptr);
+
+	if (*s == 0)
+		return true;
+
+	char *endptr;
+	long l = strtol(s, &endptr, 10);
+	if (*endptr == 0 && src_get_name(l) != nullptr) {
+		lsr_converter = l;
+		return true;
+	}
+
+	size_t length = strlen(s);
+	for (int i = 0;; ++i) {
+		const char *name = src_get_name(i);
+		if (name == nullptr)
+			break;
+
+		if (StringEqualsCaseASCII(s, name, length)) {
+			lsr_converter = i;
+			return true;
+		}
+	}
+
+	return false;
+}
+
+bool
+pcm_resample_lsr_global_init(const char *converter, Error &error)
+{
+	if (!lsr_parse_converter(converter)) {
+		error.Format(libsamplerate_domain,
+			     "unknown samplerate converter '%s'", converter);
+		return false;
+	}
+
+	FormatDebug(libsamplerate_domain,
+		    "libsamplerate converter '%s'",
+		    src_get_name(lsr_converter));
+
+	return true;
+}
+
+AudioFormat
+LibsampleratePcmResampler::Open(AudioFormat &af, unsigned new_sample_rate,
+				Error &error)
+{
+	assert(af.IsValid());
+	assert(audio_valid_sample_rate(new_sample_rate));
+
+	src_rate = af.sample_rate;
+	dest_rate = new_sample_rate;
+	channels = af.channels;
+
+	/* libsamplerate works with floating point samples */
+	af.format = SampleFormat::FLOAT;
+
+	int src_error;
+	state = src_new(lsr_converter, channels, &src_error);
+	if (!state) {
+		error.Format(libsamplerate_domain, src_error,
+			     "libsamplerate initialization has failed: %s",
+			     src_strerror(src_error));
+		return AudioFormat::Undefined();
+	}
+
+	memset(&data, 0, sizeof(data));
+
+	data.src_ratio = double(new_sample_rate) / double(af.sample_rate);
+	FormatDebug(libsamplerate_domain,
+		    "setting samplerate conversion ratio to %.2lf",
+		    data.src_ratio);
+	src_set_ratio(state, data.src_ratio);
+
+	AudioFormat result = af;
+	result.sample_rate = new_sample_rate;
+	return result;
+}
+
+void
+LibsampleratePcmResampler::Close()
+{
+	state = src_delete(state);
+}
+
+static bool
+src_process(SRC_STATE *state, SRC_DATA *data, Error &error)
+{
+	int result = src_process(state, data);
+	if (result != 0) {
+		error.Format(libsamplerate_domain, result,
+			     "libsamplerate has failed: %s",
+			     src_strerror(result));
+		return false;
+	}
+
+	return true;
+}
+
+inline ConstBuffer<float>
+LibsampleratePcmResampler::Resample2(ConstBuffer<float> src, Error &error)
+{
+	assert(src.size % channels == 0);
+
+	const unsigned src_frames = src.size / channels;
+	const unsigned dest_frames =
+		(src_frames * dest_rate + src_rate - 1) / src_rate;
+	size_t data_out_size = dest_frames * sizeof(float) * channels;
+
+	data.data_in = const_cast<float *>(src.data);
+	data.data_out = (float *)buffer.Get(data_out_size);
+	data.input_frames = src_frames;
+	data.output_frames = dest_frames;
+
+	if (!src_process(state, &data, error))
+		return nullptr;
+
+	return ConstBuffer<float>(data.data_out,
+				  data.output_frames_gen * channels);
+}
+
+ConstBuffer<void>
+LibsampleratePcmResampler::Resample(ConstBuffer<void> src, Error &error)
+{
+	return Resample2(ConstBuffer<float>::FromVoid(src), error).ToVoid();
+}

