Huge new year audio commit!

* Refactored the whole audaspace library to use float as sample format over all readers.
* Added new Readers like the linear resampler, envelope, lowpass, highpass and butterworth.
* Note: The butterworth filter isn't working correctly, some bug in there... Maybe also true for the envelope.
* Added a sound to f-curve operator that behaves mostly like the soundtracker script of technoestupido.

intern/audaspace/FX/AUD_ButterworthFactory.cpp

@@ -0,0 +1,49 @@
+/*
+ * $Id$
+ *
+ * ***** BEGIN LGPL LICENSE BLOCK *****
+ *
+ * Copyright 2009 Jörg Hermann Müller
+ *
+ * This file is part of AudaSpace.
+ *
+ * AudaSpace is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU Lesser General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * AudaSpace 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 Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public License
+ * along with AudaSpace.  If not, see <http://www.gnu.org/licenses/>.
+ *
+ * ***** END LGPL LICENSE BLOCK *****
+ */
+
+#include "AUD_ButterworthFactory.h"
+#include "AUD_ButterworthReader.h"
+
+AUD_ButterworthFactory::AUD_ButterworthFactory(AUD_IFactory* factory,
+											   float frequency) :
+		AUD_EffectFactory(factory),
+		m_frequency(frequency) {}
+
+AUD_ButterworthFactory::AUD_ButterworthFactory(float frequency) :
+		AUD_EffectFactory(0),
+		m_frequency(frequency) {}
+
+AUD_IReader* AUD_ButterworthFactory::createReader()
+{
+	AUD_IReader* reader = getReader();
+
+	if(reader != 0)
+	{
+		reader = new AUD_ButterworthReader(reader, m_frequency);
+		AUD_NEW("reader")
+	}
+
+	return reader;
+}

intern/audaspace/FX/AUD_ButterworthFactory.h

@@ -0,0 +1,59 @@
+/*
+ * $Id$
+ *
+ * ***** BEGIN LGPL LICENSE BLOCK *****
+ *
+ * Copyright 2009 Jörg Hermann Müller
+ *
+ * This file is part of AudaSpace.
+ *
+ * AudaSpace is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU Lesser General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * AudaSpace 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 Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public License
+ * along with AudaSpace.  If not, see <http://www.gnu.org/licenses/>.
+ *
+ * ***** END LGPL LICENSE BLOCK *****
+ */
+
+#ifndef AUD_BUTTERWORTHFACTORY
+#define AUD_BUTTERWORTHFACTORY
+
+#include "AUD_EffectFactory.h"
+
+/**
+ * This factory creates a butterworth filter reader.
+ */
+class AUD_ButterworthFactory : public AUD_EffectFactory
+{
+private:
+	/**
+	 * The attack value in seconds.
+	 */
+	float m_frequency;
+
+public:
+	/**
+	 * Creates a new butterworth factory.
+	 * \param factory The input factory.
+	 * \param frequency The cutoff frequency.
+	 */
+	AUD_ButterworthFactory(AUD_IFactory* factory, float frequency);
+
+	/**
+	 * Creates a new butterworth factory.
+	 * \param frequency The cutoff frequency.
+	 */
+	AUD_ButterworthFactory(float frequency);
+
+	virtual AUD_IReader* createReader();
+};
+
+#endif //AUD_BUTTERWORTHFACTORY

intern/audaspace/FX/AUD_ButterworthReader.cpp

@@ -0,0 +1,120 @@
+/*
+ * $Id$
+ *
+ * ***** BEGIN LGPL LICENSE BLOCK *****
+ *
+ * Copyright 2009 Jörg Hermann Müller
+ *
+ * This file is part of AudaSpace.
+ *
+ * AudaSpace is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU Lesser General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * AudaSpace 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 Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public License
+ * along with AudaSpace.  If not, see <http://www.gnu.org/licenses/>.
+ *
+ * ***** END LGPL LICENSE BLOCK *****
+ */
+
+#include "AUD_ButterworthReader.h"
+#include "AUD_Buffer.h"
+
+#include <cstring>
+#include <cmath>
+
+#define BWPB41 0.76536686473
+#define BWPB42 1.84775906502
+#define CC channels + channel
+
+AUD_ButterworthReader::AUD_ButterworthReader(AUD_IReader* reader,
+											 float frequency) :
+		AUD_EffectReader(reader)
+{
+	AUD_Specs specs = reader->getSpecs();
+	int samplesize = AUD_SAMPLE_SIZE(specs);
+
+	m_buffer = new AUD_Buffer(); AUD_NEW("buffer")
+
+	m_outvalues = new AUD_Buffer(samplesize * 5); AUD_NEW("buffer")
+	memset(m_outvalues->getBuffer(), 0, samplesize * 5);
+
+	m_invalues = new AUD_Buffer(samplesize * 5); AUD_NEW("buffer")
+	memset(m_invalues->getBuffer(), 0, samplesize * 5);
+
+	m_position = 0;
+
+	// calculate coefficients
+	float omega = 2 * tan(frequency * M_PI / specs.rate);
+	float o2 = omega * omega;
+	float o4 = o2 * o2;
+	float x1 = o2 + 2 * BWPB41 * omega + 4;
+	float x2 = o2 + 2 * BWPB42 * omega + 4;
+	float y1 = o2 - 2 * BWPB41 * omega + 4;
+	float y2 = o2 - 2 * BWPB42 * omega + 4;
+	float o228 = 2 * o2 - 8;
+	float norm = x1 * x2;
+	m_coeff[0][0] = 0;
+	m_coeff[0][1] = (x1 + x2) * o228 / norm;
+	m_coeff[0][2] = (x1 * y2 + x2 * y1 + o228 * o228) / norm;
+	m_coeff[0][3] = (y1 + y2) * o228 / norm;
+	m_coeff[0][4] = y1 * y2 / norm;
+	m_coeff[1][4] = m_coeff[1][0] = o4 / norm;
+	m_coeff[1][3] = m_coeff[1][1] = 4 * o4 / norm;
+	m_coeff[1][2] = 6 * o4 / norm;
+}
+
+AUD_ButterworthReader::~AUD_ButterworthReader()
+{
+	delete m_buffer; AUD_DELETE("buffer")
+
+	delete m_outvalues; AUD_DELETE("buffer")
+	delete m_invalues; AUD_DELETE("buffer");
+}
+
+void AUD_ButterworthReader::read(int & length, sample_t* & buffer)
+{
+	sample_t* buf;
+	sample_t* outvalues;
+	sample_t* invalues;
+
+	outvalues = m_outvalues->getBuffer();
+	invalues = m_invalues->getBuffer();
+
+	AUD_Specs specs = m_reader->getSpecs();
+
+	m_reader->read(length, buf);
+
+	if(m_buffer->getSize() < length * AUD_SAMPLE_SIZE(specs))
+		m_buffer->resize(length * AUD_SAMPLE_SIZE(specs));
+
+	buffer = m_buffer->getBuffer();
+	int channels = specs.channels;
+
+	for(int channel = 0; channel < channels; channel++)
+	{
+		for(int i = 0; i < length; i++)
+		{
+			invalues[m_position * CC] = buf[i * CC];
+			outvalues[m_position * CC] = 0;
+
+			for(int j = 0; j < 4; j++)
+			{
+				outvalues[m_position * CC] += m_coeff[1][j] *
+									invalues[((m_position + j) % 5) * CC] -
+									m_coeff[0][j] *
+									outvalues[((m_position + j) % 5) * CC];
+			}
+
+			buffer[i * CC] = outvalues[m_position * CC];
+
+			m_position = (m_position + 4) % 5;
+		}
+	}
+}

