pcm_convert: convert to C++

10 files changed, 228 insertions, 224 deletions

diff --git a/Makefile.am b/Makefile.am
index 488b84a1..43e3c19a 100644
--- a/Makefile.am
+++ b/Makefile.am
@@ -331,7 +331,7 @@ libevent_a_SOURCES = \
 libpcm_a_SOURCES = \
 	src/pcm_buffer.c src/pcm_buffer.h \
 	src/pcm_export.c src/pcm_export.h \
-	src/pcm_convert.c src/pcm_convert.h \
+	src/PcmConvert.cxx src/PcmConvert.hxx \
 	src/dsd2pcm/dsd2pcm.c src/dsd2pcm/dsd2pcm.h \
 	src/pcm_dsd.c src/pcm_dsd.h \
 	src/pcm_dsd_usb.c src/pcm_dsd_usb.h \
diff --git a/src/DecoderAPI.cxx b/src/DecoderAPI.cxx
index b68af19b..d86b93fb 100644
--- a/src/DecoderAPI.cxx
+++ b/src/DecoderAPI.cxx
@@ -402,10 +402,11 @@ decoder_data(struct decoder *decoder,
 	}
 
 	if (!audio_format_equals(&dc->in_audio_format, &dc->out_audio_format)) {
-		data = pcm_convert(&decoder->conv_state,
-				   &dc->in_audio_format, data, length,
-				   &dc->out_audio_format, &length,
-				   &error);
+		data = decoder->conv_state.Convert(&dc->in_audio_format,
+						   data, length,
+						   &dc->out_audio_format,
+						   &length,
+						   &error);
 		if (data == NULL) {
 			/* the PCM conversion has failed - stop
 			   playback, since we have no better way to
diff --git a/src/DecoderInternal.cxx b/src/DecoderInternal.cxx
index 80f0adfd..e390fdfd 100644
--- a/src/DecoderInternal.cxx
+++ b/src/DecoderInternal.cxx
@@ -40,8 +40,6 @@ decoder::~decoder()
 
 	if (decoder_tag != nullptr)
 		tag_free(decoder_tag);
-
-	pcm_convert_deinit(&conv_state);
 }
 
 /**
diff --git a/src/DecoderInternal.hxx b/src/DecoderInternal.hxx
index ae50a62e..3423e3f9 100644
--- a/src/DecoderInternal.hxx
+++ b/src/DecoderInternal.hxx
@@ -21,7 +21,7 @@
 #define MPD_DECODER_INTERNAL_HXX
 
 #include "decoder_command.h"
-#include "pcm_convert.h"
+#include "PcmConvert.hxx"
 #include "replay_gain_info.h"
 
 struct input_stream;
@@ -29,7 +29,7 @@ struct input_stream;
 struct decoder {
 	struct decoder_control *dc;
 
-	struct pcm_convert_state conv_state;
+	PcmConvert conv_state;
 
 	/**
 	 * The time stamp of the next data chunk, in seconds.
@@ -91,7 +91,6 @@ struct decoder {
 		 song_tag(_tag), stream_tag(nullptr), decoder_tag(nullptr),
 		 chunk(nullptr),
 		 replay_gain_serial(0) {
-		pcm_convert_init(&conv_state);
 	}
 
 	~decoder();
diff --git a/src/pcm_convert.c b/src/PcmConvert.cxx
index 1d8ae957..175584c8 100644
--- a/src/pcm_convert.c
+++ b/src/PcmConvert.cxx
@@ -18,9 +18,13 @@
  */
 
 #include "config.h"
-#include "pcm_convert.h"
+#include "PcmConvert.hxx"
+
+extern "C" {
 #include "pcm_channels.h"
 #include "pcm_format.h"
+}
+
 #include "pcm_pack.h"
 #include "audio_format.h"
 
@@ -32,58 +36,58 @@
 #undef G_LOG_DOMAIN
 #define G_LOG_DOMAIN "pcm"
 