src/pcm/LibsamplerateResampler.hxx

@@ -0,0 +1,55 @@
+/*
+ * 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_LIBSAMPLERATE_RESAMPLER_HXX
+#define MPD_PCM_LIBSAMPLERATE_RESAMPLER_HXX
+
+#include "Resampler.hxx"
+#include "PcmBuffer.hxx"
+#include "AudioFormat.hxx"
+
+#include <samplerate.h>
+
+/**
+ * A resampler using libsamplerate.
+ */
+class LibsampleratePcmResampler final : public PcmResampler {
+	unsigned src_rate, dest_rate;
+	unsigned channels;
+
+	SRC_STATE *state;
+	SRC_DATA data;
+
+	PcmBuffer buffer;
+
+public:
+	virtual AudioFormat Open(AudioFormat &af, unsigned new_sample_rate,
+				 Error &error) override;
+	virtual void Close() override;
+	virtual ConstBuffer<void> Resample(ConstBuffer<void> src,
+					   Error &error) override;
+
+private:
+	ConstBuffer<float> Resample2(ConstBuffer<float> src, Error &error);
+};
+
+bool
+pcm_resample_lsr_global_init(const char *converter, Error &error);
+
+#endif

src/pcm/PcmConvert.cxx

@@ -19,6 +19,7 @@
 
 #include "config.h"
 #include "PcmConvert.hxx"
+#include "ConfiguredResampler.hxx"
 #include "AudioFormat.hxx"
 #include "util/ConstBuffer.hxx"
 #include "util/Error.hxx"
@@ -33,7 +34,7 @@ const Domain pcm_convert_domain("pcm_convert");
 bool
 pcm_convert_global_init(Error &error)
 {
-	return pcm_resample_global_init(error);
+	return pcm_resampler_global_init(error);
 }
 
 PcmConvert::PcmConvert()
@@ -66,21 +67,35 @@ PcmConvert::Open(AudioFormat _src_format, AudioFormat _dest_format,
 	if (format.format == SampleFormat::DSD)
 		format.format = SampleFormat::FLOAT;
 
-	if (format.format != dest_format.format &&
+	enable_resampler = format.sample_rate != dest_format.sample_rate;
-	    !format_converter.Open(format.format, dest_format.format, error))
+	if (enable_resampler) {
-		return false;
+		if (!resampler.Open(format, dest_format.sample_rate, error))
-	format.format = dest_format.format;
+			return false;
 
-	if (format.channels != dest_format.channels &&
+		format.format = resampler.GetOutputSampleFormat();
-	    !channels_converter.Open(format.format, format.channels,
+		format.sample_rate = dest_format.sample_rate;
-				     dest_format.channels, error)) {
+	}
-		format_converter.Close();
+
+	enable_format = format.format != dest_format.format;
+	if (enable_format &&
+	    !format_converter.Open(format.format, dest_format.format, error)) {
+		if (enable_resampler)
+			resampler.Close();
 		return false;
 	}
 
-	if (format.sample_rate != dest_format.sample_rate &&
+	format.format = dest_format.format;
-	    !resampler.Open(format, dest_format.sample_rate, error))
+
+	enable_channels = format.channels != dest_format.channels;
+	if (enable_channels &&
+	    !channels_converter.Open(format.format, format.channels,
+				     dest_format.channels, error)) {
+		if (enable_format)
+			format_converter.Close();
+		if (enable_resampler)
+			resampler.Close();
 		return false;
+	}
 
 	return true;
 }
@@ -88,17 +103,15 @@ PcmConvert::Open(AudioFormat _src_format, AudioFormat _dest_format,
 void
 PcmConvert::Close()
 {
-	if (src_format.channels != dest_format.channels)
+	if (enable_channels)
 		channels_converter.Close();
-
+	if (enable_format)
-	if (src_format.format != dest_format.format)
 		format_converter.Close();
+	if (enable_resampler)
+		resampler.Close();
 
 	dsd.Reset();
 
-	if (src_format.sample_rate != dest_format.sample_rate)
-		resampler.Close();
-
 #ifndef NDEBUG
 	src_format.Clear();
 	dest_format.Clear();
@@ -127,7 +140,16 @@ PcmConvert::Convert(const void *src, size_t src_size,
 		format.format = SampleFormat::FLOAT;
 	}
 
-	if (format.format != dest_format.format) {
+	if (enable_resampler) {
+		buffer = resampler.Resample(buffer, error);
+		if (buffer.IsNull())
+			return nullptr;
+
+		format.format = resampler.GetOutputSampleFormat();
+		format.sample_rate = dest_format.sample_rate;
+	}
+
+	if (enable_format) {
 		buffer = format_converter.Convert(buffer, error);
 		if (buffer.IsNull())
 			return nullptr;
@@ -135,7 +157,7 @@ PcmConvert::Convert(const void *src, size_t src_size,
 		format.format = dest_format.format;
 	}
 