intern/audaspace/FX/AUD_ButterworthReader.h

@@ -0,0 +1,83 @@
+/*
+ * $Id$
+ *
+ * ***** BEGIN LGPL LICENSE BLOCK *****
+ *
+ * Copyright 2009 Jörg Hermann Müller
+ *
+ * This file is part of AudaSpace.
+ *
+ * AudaSpace is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU Lesser General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * AudaSpace 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 Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public License
+ * along with AudaSpace.  If not, see <http://www.gnu.org/licenses/>.
+ *
+ * ***** END LGPL LICENSE BLOCK *****
+ */
+
+#ifndef AUD_BUTTERWORTHREADER
+#define AUD_BUTTERWORTHREADER
+
+#include "AUD_EffectReader.h"
+class AUD_Buffer;
+
+/**
+ * This class represents a butterworth filter.
+ */
+class AUD_ButterworthReader : public AUD_EffectReader
+{
+private:
+	/**
+	 * The playback buffer.
+	 */
+	AUD_Buffer *m_buffer;
+
+	/**
+	 * The last out values buffer.
+	 */
+	AUD_Buffer *m_outvalues;
+
+	/**
+	 * The last in values buffer.
+	 */
+	AUD_Buffer *m_invalues;
+
+	/**
+	 * The position for buffer cycling.
+	 */
+	int m_position;
+
+	/**
+	 * Filter coefficients.
+	 */
+	float m_coeff[2][5];
+
+public:
+	/**
+	 * Creates a new butterworth reader.
+	 * \param reader The reader to read from.
+	 * \param attack The attack value in seconds.
+	 * \param release The release value in seconds.
+	 * \param threshold The threshold value.
+	 * \param arthreshold The attack/release threshold value.
+	 * \exception AUD_Exception Thrown if the reader specified is NULL.
+	 */
+	AUD_ButterworthReader(AUD_IReader* reader, float frequency);
+
+	/**
+	 * Destroys the reader.
+	 */
+	virtual ~AUD_ButterworthReader();
+
+	virtual void read(int & length, sample_t* & buffer);
+};
+
+#endif //AUD_BUTTERWORTHREADER

intern/audaspace/FX/AUD_DelayReader.cpp

@@ -77,31 +77,26 @@ void AUD_DelayReader::read(int & length, sample_t* & buffer)
 {
 	if(m_remdelay > 0)
 	{
-		int samplesize = AUD_SAMPLE_SIZE(m_reader->getSpecs());
+		AUD_Specs specs = m_reader->getSpecs();
+		int samplesize = AUD_SAMPLE_SIZE(specs);
 
-		if(m_buffer->getSize() < length*samplesize)
-			m_buffer->resize(length*samplesize);
+		if(m_buffer->getSize() < length * samplesize)
+			m_buffer->resize(length * samplesize);
 
 		if(length > m_remdelay)
 		{
-			if(getSpecs().format == AUD_FORMAT_U8)
-				memset(m_buffer->getBuffer(), 0x80, m_remdelay*samplesize);
-			else
-				memset(m_buffer->getBuffer(), 0, m_remdelay*samplesize);
+			memset(m_buffer->getBuffer(), 0, m_remdelay * samplesize);
 			int len = length - m_remdelay;
 			m_reader->read(len, buffer);
-			memcpy(m_buffer->getBuffer()+m_remdelay*samplesize,
-				   buffer, len*samplesize);
+			memcpy(m_buffer->getBuffer() + m_remdelay * specs.channels,
+				   buffer, len * samplesize);
 			if(len < length-m_remdelay)
 				length = m_remdelay + len;
 			m_remdelay = 0;
 		}
 		else
 		{
-			if(getSpecs().format == AUD_FORMAT_U8)
-				memset(m_buffer->getBuffer(), 0x80, length*samplesize);
-			else
-				memset(m_buffer->getBuffer(), 0, length*samplesize);
+			memset(m_buffer->getBuffer(), 0, length * samplesize);
 			m_remdelay -= length;
 		}
 		buffer = m_buffer->getBuffer();

intern/audaspace/FX/AUD_DoubleReader.cpp

@@ -137,15 +137,16 @@ void AUD_DoubleReader::read(int & length, sample_t* & buffer)
 		m_reader1->read(len, buffer);
 		if(len < length)
 		{
-			int samplesize = AUD_SAMPLE_SIZE(m_reader1->getSpecs());
+			AUD_Specs specs = m_reader1->getSpecs();
+			int samplesize = AUD_SAMPLE_SIZE(specs);
 			if(m_buffer->getSize() < length * samplesize)
 				m_buffer->resize(length * samplesize);
-			memcpy(m_buffer->getBuffer(), buffer, len*samplesize);
+			memcpy(m_buffer->getBuffer(), buffer, len * samplesize);
 			len = length - len;
 			length -= len;
 			m_reader2->read(len, buffer);
-			memcpy(m_buffer->getBuffer() + length*samplesize,
-				   buffer, len*samplesize);
+			memcpy(m_buffer->getBuffer() + length * specs.channels, buffer,
+				   len * samplesize);
 			length += len;
 			buffer = m_buffer->getBuffer();
 			m_finished1 = true;

intern/audaspace/FX/AUD_EnvelopeFactory.cpp

@@ -0,0 +1,58 @@
+/*
+ * $Id$
+ *
+ * ***** BEGIN LGPL LICENSE BLOCK *****
+ *
+ * Copyright 2009 Jörg Hermann Müller
+ *
+ * This file is part of AudaSpace.
+ *
+ * AudaSpace is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU Lesser General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * AudaSpace 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 Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public License
+ * along with AudaSpace.  If not, see <http://www.gnu.org/licenses/>.
+ *
+ * ***** END LGPL LICENSE BLOCK *****
+ */
+
+#include "AUD_EnvelopeFactory.h"
+#include "AUD_EnvelopeReader.h"
+
+AUD_EnvelopeFactory::AUD_EnvelopeFactory(AUD_IFactory* factory, float attack,
+										 float release, float threshold,
+										 float arthreshold) :
+		AUD_EffectFactory(factory),
+		m_attack(attack),
+		m_release(release),
+		m_threshold(threshold),
+		m_arthreshold(arthreshold) {}
+
+AUD_EnvelopeFactory::AUD_EnvelopeFactory(float attack, float release,
+										 float threshold, float arthreshold) :
+		AUD_EffectFactory(0),
+		m_attack(attack),
+		m_release(release),
+		m_threshold(threshold),
+		m_arthreshold(arthreshold) {}
+
+AUD_IReader* AUD_EnvelopeFactory::createReader()
+{
+	AUD_IReader* reader = getReader();
+
+	if(reader != 0)
+	{
+		reader = new AUD_EnvelopeReader(reader, m_attack, m_release,
+										m_threshold, m_arthreshold);
+		AUD_NEW("reader")
+	}
+
+	return reader;
+}

