pcm/PcmConvert: move code to new class GluePcmResampler

4 files changed, 152 insertions, 139 deletions

diff --git a/src/pcm/GlueResampler.cxx b/src/pcm/GlueResampler.cxx
new file mode 100644
index 000000000..8aee98f38
--- /dev/null
+++ b/src/pcm/GlueResampler.cxx
@@ -0,0 +1,94 @@
+/*
+ * 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 "GlueResampler.hxx"
+#include "PcmConvert.hxx"
+#include "PcmFormat.hxx"
+#include "util/ConstBuffer.hxx"
+#include "util/Error.hxx"
+
+bool
+GluePcmResampler::Open(AudioFormat _src_format, unsigned _new_sample_rate,
+		       gcc_unused Error &error)
+{
+	src_format = _src_format;
+	new_sample_rate = _new_sample_rate;
+
+	return true;
+}
+
+void
+GluePcmResampler::Close()
+{
+	resampler.Reset();
+}
+
+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;
+
+	default:
+		error.Format(pcm_convert_domain,
+			     "Resampling %s is not implemented",
+			     sample_format_to_string(src_format.format));
+		return nullptr;
+	}
+
+	return { result, size };
+}
diff --git a/src/pcm/GlueResampler.hxx b/src/pcm/GlueResampler.hxx
new file mode 100644
index 000000000..a7e0a84f2
--- /dev/null
+++ b/src/pcm/GlueResampler.hxx
@@ -0,0 +1,44 @@
+/*
+ * Copyright (C) 2003-2013 The Music Player Daemon Project
+ * http://www.musicpd.org
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with this program; if not, write to the Free Software Foundation, Inc.,
+ * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ */
+
+#ifndef MPD_GLUE_RESAMPLER_HXX
+#define MPD_GLUE_RESAMPLER_HXX
+
+#include "check.h"
+#include "AudioFormat.hxx"
+#include "PcmResample.hxx"
+
+class Error;
+template<typename T> struct ConstBuffer;
+
+class GluePcmResampler {
+	PcmResampler resampler;
+
+	AudioFormat src_format;
+	unsigned new_sample_rate;
+
+public:
+	bool Open(AudioFormat src_format, unsigned new_sample_rate,
+		  Error &error);
+	void Close();
+
+	ConstBuffer<void> Resample(ConstBuffer<void> src, Error &error);
+};
+
+#endif
diff --git a/src/pcm/PcmConvert.cxx b/src/pcm/PcmConvert.cxx
index bc289460d..0552962aa 100644
--- a/src/pcm/PcmConvert.cxx
+++ b/src/pcm/PcmConvert.cxx
@@ -78,6 +78,10 @@ PcmConvert::Open(AudioFormat _src_format, AudioFormat _dest_format,
 		return false;
 	}
 
+	if (format.sample_rate != dest_format.sample_rate &&
+	    !resampler.Open(format, dest_format.sample_rate, error))
+		return false;
+
 	return true;
 }
 
@@ -91,7 +95,9 @@ PcmConvert::Close()
 		format_converter.Close();
 
 	dsd.Reset();
-	resampler.Reset();
+
+	if (src_format.sample_rate != dest_format.sample_rate)
+		resampler.Close();
 
 #ifndef NDEBUG
 	src_format.Clear();
@@ -99,102 +105,6 @@ PcmConvert::Close()
 #endif
 }
 