-	if (format.channels != dest_format.channels) {
+	if (enable_channels) {
 		buffer = channels_converter.Convert(buffer, error);
 		if (buffer.IsNull())
 			return nullptr;
@@ -143,14 +165,6 @@ PcmConvert::Convert(const void *src, size_t src_size,
 		format.channels = dest_format.channels;
 	}
 
-	if (format.sample_rate != dest_format.sample_rate) {
-		buffer = resampler.Resample(buffer, error);
-		if (buffer.IsNull())
-			return nullptr;
-
-		format.sample_rate = dest_format.sample_rate;
-	}
-
 	*dest_size_r = buffer.size;
 	return buffer.data;
 }

src/pcm/PcmConvert.hxx

@@ -41,12 +41,14 @@ class Domain;
 class PcmConvert {
 	PcmDsd dsd;
 
+	GluePcmResampler resampler;
 	PcmFormatConverter format_converter;
 	PcmChannelsConverter channels_converter;
-	GluePcmResampler resampler;
 
 	AudioFormat src_format, dest_format;
 
+	bool enable_resampler, enable_format, enable_channels;
+
 public:
 	PcmConvert();
 	~PcmConvert();

src/pcm/PcmFormat.cxx

@@ -442,7 +442,6 @@ ConvertToFloat(float *dest,
 	constexpr float factor = 0.5 / (1 << (Traits::BITS - 2));
 	while (src != end)
 		*dest++ = float(*src++) * factor;
-
 }
 
 template<SampleFormat F, class Traits=SampleTraits<F>>

src/pcm/PcmResample.cxx

@@ -1,173 +0,0 @@
-/*
- * 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 "PcmResampleInternal.hxx"
-
-#ifdef HAVE_LIBSAMPLERATE
-#include "ConfigGlobal.hxx"
-#include "ConfigOption.hxx"
-#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(Error &error)
-{
-#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);
-	else
-		return true;
-#else
-	(void)error;
-	return true;
-#endif
-}
-
-PcmResampler::PcmResampler()
-{
-#ifdef HAVE_LIBSAMPLERATE
-	if (pcm_resample_lsr_enabled())
-		pcm_resample_lsr_init(this);
-#endif
-}
-
-PcmResampler::~PcmResampler()
-{
-#ifdef HAVE_LIBSAMPLERATE
-	if (pcm_resample_lsr_enabled())
-		pcm_resample_lsr_deinit(this);
-#endif
-}
-
-void
-PcmResampler::Reset()
-{
-#ifdef HAVE_LIBSAMPLERATE
-	pcm_resample_lsr_reset(this);
-#endif
-}
-
-const float *
-PcmResampler::ResampleFloat(unsigned channels, unsigned src_rate,
-			    const float *src_buffer, size_t src_size,
-			    unsigned dest_rate, size_t *dest_size_r,
-			    Error &error_r)
-{
-#ifdef HAVE_LIBSAMPLERATE
-	if (pcm_resample_lsr_enabled())
-		return pcm_resample_lsr_float(this, 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(buffer, channels,
-					 src_rate, (const int32_t *)src_buffer,
-					 src_size,
-					 dest_rate, dest_size_r);
-}
-
-const int16_t *
-PcmResampler::Resample16(unsigned channels,
-			 unsigned src_rate, const int16_t *src_buffer, size_t src_size,
-			 unsigned dest_rate, size_t *dest_size_r,
-			 Error &error_r)
-{
-#ifdef HAVE_LIBSAMPLERATE
-	if (pcm_resample_lsr_enabled())
-		return pcm_resample_lsr_16(this, channels,
-					   src_rate, src_buffer, src_size,
-					   dest_rate, dest_size_r,
-					   error_r);
-#else
-	(void)error_r;
-#endif
-
-	return pcm_resample_fallback_16(buffer, channels,
-					src_rate, src_buffer, src_size,
-					dest_rate, dest_size_r);
-}
-
-const int32_t *
-PcmResampler::Resample32(unsigned channels, unsigned src_rate,
-			 const int32_t *src_buffer, size_t src_size,
-			 unsigned dest_rate, size_t *dest_size_r,
-			 Error &error_r)
-{
-#ifdef HAVE_LIBSAMPLERATE
-	if (pcm_resample_lsr_enabled())
-		return pcm_resample_lsr_32(this, channels,
-					   src_rate, src_buffer, src_size,
-					   dest_rate, dest_size_r,
-					   error_r);
-#else
-	(void)error_r;
-#endif
-
-	return pcm_resample_fallback_32(buffer, channels,
-					src_rate, src_buffer, src_size,
-					dest_rate, dest_size_r);
-}
-
-const int32_t *
-PcmResampler::Resample24(unsigned channels, unsigned src_rate,
-			 const int32_t *src_buffer, size_t src_size,
-			 unsigned dest_rate, size_t *dest_size_r,
-			 Error &error_r)
-{
-#ifdef HAVE_LIBSAMPLERATE
-	if (pcm_resample_lsr_enabled())
-		return pcm_resample_lsr_24(this, channels,
-					   src_rate, src_buffer, src_size,
-					   dest_rate, dest_size_r,
-					   error_r);
-#else
-	(void)error_r;
-#endif
-
-	/* reuse the 32 bit code - the resampler code doesn't care if
-	   the upper 8 bits are actually used */
-	return pcm_resample_fallback_32(buffer, channels,
-					src_rate, src_buffer, src_size,
-					dest_rate, dest_size_r);
-}