intern/audaspace/FX/AUD_EnvelopeFactory.h

@@ -0,0 +1,82 @@
+/*
+ * $Id$
+ *
+ * ***** BEGIN LGPL LICENSE BLOCK *****
+ *
+ * Copyright 2009 Jörg Hermann Müller
+ *
+ * This file is part of AudaSpace.
+ *
+ * AudaSpace is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU Lesser General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * AudaSpace 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 Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public License
+ * along with AudaSpace.  If not, see <http://www.gnu.org/licenses/>.
+ *
+ * ***** END LGPL LICENSE BLOCK *****
+ */
+
+#ifndef AUD_ENVELOPEFACTORY
+#define AUD_ENVELOPEFACTORY
+
+#include "AUD_EffectFactory.h"
+
+/**
+ * This factory creates an envelope follower reader.
+ */
+class AUD_EnvelopeFactory : public AUD_EffectFactory
+{
+private:
+	/**
+	 * The attack value in seconds.
+	 */
+	float m_attack;
+
+	/**
+	 * The release value in seconds.
+	 */
+	float m_release;
+
+	/**
+	 * The threshold value.
+	 */
+	float m_threshold;
+
+	/**
+	 * The attack/release threshold value.
+	 */
+	float m_arthreshold;
+
+public:
+	/**
+	 * Creates a new envelope factory.
+	 * \param factory The input factory.
+	 * \param attack The attack value in seconds.
+	 * \param release The release value in seconds.
+	 * \param threshold The threshold value.
+	 * \param arthreshold The attack/release threshold value.
+	 */
+	AUD_EnvelopeFactory(AUD_IFactory* factory, float attack, float release,
+						float threshold, float arthreshold);
+
+	/**
+	 * Creates a new envelope factory.
+	 * \param attack The attack value in seconds.
+	 * \param release The release value in seconds.
+	 * \param threshold The threshold value.
+	 * \param arthreshold The attack/release threshold value.
+	 */
+	AUD_EnvelopeFactory(float attack, float release, float threshold,
+						float arthreshold);
+
+	virtual AUD_IReader* createReader();
+};
+
+#endif //AUD_ENVELOPEFACTORY

intern/audaspace/FX/AUD_EnvelopeReader.cpp

@@ -0,0 +1,86 @@
+/*
+ * $Id$
+ *
+ * ***** BEGIN LGPL LICENSE BLOCK *****
+ *
+ * Copyright 2009 Jörg Hermann Müller
+ *
+ * This file is part of AudaSpace.
+ *
+ * AudaSpace is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU Lesser General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * AudaSpace 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 Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public License
+ * along with AudaSpace.  If not, see <http://www.gnu.org/licenses/>.
+ *
+ * ***** END LGPL LICENSE BLOCK *****
+ */
+
+#include "AUD_EnvelopeReader.h"
+#include "AUD_Buffer.h"
+
+#include <cstring>
+#include <cmath>
+
+AUD_EnvelopeReader::AUD_EnvelopeReader(AUD_IReader* reader, float attack,
+									   float release, float threshold,
+									   float arthreshold) :
+		AUD_EffectReader(reader),
+		m_threshold(threshold)
+{
+	AUD_Specs specs = reader->getSpecs();
+	int samplesize = AUD_SAMPLE_SIZE(specs);
+
+	m_buffer = new AUD_Buffer(); AUD_NEW("buffer")
+
+	m_envelopes = new AUD_Buffer(samplesize);
+	AUD_NEW("buffer")
+	memset(m_envelopes->getBuffer(), 0, samplesize);
+
+	m_bAttack = pow(arthreshold, 1.0/(specs.rate * attack));
+	m_bRelease = pow(arthreshold, 1.0/(specs.rate * release));
+}
+
+AUD_EnvelopeReader::~AUD_EnvelopeReader()
+{
+	delete m_buffer; AUD_DELETE("buffer")
+	delete m_envelopes; AUD_DELETE("buffer")
+}
+
+void AUD_EnvelopeReader::read(int & length, sample_t* & buffer)
+{
+	sample_t* buf;
+	sample_t* envelopes;
+	envelopes = m_envelopes->getBuffer();
+
+	AUD_Specs specs = m_reader->getSpecs();
+
+	m_reader->read(length, buf);
+	if(m_buffer->getSize() < length * AUD_SAMPLE_SIZE(specs))
+		m_buffer->resize(length * AUD_SAMPLE_SIZE(specs));
+
+	buffer = m_buffer->getBuffer();
+
+	sample_t value;
+
+	for(int channel = 0; channel < specs.channels; channel++)
+	{
+		for(int i = 0; i < length; i++)
+		{
+			value = fabs(buf[i * specs.channels + channel]);
+			if(value < m_threshold)
+				value = 0.0f;
+
+			buffer[i * specs.channels + channel] = envelopes[channel] =
+				((value > envelopes[channel]) ? m_bAttack : m_bRelease) *
+				 (envelopes[channel] - value) + value;
+		}
+	}
+}

intern/audaspace/FX/AUD_EnvelopeReader.h

@@ -0,0 +1,84 @@
+/*
+ * $Id$
+ *
+ * ***** BEGIN LGPL LICENSE BLOCK *****
+ *
+ * Copyright 2009 Jörg Hermann Müller
+ *
+ * This file is part of AudaSpace.
+ *
+ * AudaSpace is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU Lesser General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * AudaSpace 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 Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public License
+ * along with AudaSpace.  If not, see <http://www.gnu.org/licenses/>.
+ *
+ * ***** END LGPL LICENSE BLOCK *****
+ */
+
+#ifndef AUD_ENVELOPEREADER
+#define AUD_ENVELOPEREADER
+
+#include "AUD_EffectReader.h"
+class AUD_Buffer;
+
+/**
+ * This class represents an envelope follower.
+ */
+class AUD_EnvelopeReader : public AUD_EffectReader
+{
+private:
+	/**
+	 * The playback buffer.
+	 */
+	AUD_Buffer *m_buffer;
+
+	/**
+	 * The last envelopes buffer.
+	 */
+	AUD_Buffer *m_envelopes;
+
+	/**
+	 * Attack b value.
+	 */
+	float m_bAttack;
+
+	/**
+	 * Release b value.
+	 */
+	float m_bRelease;
+
+	/**
+	 * Threshold value.
+	 */
+	float m_threshold;
+
+public:
+	/**
+	 * Creates a new envelope reader.
+	 * \param reader The reader to read from.
+	 * \param attack The attack value in seconds.
+	 * \param release The release value in seconds.
+	 * \param threshold The threshold value.
+	 * \param arthreshold The attack/release threshold value.
+	 * \exception AUD_Exception Thrown if the reader specified is NULL.
+	 */
+	AUD_EnvelopeReader(AUD_IReader* reader, float attack, float release,
+					   float threshold, float arthreshold);
+
+	/**
+	 * Destroys the reader.
+	 */
+	virtual ~AUD_EnvelopeReader();
+
+	virtual void read(int & length, sample_t* & buffer);
+};
+
+#endif //AUD_ENVELOPEREADER