-void pcm_convert_init(struct pcm_convert_state *state)
+PcmConvert::PcmConvert()
 {
-	memset(state, 0, sizeof(*state));
+	memset(this, 0, sizeof(*this));
 
-	pcm_dsd_init(&state->dsd);
-	pcm_resample_init(&state->resample);
-	pcm_dither_24_init(&state->dither);
+	pcm_dsd_init(&dsd);
+	pcm_resample_init(&resample);
+	pcm_dither_24_init(&dither);
 
-	pcm_buffer_init(&state->format_buffer);
-	pcm_buffer_init(&state->channels_buffer);
+	pcm_buffer_init(&format_buffer);
+	pcm_buffer_init(&channels_buffer);
 }
 
-void pcm_convert_deinit(struct pcm_convert_state *state)
+PcmConvert::~PcmConvert()
 {
-	pcm_dsd_deinit(&state->dsd);
-	pcm_resample_deinit(&state->resample);
+	pcm_dsd_deinit(&dsd);
+	pcm_resample_deinit(&resample);
 
-	pcm_buffer_deinit(&state->format_buffer);
-	pcm_buffer_deinit(&state->channels_buffer);
+	pcm_buffer_deinit(&format_buffer);
+	pcm_buffer_deinit(&channels_buffer);
 }
 
 void
-pcm_convert_reset(struct pcm_convert_state *state)
+PcmConvert::Reset()
 {
-	pcm_dsd_reset(&state->dsd);
-	pcm_resample_reset(&state->resample);
+	pcm_dsd_reset(&dsd);
+	pcm_resample_reset(&resample);
 }
 
-static const int16_t *
-pcm_convert_16(struct pcm_convert_state *state,
-	       const struct audio_format *src_format,
-	       const void *src_buffer, size_t src_size,
-	       const struct audio_format *dest_format, size_t *dest_size_r,
-	       GError **error_r)
+inline const int16_t *
+PcmConvert::Convert16(const audio_format *src_format,
+		      const void *src_buffer, size_t src_size,
+		      const audio_format *dest_format, size_t *dest_size_r,
+		      GError **error_r)
 {
 	const int16_t *buf;
 	size_t len;
 
 	assert(dest_format->format == SAMPLE_FORMAT_S16);
 
-	buf = pcm_convert_to_16(&state->format_buffer, &state->dither,
-				src_format->format, src_buffer, src_size,
+	buf = pcm_convert_to_16(&format_buffer, &dither,
+				sample_format(src_format->format),
+				src_buffer, src_size,
 				&len);
 	if (buf == NULL) {
 		g_set_error(error_r, pcm_convert_quark(), 0,
 			    "Conversion from %s to 16 bit is not implemented",
-			    sample_format_to_string(src_format->format));
+			    sample_format_to_string(sample_format(src_format->format)));
 		return NULL;
 	}
 
 	if (src_format->channels != dest_format->channels) {
-		buf = pcm_convert_channels_16(&state->channels_buffer,
+		buf = pcm_convert_channels_16(&channels_buffer,
 					      dest_format->channels,
 					      src_format->channels,
 					      buf, len, &len);
@@ -98,7 +102,7 @@ pcm_convert_16(struct pcm_convert_state *state,
 	}
 
 	if (src_format->sample_rate != dest_format->sample_rate) {
-		buf = pcm_resample_16(&state->resample,
+		buf = pcm_resample_16(&resample,
 				      dest_format->channels,
 				      src_format->sample_rate, buf, len,
 				      dest_format->sample_rate, &len,
@@ -111,29 +115,29 @@ pcm_convert_16(struct pcm_convert_state *state,
 	return buf;
 }
 
-static const int32_t *
-pcm_convert_24(struct pcm_convert_state *state,
-	       const struct audio_format *src_format,
-	       const void *src_buffer, size_t src_size,
-	       const struct audio_format *dest_format, size_t *dest_size_r,
-	       GError **error_r)
+inline const int32_t *
+PcmConvert::Convert24(const audio_format *src_format,
+		      const void *src_buffer, size_t src_size,
+		      const audio_format *dest_format, size_t *dest_size_r,
+		      GError **error_r)
 {
 	const int32_t *buf;
 	size_t len;
 
 	assert(dest_format->format == SAMPLE_FORMAT_S24_P32);
 