-inline ConstBuffer<int16_t>
-PcmConvert::Convert16(ConstBuffer<int16_t> src, AudioFormat format,
-		      Error &error)
-{
-	assert(format.format == SampleFormat::S16);
-	assert(dest_format.format == SampleFormat::S16);
-	assert(format.channels == dest_format.channels);
-
-	auto buf = src.data;
-	size_t len = src.size * sizeof(*src.data);
-
-	if (format.sample_rate != dest_format.sample_rate) {
-		buf = resampler.Resample16(dest_format.channels,
-					   format.sample_rate, buf, len,
-					   dest_format.sample_rate, &len,
-					   error);
-		if (buf == nullptr)
-			return nullptr;
-	}
-
-	return ConstBuffer<int16_t>::FromVoid({buf, len});
-}
-
-inline ConstBuffer<int32_t>
-PcmConvert::Convert24(ConstBuffer<int32_t> src, AudioFormat format,
-		      Error &error)
-{
-	assert(format.format == SampleFormat::S24_P32);
-	assert(dest_format.format == SampleFormat::S24_P32);
-	assert(format.channels == dest_format.channels);
-
-	auto buf = src.data;
-	size_t len = src.size * sizeof(*src.data);
-
-	if (format.sample_rate != dest_format.sample_rate) {
-		buf = resampler.Resample24(dest_format.channels,
-					   format.sample_rate, buf, len,
-					   dest_format.sample_rate, &len,
-					   error);
-		if (buf == nullptr)
-			return nullptr;
-	}
-
-	return ConstBuffer<int32_t>::FromVoid({buf, len});
-}
-
-inline ConstBuffer<int32_t>
-PcmConvert::Convert32(ConstBuffer<int32_t> src, AudioFormat format,
-		      Error &error)
-{
-	assert(format.format == SampleFormat::S32);
-	assert(dest_format.format == SampleFormat::S32);
-	assert(format.channels == dest_format.channels);
-
-	auto buf = src.data;
-	size_t len = src.size * sizeof(*src.data);
-
-	if (format.sample_rate != dest_format.sample_rate) {
-		buf = resampler.Resample32(dest_format.channels,
-					   format.sample_rate, buf, len,
-					   dest_format.sample_rate, &len,
-					   error);
-		if (buf == nullptr)
-			return nullptr;
-	}
-
-	return ConstBuffer<int32_t>::FromVoid({buf, len});
-}
-
-inline ConstBuffer<float>
-PcmConvert::ConvertFloat(ConstBuffer<float> src, AudioFormat format,
-			 Error &error)
-{
-	assert(format.format == SampleFormat::FLOAT);
-	assert(dest_format.format == SampleFormat::FLOAT);
-	assert(format.channels == dest_format.channels);
-
-	auto buffer = src.data;
-	size_t size = src.size * sizeof(*src.data);
-
-	/* resample with float, because this is the best format for
-	   libsamplerate */
-
-	if (format.sample_rate != dest_format.sample_rate) {
-		buffer = resampler.ResampleFloat(dest_format.channels,
-						 format.sample_rate,
-						 buffer, size,
-						 dest_format.sample_rate,
-						 &size, error);
-		if (buffer == nullptr)
-			return nullptr;
-	}
-
-	return ConstBuffer<float>::FromVoid({buffer, size});
-}
-
 const void *
 PcmConvert::Convert(const void *src, size_t src_size,
 		    size_t *dest_size_r,
@@ -233,32 +143,12 @@ PcmConvert::Convert(const void *src, size_t src_size,
 		format.channels = dest_format.channels;
 	}
 
-	switch (dest_format.format) {
-	case SampleFormat::S16:
-		buffer = Convert16(ConstBuffer<int16_t>::FromVoid(buffer),
-				   format, error).ToVoid();
-		break;
-
-	case SampleFormat::S24_P32:
-		buffer = Convert24(ConstBuffer<int32_t>::FromVoid(buffer),
-				   format, error).ToVoid();
-		break;
-
-	case SampleFormat::S32:
-		buffer = Convert32(ConstBuffer<int32_t>::FromVoid(buffer),
-				   format, error).ToVoid();
-		break;
-
-	case SampleFormat::FLOAT:
-		buffer = ConvertFloat(ConstBuffer<float>::FromVoid(buffer),
-				      format, error).ToVoid();
-		break;
+	if (format.sample_rate != dest_format.sample_rate) {
+		buffer = resampler.Resample(buffer, error);
+		if (buffer.IsNull())
+			return nullptr;
 
-	default:
-		error.Format(pcm_convert_domain,
-			     "PCM conversion to %s is not implemented",
-			     sample_format_to_string(dest_format.format));
-		return nullptr;
+		format.sample_rate = dest_format.sample_rate;
 	}
 
 	*dest_size_r = buffer.size;
diff --git a/src/pcm/PcmConvert.hxx b/src/pcm/PcmConvert.hxx
index d6e113915..9835045d6 100644
--- a/src/pcm/PcmConvert.hxx
+++ b/src/pcm/PcmConvert.hxx
@@ -21,10 +21,10 @@
 #define PCM_CONVERT_HXX
 
 #include "PcmDsd.hxx"
-#include "PcmResample.hxx"
 #include "PcmBuffer.hxx"
 #include "FormatConverter.hxx"
 #include "ChannelsConverter.hxx"
+#include "GlueResampler.hxx"
 #include "AudioFormat.hxx"
 
 #include <stddef.h>
@@ -41,10 +41,9 @@ class Domain;
 class PcmConvert {
 	PcmDsd dsd;
 
-	PcmResampler resampler;
-
 	PcmFormatConverter format_converter;
 	PcmChannelsConverter channels_converter;
+	GluePcmResampler resampler;
 
 	AudioFormat src_format, dest_format;
 
@@ -79,20 +78,6 @@ public:
 	const void *Convert(const void *src, size_t src_size,
 			    size_t *dest_size_r,
 			    Error &error);
-
-private:
-	ConstBuffer<int16_t> Convert16(ConstBuffer<int16_t> src,
-				       AudioFormat format,
-				       Error &error);
-	ConstBuffer<int32_t> Convert24(ConstBuffer<int32_t> src,
-				       AudioFormat format,
-				       Error &error);
-	ConstBuffer<int32_t> Convert32(ConstBuffer<int32_t> src,
-				       AudioFormat format,
-				       Error &error);
-	ConstBuffer<float> ConvertFloat(ConstBuffer<float> src,
-					AudioFormat format,
-					Error &error);
 };
 
 extern const Domain pcm_convert_domain;