intern/audaspace/FX/AUD_FaderReader.cpp

@@ -35,35 +35,6 @@ AUD_FaderReader::AUD_FaderReader(AUD_IReader* reader, AUD_FadeType type,
 		m_start(start),
 		m_length(length)
 {
-	int bigendian = 1;
-	bigendian = (((char*)&bigendian)[0]) ? 0: 1; // 1 if Big Endian
-
-	switch(m_reader->getSpecs().format)
-	{
-	case AUD_FORMAT_S16:
-		m_adjust = AUD_volume_adjust<int16_t>;
-		break;
-	case AUD_FORMAT_S32:
-		m_adjust = AUD_volume_adjust<int32_t>;
-		break;
-	case AUD_FORMAT_FLOAT32:
-		m_adjust = AUD_volume_adjust<float>;
-		break;
-	case AUD_FORMAT_FLOAT64:
-		m_adjust = AUD_volume_adjust<double>;
-		break;
-	case AUD_FORMAT_U8:
-		m_adjust = AUD_volume_adjust_u8;
-		break;
-	case AUD_FORMAT_S24:
-		m_adjust = bigendian ? AUD_volume_adjust_s24_be :
-							   AUD_volume_adjust_s24_le;
-		break;
-	default:
-		delete m_reader;
-		AUD_THROW(AUD_ERROR_READER);
-	}
-
 	m_buffer = new AUD_Buffer(); AUD_NEW("buffer")
 }
 
@@ -93,9 +64,7 @@ void AUD_FaderReader::read(int & length, sample_t* & buffer)
 		if(m_type != AUD_FADE_OUT)
 		{
 			buffer = m_buffer->getBuffer();
-			memset(buffer,
-				   specs.format == AUD_FORMAT_U8 ? 0x80 : 0,
-				   length * samplesize);
+			memset(buffer, 0, length * samplesize);
 		}
 	}
 	else if(position / (float)specs.rate >= m_start+m_length)
@@ -103,9 +72,7 @@ void AUD_FaderReader::read(int & length, sample_t* & buffer)
 		if(m_type == AUD_FADE_OUT)
 		{
 			buffer = m_buffer->getBuffer();
-			memset(buffer,
-				   specs.format == AUD_FORMAT_U8 ? 0x80 : 0,
-				   length * samplesize);
+			memset(buffer, 0, length * samplesize);
 		}
 	}
 	else
@@ -113,19 +80,21 @@ void AUD_FaderReader::read(int & length, sample_t* & buffer)
 		sample_t* buf = m_buffer->getBuffer();
 		float volume;
 
-		for(int i = 0; i < length; i++)
+		for(int i = 0; i < length * specs.channels; i++)
 		{
-			volume = (((position+i)/(float)specs.rate)-m_start) / m_length;
-			if(volume > 1.0f)
-				volume = 1.0f;
-			else if(volume < 0.0f)
-				volume = 0.0f;
+			if(i % specs.channels == 0)
+			{
+				volume = (((position+i)/(float)specs.rate)-m_start) / m_length;
+				if(volume > 1.0f)
+					volume = 1.0f;
+				else if(volume < 0.0f)
+					volume = 0.0f;
 
-			if(m_type == AUD_FADE_OUT)
-				volume = 1.0f - volume;
+				if(m_type == AUD_FADE_OUT)
+					volume = 1.0f - volume;
+			}
 
-			m_adjust(buf + i * samplesize, buffer + i * samplesize,
-					 specs.channels, volume);
+			buf[i] = buffer[i] * volume;
 		}
 
 		buffer = buf;

intern/audaspace/FX/AUD_FaderReader.h

@@ -27,7 +27,6 @@
 #define AUD_FADERREADER
 
 #include "AUD_EffectReader.h"
-#include "AUD_ConverterFunctions.h"
 class AUD_Buffer;
 
 /**
@@ -58,11 +57,6 @@ private:
 	 */
 	float m_length;
 
-	/**
-	 * Volume adjustment function.
-	 */
-	AUD_volume_adjust_f m_adjust;
-
 public:
 	/**
 	 * Creates a new fader reader.

intern/audaspace/FX/AUD_HighpassFactory.cpp

@@ -0,0 +1,51 @@
+/*
+ * $Id$
+ *
+ * ***** BEGIN LGPL LICENSE BLOCK *****
+ *
+ * Copyright 2009 Jörg Hermann Müller
+ *
+ * This file is part of AudaSpace.
+ *
+ * AudaSpace is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU Lesser General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * AudaSpace 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 Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public License
+ * along with AudaSpace.  If not, see <http://www.gnu.org/licenses/>.
+ *
+ * ***** END LGPL LICENSE BLOCK *****
+ */
+
+#include "AUD_HighpassFactory.h"
+#include "AUD_HighpassReader.h"
+
+AUD_HighpassFactory::AUD_HighpassFactory(AUD_IFactory* factory, float frequency,
+										 float Q) :
+		AUD_EffectFactory(factory),
+		m_frequency(frequency),
+		m_Q(Q) {}
+
+AUD_HighpassFactory::AUD_HighpassFactory(float frequency, float Q) :
+		AUD_EffectFactory(0),
+		m_frequency(frequency),
+		m_Q(Q) {}
+
+AUD_IReader* AUD_HighpassFactory::createReader()
+{
+	AUD_IReader* reader = getReader();
+
+	if(reader != 0)
+	{
+		reader = new AUD_HighpassReader(reader, m_frequency, m_Q);
+		AUD_NEW("reader")
+	}
+
+	return reader;
+}

intern/audaspace/FX/AUD_HighpassFactory.h

@@ -23,54 +23,44 @@
  * ***** END LGPL LICENSE BLOCK *****
  */
 
-#ifndef AUD_SDLMIXER
-#define AUD_SDLMIXER
+#ifndef AUD_HIGHPASSFACTORY
+#define AUD_HIGHPASSFACTORY
 
-#include "AUD_IMixer.h"
-class AUD_SDLMixerFactory;
-#include <list>
-
-struct AUD_SDLMixerBuffer
-{
-	sample_t* buffer;
-	int length;
-	float volume;
-};
+#include "AUD_EffectFactory.h"
 
 /**
- * This class is able to mix audiosignals with the help of SDL.
+ * This factory creates a highpass filter reader.
  */
-class AUD_SDLMixer : public AUD_IMixer
+class AUD_HighpassFactory : public AUD_EffectFactory
 {
 private:
 	/**
-	 * The mixer factory that prepares all readers for superposition.
+	 * The attack value in seconds.
 	 */
-	AUD_SDLMixerFactory* m_factory;
+	float m_frequency;
 
 	/**
-	 * The list of buffers to superpose.
+	 * The Q factor.
 	 */
-	std::list<AUD_SDLMixerBuffer> m_buffers;
-
-	/**
-	 * The size of an output sample.
-	 */
-	int m_samplesize;
+	float m_Q;
 
 public:
 	/**
-	 * Creates the mixer.
+	 * Creates a new highpass factory.
+	 * \param factory The input factory.
+	 * \param frequency The cutoff frequency.
+	 * \param Q The Q factor.
 	 */
-	AUD_SDLMixer();
+	AUD_HighpassFactory(AUD_IFactory* factory, float frequency, float Q = 1.0);
 