-	buf = pcm_convert_to_24(&state->format_buffer, src_format->format,
+	buf = pcm_convert_to_24(&format_buffer,
+				sample_format(src_format->format),
 				src_buffer, src_size, &len);
 	if (buf == NULL) {
 		g_set_error(error_r, pcm_convert_quark(), 0,
 			    "Conversion from %s to 24 bit is not implemented",
-			    sample_format_to_string(src_format->format));
+			    sample_format_to_string(sample_format(src_format->format)));
 		return NULL;
 	}
 
 	if (src_format->channels != dest_format->channels) {
-		buf = pcm_convert_channels_24(&state->channels_buffer,
+		buf = pcm_convert_channels_24(&channels_buffer,
 					      dest_format->channels,
 					      src_format->channels,
 					      buf, len, &len);
@@ -148,7 +152,7 @@ pcm_convert_24(struct pcm_convert_state *state,
 	}
 
 	if (src_format->sample_rate != dest_format->sample_rate) {
-		buf = pcm_resample_24(&state->resample,
+		buf = pcm_resample_24(&resample,
 				      dest_format->channels,
 				      src_format->sample_rate, buf, len,
 				      dest_format->sample_rate, &len,
@@ -161,29 +165,29 @@ pcm_convert_24(struct pcm_convert_state *state,
 	return buf;
 }
 
-static const int32_t *
-pcm_convert_32(struct pcm_convert_state *state,
-	       const struct audio_format *src_format,
-	       const void *src_buffer, size_t src_size,
-	       const struct audio_format *dest_format, size_t *dest_size_r,
-	       GError **error_r)
+inline const int32_t *
+PcmConvert::Convert32(const audio_format *src_format,
+		      const void *src_buffer, size_t src_size,
+		      const audio_format *dest_format, size_t *dest_size_r,
+		      GError **error_r)
 {
 	const int32_t *buf;
 	size_t len;
 
 	assert(dest_format->format == SAMPLE_FORMAT_S32);
 
-	buf = pcm_convert_to_32(&state->format_buffer, src_format->format,
+	buf = pcm_convert_to_32(&format_buffer,
+				sample_format(src_format->format),
 				src_buffer, src_size, &len);
 	if (buf == NULL) {
 		g_set_error(error_r, pcm_convert_quark(), 0,
 			    "Conversion from %s to 32 bit is not implemented",
-			    sample_format_to_string(src_format->format));
+			    sample_format_to_string(sample_format(src_format->format)));
 		return NULL;
 	}
 
 	if (src_format->channels != dest_format->channels) {
-		buf = pcm_convert_channels_32(&state->channels_buffer,
+		buf = pcm_convert_channels_32(&channels_buffer,
 					      dest_format->channels,
 					      src_format->channels,
 					      buf, len, &len);
@@ -198,7 +202,7 @@ pcm_convert_32(struct pcm_convert_state *state,
 	}
 
 	if (src_format->sample_rate != dest_format->sample_rate) {
-		buf = pcm_resample_32(&state->resample,
+		buf = pcm_resample_32(&resample,
 				      dest_format->channels,
 				      src_format->sample_rate, buf, len,
 				      dest_format->sample_rate, &len,
@@ -211,34 +215,33 @@ pcm_convert_32(struct pcm_convert_state *state,
 	return buf;
 }
 