src/pcm/PcmResample.hxx

@@ -1,133 +0,0 @@
-/*
- * 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_RESAMPLE_HXX
-#define MPD_PCM_RESAMPLE_HXX
-
-#include "check.h"
-#include "PcmBuffer.hxx"
-
-#include <stdint.h>
-#include <stddef.h>
-
-#ifdef HAVE_LIBSAMPLERATE
-#include <samplerate.h>
-#endif
-
-class Error;
-
-/**
- * This object is statically allocated (within another struct), and
- * holds buffer allocations and the state for the resampler.
- */
-struct PcmResampler {
-#ifdef HAVE_LIBSAMPLERATE
-	SRC_STATE *state;
-	SRC_DATA data;
-
-	PcmBuffer in, out;
-
-	struct {
-		unsigned src_rate;
-		unsigned dest_rate;
-		unsigned channels;
-	} prev;
-
-	int error;
-#endif
-
-	PcmBuffer buffer;
-
-	PcmResampler();
-	~PcmResampler();
-
-	/**
-	 * @see pcm_convert_reset()
-	 */
-	void Reset();
-
-	/**
-	 * Resamples 32 bit float data.
-	 *
-	 * @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 *ResampleFloat(unsigned channels, unsigned src_rate,
-				   const float *src_buffer, size_t src_size,
-				   unsigned dest_rate, size_t *dest_size_r,
-				   Error &error_r);
-
-	/**
-	 * Resamples 16 bit PCM data.
-	 *
-	 * @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 *Resample16(unsigned channels, unsigned src_rate,
-				  const int16_t *src_buffer, size_t src_size,
-				  unsigned dest_rate, size_t *dest_size_r,
-				  Error &error_r);
-
-	/**
-	 * Resamples 32 bit PCM data.
-	 *
-	 * @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 *Resample32(unsigned channels, unsigned src_rate,
-				  const int32_t *src_buffer, size_t src_size,
-				  unsigned dest_rate, size_t *dest_size_r,
-				  Error &error_r);
-
-	/**
-	 * Resamples 24 bit PCM data.
-	 *
-	 * @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 *Resample24(unsigned channels, unsigned src_rate,
-				  const int32_t *src_buffer, size_t src_size,
-				  unsigned dest_rate, size_t *dest_size_r,
-				  Error &error_r);
-};
-
-bool
-pcm_resample_global_init(Error &error);
-
-#endif