-	virtual ~AUD_SDLMixer();
+	/**
+	 * Creates a new highpass factory.
+	 * \param frequency The cutoff frequency.
+	 * \param Q The Q factor.
+	 */
+	AUD_HighpassFactory(float frequency, float Q = 1.0);
 
-	virtual AUD_IReader* prepare(AUD_IReader* reader);
-	virtual void setSpecs(AUD_Specs specs);
-	virtual void add(sample_t* buffer, AUD_Specs specs, int length,
-					 float volume);
-	virtual void superpose(sample_t* buffer, int length, float volume);
+	virtual AUD_IReader* createReader();
 };
 
-#endif //AUD_SDLMIXER
+#endif //AUD_HIGHPASSFACTORY

intern/audaspace/FX/AUD_HighpassReader.cpp

@@ -0,0 +1,112 @@
+/*
+ * $Id$
+ *
+ * ***** BEGIN LGPL LICENSE BLOCK *****
+ *
+ * Copyright 2009 Jörg Hermann Müller
+ *
+ * This file is part of AudaSpace.
+ *
+ * AudaSpace is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU Lesser General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * AudaSpace 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 Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public License
+ * along with AudaSpace.  If not, see <http://www.gnu.org/licenses/>.
+ *
+ * ***** END LGPL LICENSE BLOCK *****
+ */
+
+#include "AUD_HighpassReader.h"
+#include "AUD_Buffer.h"
+
+#include <cstring>
+#include <cmath>
+
+#define CC channels + channel
+
+AUD_HighpassReader::AUD_HighpassReader(AUD_IReader* reader, float frequency,
+									   float Q) :
+		AUD_EffectReader(reader)
+{
+	AUD_Specs specs = reader->getSpecs();
+	int samplesize = AUD_SAMPLE_SIZE(specs);
+
+	m_buffer = new AUD_Buffer(); AUD_NEW("buffer")
+
+	m_outvalues = new AUD_Buffer(samplesize * AUD_HIGHPASS_ORDER);
+	AUD_NEW("buffer")
+	memset(m_outvalues->getBuffer(), 0, samplesize * AUD_HIGHPASS_ORDER);
+
+	m_invalues = new AUD_Buffer(samplesize * AUD_HIGHPASS_ORDER);
+	AUD_NEW("buffer")
+	memset(m_invalues->getBuffer(), 0, samplesize * AUD_HIGHPASS_ORDER);
+
+	m_position = 0;
+
+	// calculate coefficients
+	float w0 = 2.0 * M_PI * frequency / specs.rate;
+	float alpha = sin(w0) / (2 * Q);
+	float norm = 1 + alpha;
+	m_coeff[0][0] = 0;
+	m_coeff[0][1] = -2 * cos(w0) / norm;
+	m_coeff[0][2] = (1 - alpha) / norm;
+	m_coeff[1][2] = m_coeff[1][0] = (1 + cos(w0)) / (2 * norm);
+	m_coeff[1][1] = (-1 - cos(w0)) / norm;
+}
+
+AUD_HighpassReader::~AUD_HighpassReader()
+{
+	delete m_buffer; AUD_DELETE("buffer")
+
+	delete m_outvalues; AUD_DELETE("buffer")
+	delete m_invalues; AUD_DELETE("buffer");
+}
+
+void AUD_HighpassReader::read(int & length, sample_t* & buffer)
+{
+	sample_t* buf;
+	sample_t* outvalues;
+	sample_t* invalues;
+
+	outvalues = m_outvalues->getBuffer();
+	invalues = m_invalues->getBuffer();
+
+	AUD_Specs specs = m_reader->getSpecs();
+
+	m_reader->read(length, buf);
+
+	if(m_buffer->getSize() < length * AUD_SAMPLE_SIZE(specs))
+		m_buffer->resize(length * AUD_SAMPLE_SIZE(specs));
+
+	buffer = m_buffer->getBuffer();
+	int channels = specs.channels;
+
+	for(int channel = 0; channel < channels; channel++)
+	{
+		for(int i = 0; i < length; i++)
+		{
+			invalues[m_position * CC] = buf[i * CC];
+			outvalues[m_position * CC] = 0;
+
+			for(int j = 0; j < AUD_HIGHPASS_ORDER; j++)
+			{
+				outvalues[m_position * CC] += m_coeff[1][j] *
+						invalues[((m_position + j) % AUD_HIGHPASS_ORDER) * CC] -
+						m_coeff[0][j] *
+						outvalues[((m_position + j) % AUD_HIGHPASS_ORDER) * CC];
+			}
+
+			buffer[i * CC] = outvalues[m_position * CC];
+
+			m_position = (m_position + AUD_HIGHPASS_ORDER-1) %
+						 AUD_HIGHPASS_ORDER;
+		}
+	}
+}

intern/audaspace/FX/AUD_HighpassReader.h

@@ -0,0 +1,83 @@
+/*
+ * $Id$
+ *
+ * ***** BEGIN LGPL LICENSE BLOCK *****
+ *
+ * Copyright 2009 Jörg Hermann Müller
+ *
+ * This file is part of AudaSpace.
+ *
+ * AudaSpace is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU Lesser General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * AudaSpace 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 Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public License
+ * along with AudaSpace.  If not, see <http://www.gnu.org/licenses/>.
+ *
+ * ***** END LGPL LICENSE BLOCK *****
+ */
+
+#ifndef AUD_HIGHPASSREADER
+#define AUD_HIGHPASSREADER
+
+#include "AUD_EffectReader.h"
+class AUD_Buffer;
+
+#define AUD_HIGHPASS_ORDER 3
+
+/**
+ * This class represents a highpass filter.
+ */
+class AUD_HighpassReader : public AUD_EffectReader
+{
+private:
+	/**
+	 * The playback buffer.
+	 */
+	AUD_Buffer *m_buffer;
+
+	/**
+	 * The last out values buffer.
+	 */
+	AUD_Buffer *m_outvalues;
+
+	/**
+	 * The last in values buffer.
+	 */
+	AUD_Buffer *m_invalues;
+
+	/**
+	 * The position for buffer cycling.
+	 */
+	int m_position;
+
+	/**
+	 * Filter coefficients.
+	 */
+	float m_coeff[2][AUD_HIGHPASS_ORDER];
+
+public:
+	/**
+	 * Creates a new highpass reader.
+	 * \param reader The reader to read from.
+	 * \param frequency The cutoff frequency.
+	 * \param Q The Q factor.
+	 * \exception AUD_Exception Thrown if the reader specified is NULL.
+	 */
+	AUD_HighpassReader(AUD_IReader* reader, float frequency, float Q);
+
+	/**
+	 * Destroys the reader.
+	 */
+	virtual ~AUD_HighpassReader();
+
+	virtual void read(int & length, sample_t* & buffer);
+};
+
+#endif //AUD_HIGHPASSREADER

intern/audaspace/FX/AUD_LoopReader.cpp

@@ -27,7 +27,6 @@
 #include "AUD_Buffer.h"
 