-static const float *
-pcm_convert_float(struct pcm_convert_state *state,
-		  const struct audio_format *src_format,
-		  const void *src_buffer, size_t src_size,
-		  const struct audio_format *dest_format, size_t *dest_size_r,
-		  GError **error_r)
+inline const float *
+PcmConvert::ConvertFloat(const audio_format *src_format,
+			 const void *src_buffer, size_t src_size,
+			 const audio_format *dest_format, size_t *dest_size_r,
+			 GError **error_r)
 {
-	const float *buffer = src_buffer;
+	const float *buffer = (const float *)src_buffer;
 	size_t size = src_size;
 
 	assert(dest_format->format == SAMPLE_FORMAT_FLOAT);
 
 	/* convert to float now */
 
-	buffer = pcm_convert_to_float(&state->format_buffer,
-				      src_format->format,
+	buffer = pcm_convert_to_float(&format_buffer,
+				      sample_format(src_format->format),
 				      buffer, size, &size);
 	if (buffer == NULL) {
 		g_set_error(error_r, pcm_convert_quark(), 0,
 			    "Conversion from %s to float is not implemented",
-			    sample_format_to_string(src_format->format));
+			    sample_format_to_string(sample_format(src_format->format)));
 		return NULL;
 	}
 
 	/* convert channels */
 
 	if (src_format->channels != dest_format->channels) {
-		buffer = pcm_convert_channels_float(&state->channels_buffer,
+		buffer = pcm_convert_channels_float(&channels_buffer,
 						    dest_format->channels,
 						    src_format->channels,
 						    buffer, size, &size);
@@ -256,7 +259,7 @@ pcm_convert_float(struct pcm_convert_state *state,
 	   libsamplerate */
 
 	if (src_format->sample_rate != dest_format->sample_rate) {
-		buffer = pcm_resample_float(&state->resample,
+		buffer = pcm_resample_float(&resample,
 					    dest_format->channels,
 					    src_format->sample_rate,
 					    buffer, size,
@@ -271,20 +274,19 @@ pcm_convert_float(struct pcm_convert_state *state,
 }
 
 const void *
-pcm_convert(struct pcm_convert_state *state,
-	    const struct audio_format *src_format,
-	    const void *src, size_t src_size,
-	    const struct audio_format *dest_format,
-	    size_t *dest_size_r,
-	    GError **error_r)
+PcmConvert::Convert(const audio_format *src_format,
+		    const void *src, size_t src_size,
+		    const audio_format *dest_format,
+		    size_t *dest_size_r,
+		    GError **error_r)
 {
 	struct audio_format float_format;
 	if (src_format->format == SAMPLE_FORMAT_DSD) {
 		size_t f_size;
-		const float *f = pcm_dsd_to_float(&state->dsd,
+		const float *f = pcm_dsd_to_float(&dsd,
 						  src_format->channels,
-						  false, src, src_size,
-						  &f_size);
+						  false, (const uint8_t *)src,
+						  src_size, &f_size);
 		if (f == NULL) {
 			g_set_error_literal(error_r, pcm_convert_quark(), 0,
 					    "DSD to PCM conversion failed");
@@ -299,35 +301,31 @@ pcm_convert(struct pcm_convert_state *state,
 		src_size = f_size;
 	}
 
-	switch (dest_format->format) {
+	switch (sample_format(dest_format->format)) {
 	case SAMPLE_FORMAT_S16:
-		return pcm_convert_16(state,
-				      src_format, src, src_size,
-				      dest_format, dest_size_r,
-				      error_r);
+		return Convert16(src_format, src, src_size,
+				 dest_format, dest_size_r,
+				 error_r);
 
 	case SAMPLE_FORMAT_S24_P32:
-		return pcm_convert_24(state,
-				      src_format, src, src_size,
-				      dest_format, dest_size_r,
-				      error_r);
+		return Convert24(src_format, src, src_size,
+				 dest_format, dest_size_r,
+				 error_r);
 
 	case SAMPLE_FORMAT_S32:
-		return pcm_convert_32(state,
-				      src_format, src, src_size,
-				      dest_format, dest_size_r,
-				      error_r);
+		return Convert32(src_format, src, src_size,
+				 dest_format, dest_size_r,
+				 error_r);
 