src/pcm/PcmResampleFallback.cxx

@@ -1,106 +0,0 @@
-/*
- * 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 "PcmResampleInternal.hxx"
-
-#include <assert.h>
-
-/* resampling code blatantly ripped from ESD */
-const int16_t *
-pcm_resample_fallback_16(PcmBuffer &buffer,
-			 unsigned channels,
-			 unsigned src_rate,
-			 const int16_t *src_buffer, size_t src_size,
-			 unsigned dest_rate,
-			 size_t *dest_size_r)
-{
-	unsigned 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 = (int16_t *)buffer.Get(dest_size);
-
-	assert((src_size % (sizeof(*src_buffer) * channels)) == 0);
-
-	switch (channels) {
-	case 1:
-		while (dest_pos < dest_samples) {
-			unsigned src_pos = dest_pos * src_rate / dest_rate;
-
-			dest_buffer[dest_pos++] = src_buffer[src_pos];
-		}
-		break;
-	case 2:
-		while (dest_pos < dest_samples) {
-			unsigned 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(PcmBuffer &buffer,
-			 unsigned channels,
-			 unsigned src_rate,
-			 const int32_t *src_buffer, size_t src_size,
-			 unsigned dest_rate,
-			 size_t *dest_size_r)
-{
-	unsigned 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 = (int32_t *)buffer.Get(dest_size);
-
-	assert((src_size % (sizeof(*src_buffer) * channels)) == 0);
-
-	switch (channels) {
-	case 1:
-		while (dest_pos < dest_samples) {
-			unsigned src_pos = dest_pos * src_rate / dest_rate;
-
-			dest_buffer[dest_pos++] = src_buffer[src_pos];
-		}
-		break;
-	case 2:
-		while (dest_pos < dest_samples) {
-			unsigned 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/PcmResampleInternal.hxx

@@ -1,100 +0,0 @@
-/*
- * 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.
- */
-
-/** \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_HXX
-#define MPD_PCM_RESAMPLE_INTERNAL_HXX
-
-#include "check.h"
-#include "PcmResample.hxx"
-
-#ifdef HAVE_LIBSAMPLERATE
-
-bool
-pcm_resample_lsr_global_init(const char *converter, Error &error);
-
-void
-pcm_resample_lsr_init(PcmResampler *state);
-
-void
-pcm_resample_lsr_deinit(PcmResampler *state);
-
-void
-pcm_resample_lsr_reset(PcmResampler *state);
-
-const float *
-pcm_resample_lsr_float(PcmResampler *state,
-		       unsigned channels,
-		       unsigned src_rate,
-		       const float *src_buffer, size_t src_size,
-		       unsigned dest_rate, size_t *dest_size_r,
-		       Error &error);
-
-const int16_t *
-pcm_resample_lsr_16(PcmResampler *state,
-		    unsigned channels,
-		    unsigned src_rate,
-		    const int16_t *src_buffer, size_t src_size,
-		    unsigned dest_rate, size_t *dest_size_r,
-		    Error &error);
-
-const int32_t *
-pcm_resample_lsr_32(PcmResampler *state,
-		    unsigned channels,
-		    unsigned src_rate,
-		    const int32_t *src_buffer,
-		    size_t src_size,
-		    unsigned dest_rate, size_t *dest_size_r,
-		    Error &error);
-
-const int32_t *
-pcm_resample_lsr_24(PcmResampler *state,
-		    unsigned channels,
-		    unsigned src_rate,
-		    const int32_t *src_buffer,
-		    size_t src_size,
-		    unsigned dest_rate, size_t *dest_size_r,
-		    Error &error);
-
-#endif
-
-const int16_t *
-pcm_resample_fallback_16(PcmBuffer &buffer,
-			 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(PcmBuffer &buffer,
-			 unsigned channels,
-			 unsigned src_rate,
-			 const int32_t *src_buffer,
-			 size_t src_size,
-			 unsigned dest_rate,
-			 size_t *dest_size_r);
-
-#endif