 #include <cstring>
-#include <stdio.h>
 
 AUD_LoopReader::AUD_LoopReader(AUD_IReader* reader, int loop) :
 		AUD_EffectReader(reader), m_loop(loop)
@@ -62,7 +61,8 @@ bool AUD_LoopReader::notify(AUD_Message &message)
 
 void AUD_LoopReader::read(int & length, sample_t* & buffer)
 {
-	int samplesize = AUD_SAMPLE_SIZE(m_reader->getSpecs());
+	AUD_Specs specs = m_reader->getSpecs();
+	int samplesize = AUD_SAMPLE_SIZE(specs);
 
 	int len = length;
 
@@ -72,10 +72,10 @@ void AUD_LoopReader::read(int & length, sample_t* & buffer)
 	{
 		int pos = 0;
 
-		if(m_buffer->getSize() < length*samplesize)
-			m_buffer->resize(length*samplesize);
+		if(m_buffer->getSize() < length * samplesize)
+			m_buffer->resize(length * samplesize);
 
-		memcpy(m_buffer->getBuffer() + pos * samplesize,
+		memcpy(m_buffer->getBuffer() + pos * specs.channels,
 			   buffer, len * samplesize);
 
 		pos += len;
@@ -93,7 +93,7 @@ void AUD_LoopReader::read(int & length, sample_t* & buffer)
 			if(!len)
 				break;
 
-			memcpy(m_buffer->getBuffer() + pos * samplesize,
+			memcpy(m_buffer->getBuffer() + pos * specs.channels,
 				   buffer, len * samplesize);
 
 			pos += len;

intern/audaspace/FX/AUD_LowpassFactory.cpp

@@ -0,0 +1,51 @@
+/*
+ * $Id$
+ *
+ * ***** BEGIN LGPL LICENSE BLOCK *****
+ *
+ * Copyright 2009 Jörg Hermann Müller
+ *
+ * This file is part of AudaSpace.
+ *
+ * AudaSpace is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU Lesser General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * AudaSpace 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 Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public License
+ * along with AudaSpace.  If not, see <http://www.gnu.org/licenses/>.
+ *
+ * ***** END LGPL LICENSE BLOCK *****
+ */
+
+#include "AUD_LowpassFactory.h"
+#include "AUD_LowpassReader.h"
+
+AUD_LowpassFactory::AUD_LowpassFactory(AUD_IFactory* factory, float frequency,
+									   float Q) :
+		AUD_EffectFactory(factory),
+		m_frequency(frequency),
+		m_Q(Q) {}
+
+AUD_LowpassFactory::AUD_LowpassFactory(float frequency, float Q) :
+		AUD_EffectFactory(0),
+		m_frequency(frequency),
+		m_Q(Q) {}
+
+AUD_IReader* AUD_LowpassFactory::createReader()
+{
+	AUD_IReader* reader = getReader();
+
+	if(reader != 0)
+	{
+		reader = new AUD_LowpassReader(reader, m_frequency, m_Q);
+		AUD_NEW("reader")
+	}
+
+	return reader;
+}

intern/audaspace/FX/AUD_LowpassFactory.h

@@ -0,0 +1,66 @@
+/*
+ * $Id$
+ *
+ * ***** BEGIN LGPL LICENSE BLOCK *****
+ *
+ * Copyright 2009 Jörg Hermann Müller
+ *
+ * This file is part of AudaSpace.
+ *
+ * AudaSpace is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU Lesser General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * AudaSpace 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 Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public License
+ * along with AudaSpace.  If not, see <http://www.gnu.org/licenses/>.
+ *
+ * ***** END LGPL LICENSE BLOCK *****
+ */
+
+#ifndef AUD_LOWPASSFACTORY
+#define AUD_LOWPASSFACTORY
+
+#include "AUD_EffectFactory.h"
+
+/**
+ * This factory creates a lowpass filter reader.
+ */
+class AUD_LowpassFactory : public AUD_EffectFactory
+{
+private:
+	/**
+	 * The attack value in seconds.
+	 */
+	float m_frequency;
+
+	/**
+	 * The Q factor.
+	 */
+	float m_Q;
+
+public:
+	/**
+	 * Creates a new lowpass factory.
+	 * \param factory The input factory.
+	 * \param frequency The cutoff frequency.
+	 * \param Q The Q factor.
+	 */
+	AUD_LowpassFactory(AUD_IFactory* factory, float frequency, float Q = 1.0);
+
+	/**
+	 * Creates a new lowpass factory.
+	 * \param frequency The cutoff frequency.
+	 * \param Q The Q factor.
+	 */
+	AUD_LowpassFactory(float frequency, float Q = 1.0);
+
+	virtual AUD_IReader* createReader();
+};
+
+#endif //AUD_LOWPASSFACTORY

intern/audaspace/FX/AUD_LowpassReader.cpp

@@ -0,0 +1,111 @@
+/*
+ * $Id$
+ *
+ * ***** BEGIN LGPL LICENSE BLOCK *****
+ *
+ * Copyright 2009 Jörg Hermann Müller
+ *
+ * This file is part of AudaSpace.
+ *
+ * AudaSpace is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU Lesser General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * AudaSpace 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 Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public License
+ * along with AudaSpace.  If not, see <http://www.gnu.org/licenses/>.
+ *
+ * ***** END LGPL LICENSE BLOCK *****
+ */
+
+#include "AUD_LowpassReader.h"
+#include "AUD_Buffer.h"
+
+#include <cstring>
+#include <cmath>
+
+#define CC channels + channel
+
+AUD_LowpassReader::AUD_LowpassReader(AUD_IReader* reader, float frequency,
+									 float Q) :
+		AUD_EffectReader(reader)
+{
+	AUD_Specs specs = reader->getSpecs();
+	int samplesize = AUD_SAMPLE_SIZE(specs);
+
+	m_buffer = new AUD_Buffer(); AUD_NEW("buffer")
+
+	m_outvalues = new AUD_Buffer(samplesize * AUD_LOWPASS_ORDER);
+	AUD_NEW("buffer")
+	memset(m_outvalues->getBuffer(), 0, samplesize * AUD_LOWPASS_ORDER);
+
+	m_invalues = new AUD_Buffer(samplesize * AUD_LOWPASS_ORDER);
+	AUD_NEW("buffer")
+	memset(m_invalues->getBuffer(), 0, samplesize * AUD_LOWPASS_ORDER);
+
+	m_position = 0;
+
+	// calculate coefficients
+	float w0 = 2.0 * M_PI * frequency / specs.rate;
+	float alpha = sin(w0) / (2 * Q);
+	float norm = 1 + alpha;
+	m_coeff[0][0] = 0;
+	m_coeff[0][1] = -2 * cos(w0) / norm;
+	m_coeff[0][2] = (1 - alpha) / norm;
+	m_coeff[1][2] = m_coeff[1][0] = (1 - cos(w0)) / (2 * norm);
+	m_coeff[1][1] = (1 - cos(w0)) / norm;
+}
+
+AUD_LowpassReader::~AUD_LowpassReader()
+{
+	delete m_buffer; AUD_DELETE("buffer")
+
+	delete m_outvalues; AUD_DELETE("buffer")
+	delete m_invalues; AUD_DELETE("buffer");
+}
+
+void AUD_LowpassReader::read(int & length, sample_t* & buffer)
+{
+	sample_t* buf;
+	sample_t* outvalues;
+	sample_t* invalues;
+
+	outvalues = m_outvalues->getBuffer();
+	invalues = m_invalues->getBuffer();
+
+	AUD_Specs specs = m_reader->getSpecs();
+
+	m_reader->read(length, buf);
+
+	if(m_buffer->getSize() < length * AUD_SAMPLE_SIZE(specs))
+		m_buffer->resize(length * AUD_SAMPLE_SIZE(specs));
+
+	buffer = m_buffer->getBuffer();
+	int channels = specs.channels;
+
+	for(int channel = 0; channel < channels; channel++)
+	{
+		for(int i = 0; i < length; i++)
+		{
+			invalues[m_position * CC] = buf[i * CC];
+			outvalues[m_position * CC] = 0;
+
+			for(int j = 0; j < AUD_LOWPASS_ORDER; j++)
+			{
+				outvalues[m_position * CC] += m_coeff[1][j] *
+						invalues[((m_position + j) % AUD_LOWPASS_ORDER) * CC] -
+						m_coeff[0][j] *
+						outvalues[((m_position + j) % AUD_LOWPASS_ORDER) * CC];
+			}
+
+			buffer[i * CC] = outvalues[m_position * CC];
+
+			m_position = (m_position + AUD_LOWPASS_ORDER-1) % AUD_LOWPASS_ORDER;
+		}
+	}
+}

