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.

18 files changed, 699 insertions, 890 deletions

diff --git a/Makefile.am b/Makefile.am
index 864f0dbbb..2f3cd4531 100644
--- a/Makefile.am
+++ b/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/PcmResampleFallback.cxx \
-	src/pcm/PcmResampleInternal.hxx \
+	src/pcm/FallbackResampler.cxx src/pcm/FallbackResampler.hxx \
+	src/pcm/ConfiguredResampler.cxx src/pcm/ConfiguredResampler.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
diff --git a/src/pcm/ConfiguredResampler.cxx b/src/pcm/ConfiguredResampler.cxx
new file mode 100644
index 000000000..92f3d0903
--- /dev/null
+++ b/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();
+}
diff --git a/src/pcm/ConfiguredResampler.hxx b/src/pcm/ConfiguredResampler.hxx
new file mode 100644
index 000000000..6d12ee9c6
--- /dev/null
+++ b/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
diff --git a/src/pcm/FallbackResampler.cxx b/src/pcm/FallbackResampler.cxx
new file mode 100644
index 000000000..a3b6b78ee
--- /dev/null
+++ b/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();
+}
diff --git a/src/pcm/FallbackResampler.hxx b/src/pcm/FallbackResampler.hxx
new file mode 100644
index 000000000..1b8d0377d
--- /dev/null
+++ b/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
diff --git a/src/pcm/GlueResampler.cxx b/src/pcm/GlueResampler.cxx
index 8aee98f38..ef80e08a5 100644
--- a/src/pcm/GlueResampler.cxx
+++ b/src/pcm/GlueResampler.cxx
@@ -19,76 +19,67 @@
 
 #include "config.h"
 #include "GlueResampler.hxx"
-#include "PcmConvert.hxx"
-#include "PcmFormat.hxx"
-#include "util/ConstBuffer.hxx"
-#include "util/Error.hxx"
+#include "ConfiguredResampler.hxx"
+#include "Resampler.hxx"
+
+#include <assert.h>
+
+GluePcmResampler::GluePcmResampler()
+	:resampler(pcm_resampler_create()) {}
+
+GluePcmResampler::~GluePcmResampler()
+{
+	delete resampler;
+}
 
 bool
-GluePcmResampler::Open(AudioFormat _src_format, unsigned _new_sample_rate,
-		       gcc_unused Error &error)
+GluePcmResampler::Open(AudioFormat src_format, unsigned new_sample_rate,
+		       Error &error)
 {
-	src_format = _src_format;
-	new_sample_rate = _new_sample_rate;
+	assert(src_format.IsValid());
+	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;
-	size_t size;
-
-	switch (src_format.format) {
-	case SampleFormat::S16:
-		result = resampler.Resample16(src_format.channels,
-					      src_format.sample_rate,
-					      (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;
+	assert(!src.IsNull());
 
-	default:
-		error.Format(pcm_convert_domain,
-			     "Resampling %s is not implemented",
-			     sample_format_to_string(src_format.format));
-		return nullptr;
+	if (requested_sample_format != src_sample_format) {
+		src = format_converter.Convert(src, error);
+		if (src.IsNull())
+			return nullptr;
 	}
 
-	return { result, size };
+	return resampler->Resample(src, error);
 }
diff --git a/src/pcm/GlueResampler.hxx b/src/pcm/GlueResampler.hxx
index a7e0a84f2..7bd923bab 100644
--- a/src/pcm/GlueResampler.hxx
+++ b/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;
+
+	SampleFormat src_sample_format, requested_sample_format;
+	SampleFormat output_sample_format;
 
-	AudioFormat src_format;
-	unsigned new_sample_rate;
+	/**
+	 * 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);
 };
 
diff --git a/src/pcm/LibsamplerateResampler.cxx b/src/pcm/LibsamplerateResampler.cxx
new file mode 100644
index 000000000..586391e67
--- /dev/null
+++ b/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();
+}
diff --git a/src/pcm/LibsamplerateResampler.hxx b/src/pcm/LibsamplerateResampler.hxx
new file mode 100644
index 000000000..0c1f613c8
--- /dev/null
+++ b/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
diff --git a/src/pcm/PcmConvert.cxx b/src/pcm/PcmConvert.cxx
index 0552962aa..5501d8ddf 100644
--- a/src/pcm/PcmConvert.cxx
+++ b/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()
@@ -66,39 +67,51 @@ PcmConvert::Open(AudioFormat _src_format, AudioFormat _dest_format,
 	if (format.format == SampleFormat::DSD)
 		format.format = SampleFormat::FLOAT;
 