src/pcm/PcmResampleLibsamplerate.cxx

@@ -1,290 +0,0 @@
-/*
- * 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 "PcmResampleInternal.hxx"
-#include "PcmUtils.hxx"
-#include "util/ASCII.hxx"
-#include "util/Error.hxx"
-#include "util/Domain.hxx"
-#include "Log.hxx"
-
-#include <assert.h>
-#include <stdlib.h>
-#include <string.h>
-
-static int lsr_converter = SRC_SINC_FASTEST;
-
-static constexpr Domain libsamplerate_domain("libsamplerate");
-
-static bool
-lsr_parse_converter(const char *s)
-{
-	assert(s != nullptr);
-
-	if (*s == 0)
-		return true;
-
-	char *endptr;
-	long l = strtol(s, &endptr, 10);
-	if (*endptr == 0 && src_get_name(l) != nullptr) {
-		lsr_converter = l;
-		return true;
-	}
-
-	size_t length = strlen(s);
-	for (int i = 0;; ++i) {
-		const char *name = src_get_name(i);
-		if (name == nullptr)
-			break;
-
-		if (StringEqualsCaseASCII(s, name, length)) {
-			lsr_converter = i;
-			return true;
-		}
-	}
-
-	return false;
-}
-
-bool
-pcm_resample_lsr_global_init(const char *converter, Error &error)
-{
-	if (!lsr_parse_converter(converter)) {
-		error.Format(libsamplerate_domain,
-			     "unknown samplerate converter '%s'", converter);
-		return false;
-	}
-
-	FormatDebug(libsamplerate_domain,
-		    "libsamplerate converter '%s'",
-		    src_get_name(lsr_converter));
-
-	return true;
-}
-
-void
-pcm_resample_lsr_init(PcmResampler *state)
-{
-	state->state = nullptr;
-	memset(&state->data, 0, sizeof(state->data));
-	memset(&state->prev, 0, sizeof(state->prev));
-	state->error = 0;
-}
-
-void
-pcm_resample_lsr_deinit(PcmResampler *state)
-{
-	if (state->state != nullptr)
-		state->state = src_delete(state->state);
-}
-
-void
-pcm_resample_lsr_reset(PcmResampler *state)
-{
-	if (state->state != nullptr)
-		src_reset(state->state);
-}
-
-static bool
-pcm_resample_set(PcmResampler *state,
-		 unsigned channels, unsigned src_rate, unsigned dest_rate,
-		 Error &error_r)
-{
-	/* (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);
-
-	int error;
-	state->state = src_new(lsr_converter, channels, &error);
-	if (!state->state) {
-		error_r.Format(libsamplerate_domain, error,
-			       "libsamplerate initialization has failed: %s",
-			       src_strerror(error));
-		return false;
-	}
-
-	SRC_DATA *data = &state->data;
-	data->src_ratio = (double)dest_rate / (double)src_rate;
-	FormatDebug(libsamplerate_domain,
-		    "setting samplerate conversion ratio to %.2lf",
-		    data->src_ratio);
-	src_set_ratio(state->state, data->src_ratio);
-
-	return true;
-}
-
-static bool
-lsr_process(PcmResampler *state, Error &error)
-{
-	if (state->error == 0)
-		state->error = src_process(state->state, &state->data);
-	if (state->error) {
-		error.Format(libsamplerate_domain, state->error,
-			     "libsamplerate has failed: %s",
-			     src_strerror(state->error));
-		return false;
-	}
-
-	return true;
-}
-
-const float *
-pcm_resample_lsr_float(PcmResampler *state,
-		       unsigned channels,
-		       unsigned src_rate,
-		       const float *src_buffer, size_t src_size,
-		       unsigned dest_rate, size_t *dest_size_r,
-		       Error &error)
-{
-	SRC_DATA *data = &state->data;
-
-	assert((src_size % (sizeof(*src_buffer) * channels)) == 0);
-
-	if (!pcm_resample_set(state, channels, src_rate, dest_rate, error))
-		return nullptr;
-
-	data->input_frames = src_size / sizeof(*src_buffer) / channels;
-	data->data_in = const_cast<float *>(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 = (float *)state->out.Get(data_out_size);
-
-	if (!lsr_process(state, error))
-		return nullptr;
-
-	*dest_size_r = data->output_frames_gen *
-		sizeof(*data->data_out) * channels;
-	return data->data_out;
-}
-
-const int16_t *
-pcm_resample_lsr_16(PcmResampler *state,
-		    unsigned channels,
-		    unsigned src_rate,
-		    const int16_t *src_buffer, size_t src_size,
-		    unsigned dest_rate, size_t *dest_size_r,
-		    Error &error)
-{
-	SRC_DATA *data = &state->data;
-
-	assert((src_size % (sizeof(*src_buffer) * channels)) == 0);
-
-	if (!pcm_resample_set(state, channels, src_rate, dest_rate,
-			      error))
-		return nullptr;
-
-	data->input_frames = src_size / sizeof(*src_buffer) / channels;
-	size_t data_in_size = data->input_frames * sizeof(float) * channels;
-	data->data_in = (float *)state->in.Get(data_in_size);
-
-	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 = (float *)state->out.Get(data_out_size);
-
-	src_short_to_float_array(src_buffer, data->data_in,
-				 data->input_frames * channels);
-
-	if (!lsr_process(state, error))
-		return nullptr;
-
-	int16_t *dest_buffer;
-	*dest_size_r = data->output_frames_gen *
-		sizeof(*dest_buffer) * channels;
-	dest_buffer = (int16_t *)state->buffer.Get(*dest_size_r);
-	src_float_to_short_array(data->data_out, dest_buffer,
-				 data->output_frames_gen * channels);
-
-	return dest_buffer;
-}
-
-const int32_t *
-pcm_resample_lsr_32(PcmResampler *state,
-		    unsigned channels,
-		    unsigned src_rate,
-		    const int32_t *src_buffer, size_t src_size,
-		    unsigned dest_rate, size_t *dest_size_r,
-		    Error &error)
-{
-	SRC_DATA *data = &state->data;
-
-	assert((src_size % (sizeof(*src_buffer) * channels)) == 0);
-
-	if (!pcm_resample_set(state, channels, src_rate, dest_rate,
-			      error))
-		return nullptr;
-
-	data->input_frames = src_size / sizeof(*src_buffer) / channels;
-	size_t data_in_size = data->input_frames * sizeof(float) * channels;
-	data->data_in = (float *)state->in.Get(data_in_size);
-
-	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 = (float *)state->out.Get(data_out_size);
-
-	src_int_to_float_array(src_buffer, data->data_in,
-			       data->input_frames * channels);
-
-	if (!lsr_process(state, error))
-		return nullptr;
-
-	int32_t *dest_buffer;
-	*dest_size_r = data->output_frames_gen *
-		sizeof(*dest_buffer) * channels;
-	dest_buffer = (int32_t *)state->buffer.Get(*dest_size_r);
-	src_float_to_int_array(data->data_out, dest_buffer,
-			       data->output_frames_gen * channels);
-
-	return dest_buffer;
-}
-
-const int32_t *
-pcm_resample_lsr_24(PcmResampler *state,
-		    unsigned channels,
-		    unsigned src_rate,
-		    const int32_t *src_buffer, size_t src_size,
-		    unsigned dest_rate, size_t *dest_size_r,
-		    Error &error)
-{
-	const auto result = pcm_resample_lsr_32(state, channels,
-						src_rate, src_buffer, src_size,
-						dest_rate, dest_size_r,
-						error);
-	if (result != nullptr)
-		/* src_float_to_int_array() clamps for 32 bit
-		   integers; now make sure everything's fine for 24
-		   bit */
-		/* TODO: eliminate the 32 bit clamp to reduce overhead */
-		PcmClampN<int32_t, int32_t, 24>(const_cast<int32_t *>(result),
-						result,
-						*dest_size_r / sizeof(*result));
-
-	return result;
-}