intern/audaspace/FX/AUD_LowpassReader.h

@@ -0,0 +1,83 @@
+/*
+ * $Id$
+ *
+ * ***** BEGIN LGPL LICENSE BLOCK *****
+ *
+ * Copyright 2009 Jörg Hermann Müller
+ *
+ * This file is part of AudaSpace.
+ *
+ * AudaSpace is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU Lesser General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * AudaSpace 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 Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public License
+ * along with AudaSpace.  If not, see <http://www.gnu.org/licenses/>.
+ *
+ * ***** END LGPL LICENSE BLOCK *****
+ */
+
+#ifndef AUD_LOWPASSREADER
+#define AUD_LOWPASSREADER
+
+#include "AUD_EffectReader.h"
+class AUD_Buffer;
+
+#define AUD_LOWPASS_ORDER 3
+
+/**
+ * This class represents a lowpass filter.
+ */
+class AUD_LowpassReader : public AUD_EffectReader
+{
+private:
+	/**
+	 * The playback buffer.
+	 */
+	AUD_Buffer *m_buffer;
+
+	/**
+	 * The last out values buffer.
+	 */
+	AUD_Buffer *m_outvalues;
+
+	/**
+	 * The last in values buffer.
+	 */
+	AUD_Buffer *m_invalues;
+
+	/**
+	 * The position for buffer cycling.
+	 */
+	int m_position;
+
+	/**
+	 * Filter coefficients.
+	 */
+	float m_coeff[2][AUD_LOWPASS_ORDER];
+
+public:
+	/**
+	 * Creates a new lowpass reader.
+	 * \param reader The reader to read from.
+	 * \param frequency The cutoff frequency.
+	 * \param Q The Q factor.
+	 * \exception AUD_Exception Thrown if the reader specified is NULL.
+	 */
+	AUD_LowpassReader(AUD_IReader* reader, float frequency, float Q);
+
+	/**
+	 * Destroys the reader.
+	 */
+	virtual ~AUD_LowpassReader();
+
+	virtual void read(int & length, sample_t* & buffer);
+};
+
+#endif //AUD_LOWPASSREADER

intern/audaspace/FX/AUD_RectifyReader.cpp

@@ -26,39 +26,11 @@
 #include "AUD_RectifyReader.h"
 #include "AUD_Buffer.h"
 
-#include <cstring>
+#include <cmath>
 
 AUD_RectifyReader::AUD_RectifyReader(AUD_IReader* reader) :
 		AUD_EffectReader(reader)
 {
-	int bigendian = 1;
-	bigendian = (((char*)&bigendian)[0]) ? 0: 1; // 1 if Big Endian
-
-	switch(m_reader->getSpecs().format)
-	{
-	case AUD_FORMAT_S16:
-		m_rectify = AUD_rectify<int16_t>;
-		break;
-	case AUD_FORMAT_S32:
-		m_rectify = AUD_rectify<int32_t>;
-		break;
-	case AUD_FORMAT_FLOAT32:
-		m_rectify = AUD_rectify<float>;
-		break;
-	case AUD_FORMAT_FLOAT64:
-		m_rectify = AUD_rectify<double>;
-		break;
-	case AUD_FORMAT_U8:
-		m_rectify = AUD_rectify_u8;
-		break;
-	case AUD_FORMAT_S24:
-		m_rectify = bigendian ? AUD_rectify_s24_be : AUD_rectify_s24_le;
-		break;
-	default:
-		delete m_reader;
-		AUD_THROW(AUD_ERROR_READER);
-	}
-
 	m_buffer = new AUD_Buffer(); AUD_NEW("buffer")
 }
 
@@ -73,10 +45,11 @@ void AUD_RectifyReader::read(int & length, sample_t* & buffer)
 	AUD_Specs specs = m_reader->getSpecs();
 
 	m_reader->read(length, buf);
-	if(m_buffer->getSize() < length*AUD_SAMPLE_SIZE(specs))
-		m_buffer->resize(length*AUD_SAMPLE_SIZE(specs));
+	if(m_buffer->getSize() < length * AUD_SAMPLE_SIZE(specs))
+		m_buffer->resize(length * AUD_SAMPLE_SIZE(specs));
 
 	buffer = m_buffer->getBuffer();
 
-	m_rectify(buffer, buf, length * specs.channels);
+	for(int i = 0; i < length * specs.channels; i++)
+		buffer[i] = fabs(buf[i]);
 }

intern/audaspace/FX/AUD_RectifyReader.h

@@ -27,7 +27,6 @@
 #define AUD_RECTIFYREADER
 
 #include "AUD_EffectReader.h"
-#include "AUD_ConverterFunctions.h"
 class AUD_Buffer;
 
 /**
@@ -41,11 +40,6 @@ private:
 	 */
 	AUD_Buffer *m_buffer;
 
-	/**
-	 * Rectifying function.
-	 */
-	AUD_rectify_f m_rectify;
-
 public:
 	/**
 	 * Creates a new rectify reader.

intern/audaspace/FX/AUD_ReverseReader.cpp

@@ -65,7 +65,7 @@ int AUD_ReverseReader::getPosition()
 void AUD_ReverseReader::read(int & length, sample_t* & buffer)
 {
 	// first correct the length
-	if(m_position+length > m_length)
+	if(m_position + length > m_length)
 		length = m_length-m_position;
 
 	if(length <= 0)
@@ -74,7 +74,8 @@ void AUD_ReverseReader::read(int & length, sample_t* & buffer)
 		return;
 	}
 
-	int samplesize = AUD_SAMPLE_SIZE(getSpecs());
+	AUD_Specs specs = getSpecs();
+	int samplesize = AUD_SAMPLE_SIZE(specs);
 
 	// resize buffer if needed
 	if(m_buffer->getSize() < length * samplesize)
@@ -86,23 +87,20 @@ void AUD_ReverseReader::read(int & length, sample_t* & buffer)
 	int len = length;
 