 	case SAMPLE_FORMAT_FLOAT:
-		return pcm_convert_float(state,
-					 src_format, src, src_size,
-					 dest_format, dest_size_r,
-					 error_r);
+		return ConvertFloat(src_format, src, src_size,
+				    dest_format, dest_size_r,
+				    error_r);
 
 	default:
 		g_set_error(error_r, pcm_convert_quark(), 0,
 			    "PCM conversion to %s is not implemented",
-			    sample_format_to_string(dest_format->format));
+			    sample_format_to_string(sample_format(dest_format->format)));
 		return NULL;
 	}
 }
diff --git a/src/PcmConvert.hxx b/src/PcmConvert.hxx
new file mode 100644
index 00000000..bec30af4
--- /dev/null
+++ b/src/PcmConvert.hxx
@@ -0,0 +1,114 @@
+/*
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#ifndef PCM_CONVERT_HXX
+#define PCM_CONVERT_HXX
+
+extern "C" {
+#include "pcm_dsd.h"
+#include "pcm_resample.h"
+#include "pcm_dither.h"
+#include "pcm_buffer.h"
+}
+
+#include <glib.h>
+
+struct audio_format;
+
+/**
+ * This object is statically allocated (within another struct), and
+ * holds buffer allocations and the state for all kinds of PCM
+ * conversions.
+ */
+class PcmConvert {
+	struct pcm_dsd dsd;
+
+	struct pcm_resample_state resample;
+
+	struct pcm_dither dither;
+
+	/** the buffer for converting the sample format */
+	struct pcm_buffer format_buffer;
+
+	/** the buffer for converting the channel count */
+	struct pcm_buffer channels_buffer;
+
+public:
+	PcmConvert();
+	~PcmConvert();
+
+
+	/**
+	 * Reset the pcm_convert_state object.  Use this at the
+	 * boundary between two distinct songs and each time the
+	 * format changes.
+	 */
+	void Reset();
+
+	/**
+	 * Converts PCM data between two audio formats.
+	 *
+	 * @param src_format the source audio format
+	 * @param src the source PCM buffer
+	 * @param src_size the size of #src in bytes
+	 * @param dest_format the requested destination audio format
+	 * @param dest_size_r returns the number of bytes of the destination buffer
+	 * @param error_r location to store the error occurring, or NULL to
+	 * ignore errors
+	 * @return the destination buffer, or NULL on error
+	 */
+	const void *Convert(const audio_format *src_format,
+			    const void *src, size_t src_size,
+			    const audio_format *dest_format,
+			    size_t *dest_size_r,
+			    GError **error_r);
+
+private:
+	const int16_t *Convert16(const audio_format *src_format,
+				 const void *src_buffer, size_t src_size,
+				 const audio_format *dest_format,
+				 size_t *dest_size_r,
+				 GError **error_r);
+
+	const int32_t *Convert24(const audio_format *src_format,
+				 const void *src_buffer, size_t src_size,
+				 const audio_format *dest_format,
+				 size_t *dest_size_r,
+				 GError **error_r);
+
+	const int32_t *Convert32(const audio_format *src_format,
+				 const void *src_buffer, size_t src_size,
+				 const audio_format *dest_format,
+				 size_t *dest_size_r,
+				 GError **error_r);
+
+	const float *ConvertFloat(const audio_format *src_format,
+				  const void *src_buffer, size_t src_size,
+				  const audio_format *dest_format,
+				  size_t *dest_size_r,
+				  GError **error_r);
+};
+
+static inline GQuark
+pcm_convert_quark(void)
+{
+	return g_quark_from_static_string("pcm_convert");
+}
+
+#endif
diff --git a/src/filter/ConvertFilterPlugin.cxx b/src/filter/ConvertFilterPlugin.cxx
index 04f34842..23c912b0 100644
--- a/src/filter/ConvertFilterPlugin.cxx
+++ b/src/filter/ConvertFilterPlugin.cxx
@@ -23,7 +23,8 @@
 #include "filter_internal.h"
 #include "filter_registry.h"
 #include "conf.h"
-#include "pcm_convert.h"
+#include "PcmConvert.hxx"
+#include "util/Manual.hxx"
 #include "audio_format.h"
 #include "poison.h"
 