-	if (format.format != dest_format.format &&
-	    !format_converter.Open(format.format, dest_format.format, error))
+	enable_resampler = format.sample_rate != dest_format.sample_rate;
+	if (enable_resampler) {
+		if (!resampler.Open(format, dest_format.sample_rate, error))
+			return false;
+
+		format.format = resampler.GetOutputSampleFormat();
+		format.sample_rate = dest_format.sample_rate;
+	}
+
+	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;
+	}
+
 	format.format = dest_format.format;
 
-	if (format.channels != dest_format.channels &&
+	enable_channels = format.channels != dest_format.channels;
+	if (enable_channels &&
 	    !channels_converter.Open(format.format, format.channels,
 				     dest_format.channels, error)) {
-		format_converter.Close();
+		if (enable_format)
+			format_converter.Close();
+		if (enable_resampler)
+			resampler.Close();
 		return false;
 	}
 
-	if (format.sample_rate != dest_format.sample_rate &&
-	    !resampler.Open(format, dest_format.sample_rate, error))
-		return false;
-
 	return true;
 }
 
 void
 PcmConvert::Close()
 {
-	if (src_format.channels != dest_format.channels)
+	if (enable_channels)
 		channels_converter.Close();
-
-	if (src_format.format != dest_format.format)
+	if (enable_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();
@@ -127,28 +140,29 @@ PcmConvert::Convert(const void *src, size_t src_size,
 		format.format = SampleFormat::FLOAT;
 	}
 
-	if (format.format != dest_format.format) {
-		buffer = format_converter.Convert(buffer, error);
+	if (enable_resampler) {
+		buffer = resampler.Resample(buffer, error);
 		if (buffer.IsNull())
 			return nullptr;
 
-		format.format = dest_format.format;
+		format.format = resampler.GetOutputSampleFormat();
+		format.sample_rate = dest_format.sample_rate;
 	}
 
-	if (format.channels != dest_format.channels) {
-		buffer = channels_converter.Convert(buffer, error);
+	if (enable_format) {
+		buffer = format_converter.Convert(buffer, error);
 		if (buffer.IsNull())
 			return nullptr;
 
-		format.channels = dest_format.channels;
+		format.format = dest_format.format;
 	}
 
-	if (format.sample_rate != dest_format.sample_rate) {
-		buffer = resampler.Resample(buffer, error);
+	if (enable_channels) {
+		buffer = channels_converter.Convert(buffer, error);
 		if (buffer.IsNull())
 			return nullptr;
 
-		format.sample_rate = dest_format.sample_rate;
+		format.channels = dest_format.channels;
 	}
 
 	*dest_size_r = buffer.size;
diff --git a/src/pcm/PcmConvert.hxx b/src/pcm/PcmConvert.hxx
index 9835045d6..13dff4427 100644
--- a/src/pcm/PcmConvert.hxx
+++ b/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();
diff --git a/src/pcm/PcmFormat.cxx b/src/pcm/PcmFormat.cxx
index 4565c71c6..b7a496264 100644
--- a/src/pcm/PcmFormat.cxx
+++ b/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>>
diff --git a/src/pcm/PcmResample.cxx b/src/pcm/PcmResample.cxx
deleted file mode 100644
index b30e01407..000000000
--- a/src/pcm/PcmResample.cxx
+++ /dev/null
@@ -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);
-}
diff --git a/src/pcm/PcmResample.hxx b/src/pcm/PcmResample.hxx
deleted file mode 100644
index e839d6ecd..000000000
--- a/src/pcm/PcmResample.hxx
+++ /dev/null
@@ -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
diff --git a/src/pcm/PcmResampleFallback.cxx b/src/pcm/PcmResampleFallback.cxx
deleted file mode 100644
index ca92e5a83..000000000
--- a/src/pcm/PcmResampleFallback.cxx
+++ /dev/null
@@ -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;
-}
diff --git a/src/pcm/PcmResampleInternal.hxx b/src/pcm/PcmResampleInternal.hxx
deleted file mode 100644
index 94cef94ff..000000000
--- a/src/pcm/PcmResampleInternal.hxx
+++ /dev/null
@@ -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
diff --git a/src/pcm/PcmResampleLibsamplerate.cxx b/src/pcm/PcmResampleLibsamplerate.cxx
deleted file mode 100644
index e61ff2edf..000000000
--- a/src/pcm/PcmResampleLibsamplerate.cxx
+++ /dev/null
@@ -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;
-}
diff --git a/src/pcm/Resampler.hxx b/src/pcm/Resampler.hxx
new file mode 100644
index 000000000..a74ef4e77
--- /dev/null
+++ b/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