src/pcm/Resampler.hxx

@@ -0,0 +1,74 @@
+/*
+ * 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_RESAMPLER_HXX
+#define MPD_PCM_RESAMPLER_HXX
+
+#include "util/ConstBuffer.hxx"
+#include "Compiler.h"
+
+struct AudioFormat;
+class Error;
+
+/**
+ * This is an interface for plugins that convert PCM data to a
+ * specific sample rate.
+ */
+class PcmResampler {
+public:
+	virtual ~PcmResampler() {}
+
+	/**
+	 * Opens the resampler, preparing it for Resample().
+	 *
+	 * @param af the audio format of incoming data; the plugin may
+	 * modify the object to enforce another input format (however,
+	 * it may not request a different input sample rate)
+	 * @param new_sample_rate the requested output sample rate
+	 * @param error location to store the error
+	 * @return the format of outgoing data or
+	 * AudioFormat::Undefined() on error
+	 */
+	virtual AudioFormat Open(AudioFormat &af, unsigned new_sample_rate,
+				 Error &error) = 0;
+
+	/**
+	 * Closes the resampler.  After that, you may call Open()
+	 * again.
+	 */
+	virtual void Close() = 0;
+
+	/**
+	 * Resamples a block of PCM data.
+	 *
+	 * @param src the input buffer
+	 * @param src_size the size of #src_buffer in bytes
+	 * @param dest_size_r the size of the returned buffer
+	 * @param error location to store the error occurring, or nullptr
+	 * to ignore errors.
+	 * @return the destination buffer on success (will be
+	 * invalidated by filter_close() or filter_filter()), nullptr on
+	 * error
+	 */
+	gcc_pure
+	virtual ConstBuffer<void> Resample(ConstBuffer<void> src,
+					   Error &error) = 0;
+};
+
+#endif