@@ -51,7 +52,7 @@ struct ConvertFilter {
 	 */
 	struct audio_format out_audio_format;
 
-	struct pcm_convert_state state;
+	Manual<PcmConvert> state;
 
 	ConvertFilter() {
 		filter_init(&base, &convert_filter_plugin);
@@ -81,7 +82,7 @@ convert_filter_open(struct filter *_filter, struct audio_format *audio_format,
 	assert(audio_format_valid(audio_format));
 
 	filter->in_audio_format = filter->out_audio_format = *audio_format;
-	pcm_convert_init(&filter->state);
+	filter->state.Construct();
 
 	return &filter->in_audio_format;
 }
@@ -91,7 +92,7 @@ convert_filter_close(struct filter *_filter)
 {
 	ConvertFilter *filter = (ConvertFilter *)_filter;
 
-	pcm_convert_deinit(&filter->state);
+	filter->state.Destruct();
 
 	poison_undefined(&filter->in_audio_format,
 			 sizeof(filter->in_audio_format));
@@ -113,10 +114,10 @@ convert_filter_filter(struct filter *_filter, const void *src, size_t src_size,
 		return src;
 	}
 
-	dest = pcm_convert(&filter->state, &filter->in_audio_format,
-			   src, src_size,
-			   &filter->out_audio_format, dest_size_r,
-			   error_r);
+	dest = filter->state->Convert(&filter->in_audio_format,
+				      src, src_size,
+				      &filter->out_audio_format, dest_size_r,
+				      error_r);
 	if (dest == NULL)
 		return NULL;
 
diff --git a/src/pcm_convert.h b/src/pcm_convert.h
deleted file mode 100644
index 0668a2b6..00000000
--- a/src/pcm_convert.h
+++ /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.
- */
-
-#ifndef PCM_CONVERT_H
-#define PCM_CONVERT_H
-
-#include "pcm_dsd.h"
-#include "pcm_resample.h"
-#include "pcm_dither.h"
-#include "pcm_buffer.h"
-
-#include <glib.h>
-
-struct audio_format;
-
-/**
- * This object is statically allocated (within another struct), and
- * holds buffer allocations and the state for all kinds of PCM
- * conversions.
- */
-struct pcm_convert_state {
-	struct pcm_dsd dsd;
-
-	struct pcm_resample_state resample;
-
-	struct pcm_dither dither;
-
-	/** the buffer for converting the sample format */
-	struct pcm_buffer format_buffer;
-
-	/** the buffer for converting the channel count */
-	struct pcm_buffer channels_buffer;
-};
-
-static inline GQuark
-pcm_convert_quark(void)
-{
-	return g_quark_from_static_string("pcm_convert");
-}
-
-G_BEGIN_DECLS
-
-/**
- * Initializes a pcm_convert_state object.
- */
-void pcm_convert_init(struct pcm_convert_state *state);
-
-/**
- * Deinitializes a pcm_convert_state object and frees allocated
- * memory.
- */
-void pcm_convert_deinit(struct pcm_convert_state *state);
-
-/**
- * Reset the pcm_convert_state object.  Use this at the boundary
- * between two distinct songs and each time the format changes.
- */
-void
-pcm_convert_reset(struct pcm_convert_state *state);
-
-/**
- * Converts PCM data between two audio formats.
- *
- * @param state an initialized pcm_convert_state object
- * @param src_format the source audio format
- * @param src the source PCM buffer
- * @param src_size the size of #src in bytes
- * @param dest_format the requested destination audio format
- * @param dest_size_r returns the number of bytes of the destination buffer
- * @param error_r location to store the error occurring, or NULL to
- * ignore errors
- * @return the destination buffer, or NULL on error
- */
-const void *
-pcm_convert(struct pcm_convert_state *state,
-	    const struct audio_format *src_format,
-	    const void *src, size_t src_size,
-	    const struct audio_format *dest_format,
-	    size_t *dest_size_r,
-	    GError **error_r);
-
-G_END_DECLS
-
-#endif
diff --git a/test/run_convert.cxx b/test/run_convert.cxx
index 1e54f2d3..e66d0be2 100644
--- a/test/run_convert.cxx
+++ b/test/run_convert.cxx
@@ -26,7 +26,7 @@
 #include "config.h"
 #include "AudioParser.hxx"
 #include "audio_format.h"
-#include "pcm_convert.h"
+#include "PcmConvert.hxx"
 #include "conf.h"
 #include "util/fifo_buffer.h"
 #include "stdbin.h"
@@ -58,7 +58,6 @@ int main(int argc, char **argv)
 {
 	GError *error = NULL;
 	struct audio_format in_audio_format, out_audio_format;
-	struct pcm_convert_state state;
 	const void *output;
 	ssize_t nbytes;
 	size_t length;
@@ -90,7 +89,7 @@ int main(int argc, char **argv)
 
 	const size_t in_frame_size = audio_format_frame_size(&in_audio_format);
 
-	pcm_convert_init(&state);
+	PcmConvert state;
 
 	struct fifo_buffer *buffer = fifo_buffer_new(4096);
 
@@ -113,8 +112,8 @@ int main(int argc, char **argv)
 
 		fifo_buffer_consume(buffer, length);
 
-		output = pcm_convert(&state, &in_audio_format, src, length,
-				     &out_audio_format, &length, &error);
+		output = state.Convert(&in_audio_format, src, length,
+				       &out_audio_format, &length, &error);
 		if (output == NULL) {
 			g_printerr("Failed to convert: %s\n", error->message);
 			return 2;
@@ -123,5 +122,5 @@ int main(int argc, char **argv)
 		G_GNUC_UNUSED ssize_t ignored = write(1, output, length);
 	}
 
-	pcm_convert_deinit(&state);
+	return EXIT_SUCCESS;
 }
diff --git a/test/run_output.cxx b/test/run_output.cxx
index e9daf3c1..f76e6fc7 100644
--- a/test/run_output.cxx
+++ b/test/run_output.cxx
@@ -27,12 +27,12 @@
 #include "IOThread.hxx"
 #include "fs/Path.hxx"
 #include "AudioParser.hxx"
+#include "PcmConvert.hxx"
 
 extern "C" {
 #include "output_plugin.h"
 #include "output_internal.h"
 #include "filter_registry.h"
-#include "pcm_convert.h"
 }
 
 #include "PlayerControl.hxx"
@@ -52,21 +52,15 @@ GlobalEvents::Emit(gcc_unused Event event)
 {
 }
 
-void pcm_convert_init(G_GNUC_UNUSED struct pcm_convert_state *state)
-{
-}
-
-void pcm_convert_deinit(G_GNUC_UNUSED struct pcm_convert_state *state)
-{
-}
+PcmConvert::PcmConvert() {}
+PcmConvert::~PcmConvert() {}
 
 const void *
-pcm_convert(G_GNUC_UNUSED struct pcm_convert_state *state,
-	    G_GNUC_UNUSED const struct audio_format *src_format,
-	    G_GNUC_UNUSED const void *src, G_GNUC_UNUSED size_t src_size,
-	    G_GNUC_UNUSED const struct audio_format *dest_format,
-	    G_GNUC_UNUSED size_t *dest_size_r,
-	    GError **error_r)
+PcmConvert::Convert(gcc_unused const audio_format *src_format,
+		    gcc_unused const void *src, gcc_unused size_t src_size,
+		    gcc_unused const audio_format *dest_format,
+		    gcc_unused size_t *dest_size_r,
+		    gcc_unused GError **error_r)
 {
 	g_set_error(error_r, pcm_convert_quark(), 0,
 		    "Not implemented");