 	// read from reader
-	m_reader->seek(m_length-m_position-len);
+	m_reader->seek(m_length - m_position - len);
 	m_reader->read(len, buf);
 
 	// set null if reader didn't give enough data
 	if(len < length)
 	{
-		if(getSpecs().format == AUD_FORMAT_U8)
-			memset(buffer, 0x80, (length-len)*samplesize);
-		else
-			memset(buffer, 0, (length-len)*samplesize);
-		buffer += length-len;
+		memset(buffer, 0, (length - len) * samplesize);
+		buffer += (length - len) * specs.channels;
 	}
 
 	// copy the samples reverted
 	for(int i = 0; i < len; i++)
-		memcpy(buffer + i * samplesize,
-			   buf + (len - 1 - i) * samplesize,
+		memcpy(buffer + i * specs.channels,
+			   buf + (len - 1 - i) * specs.channels,
 			   samplesize);
 
 	m_position += length;

intern/audaspace/FX/AUD_VolumeReader.cpp

@@ -32,35 +32,6 @@ AUD_VolumeReader::AUD_VolumeReader(AUD_IReader* reader, float volume) :
 		AUD_EffectReader(reader),
 		m_volume(volume)
 {
-	int bigendian = 1;
-	bigendian = (((char*)&bigendian)[0]) ? 0: 1; // 1 if Big Endian
-
-	switch(m_reader->getSpecs().format)
-	{
-	case AUD_FORMAT_S16:
-		m_adjust = AUD_volume_adjust<int16_t>;
-		break;
-	case AUD_FORMAT_S32:
-		m_adjust = AUD_volume_adjust<int32_t>;
-		break;
-	case AUD_FORMAT_FLOAT32:
-		m_adjust = AUD_volume_adjust<float>;
-		break;
-	case AUD_FORMAT_FLOAT64:
-		m_adjust = AUD_volume_adjust<double>;
-		break;
-	case AUD_FORMAT_U8:
-		m_adjust = AUD_volume_adjust_u8;
-		break;
-	case AUD_FORMAT_S24:
-		m_adjust = bigendian ? AUD_volume_adjust_s24_be :
-							   AUD_volume_adjust_s24_le;
-		break;
-	default:
-		delete m_reader;
-		AUD_THROW(AUD_ERROR_READER);
-	}
-
 	m_buffer = new AUD_Buffer(); AUD_NEW("buffer")
 }
 
@@ -93,5 +64,6 @@ void AUD_VolumeReader::read(int & length, sample_t* & buffer)
 
 	buffer = m_buffer->getBuffer();
 
-	m_adjust(buffer, buf, length * specs.channels, m_volume);
+	for(int i = 0; i < length * specs.channels; i++)
+		buffer[i] = buf[i] * m_volume;
 }

intern/audaspace/FX/AUD_VolumeReader.h

@@ -27,7 +27,6 @@
 #define AUD_VOLUMEREADER
 
 #include "AUD_EffectReader.h"
-#include "AUD_ConverterFunctions.h"
 class AUD_Buffer;
 
 /**
@@ -46,11 +45,6 @@ private:
 	 */
 	float m_volume;
 
-	/**
-	 * Volume adjustment function.
-	 */
-	AUD_volume_adjust_f m_adjust;
-
 public:
 	/**
 	 * Creates a new volume reader.

intern/audaspace/OpenAL/AUD_OpenALDevice.cpp

@@ -25,7 +25,6 @@
 
 #include "AUD_OpenALDevice.h"
 #include "AUD_IReader.h"
-#include "AUD_IMixer.h"
 #include "AUD_ConverterFactory.h"
 #include "AUD_SourceCaps.h"
 
@@ -119,7 +118,7 @@ void AUD_OpenALDevice::updateStreams()
 	sample_t* buffer;
 
 	ALint info;
-	AUD_Specs specs;
+	AUD_DeviceSpecs specs = m_specs;
 
 	while(1)
 	{
@@ -145,7 +144,7 @@ void AUD_OpenALDevice::updateStreams()
 
 					if(info)
 					{
-						specs = sound->reader->getSpecs();
+						specs.specs = sound->reader->getSpecs();
 
 						// for all empty buffers
 						while(info--)
@@ -177,8 +176,8 @@ void AUD_OpenALDevice::updateStreams()
 								// fill with new data
 								alBufferData(sound->buffers[sound->current],
 											 sound->format,
-											 buffer,
-											 length * AUD_SAMPLE_SIZE(specs),
+											 buffer, length *
+											 AUD_DEVICE_SAMPLE_SIZE(specs),
 											 specs.rate);
 
 								if(alGetError() != AL_NO_ERROR)
@@ -264,7 +263,7 @@ bool AUD_OpenALDevice::isValid(AUD_Handle* handle)
 	return false;
 }
 
-AUD_OpenALDevice::AUD_OpenALDevice(AUD_Specs specs, int buffersize)
+AUD_OpenALDevice::AUD_OpenALDevice(AUD_DeviceSpecs specs, int buffersize)
 {
 	// cannot determine how many channels or which format OpenAL uses, but
 	// it at least is able to play 16 bit stereo audio
@@ -289,14 +288,17 @@ AUD_OpenALDevice::AUD_OpenALDevice(AUD_Specs specs, int buffersize)
 
 	// check for specific formats and channel counts to be played back
 	if(alIsExtensionPresent("AL_EXT_FLOAT32") == AL_TRUE)
+	{
 		specs.format = AUD_FORMAT_FLOAT32;
+		m_converter = NULL;
+	}
+	else
+		m_converter = new AUD_ConverterFactory(specs); AUD_NEW("factory")
 
 	m_useMC = alIsExtensionPresent("AL_EXT_MCFORMATS") == AL_TRUE;
 
 	alGetError();
 
-	m_converter = new AUD_ConverterFactory(specs); AUD_NEW("factory")
-
 	m_specs = specs;
 	m_buffersize = buffersize;
 	m_playing = false;
@@ -378,12 +380,13 @@ AUD_OpenALDevice::~AUD_OpenALDevice()
 	alcDestroyContext(m_context);
 	alcCloseDevice(m_device);
 
-	delete m_converter; AUD_DELETE("factory")
+	if(m_converter)
+		delete m_converter; AUD_DELETE("factory")
 
 	pthread_mutex_destroy(&m_mutex);
 }
 
-AUD_Specs AUD_OpenALDevice::getSpecs()
+AUD_DeviceSpecs AUD_OpenALDevice::getSpecs()
 {
 	return m_specs;
 }
@@ -393,45 +396,8 @@ bool AUD_OpenALDevice::getFormat(ALenum &format, AUD_Specs specs)
 	bool valid = true;
 	format = 0;
 
-	switch(specs.format)
+	switch(m_specs.format)
 	{
-	case AUD_FORMAT_U8:
-		switch(specs.channels)
-		{
-		case AUD_CHANNELS_MONO:
-			format = AL_FORMAT_MONO8;
-			break;
-		case AUD_CHANNELS_STEREO:
-			format = AL_FORMAT_STEREO8;
-			break;
-		case AUD_CHANNELS_SURROUND4:
-			if(m_useMC)
-			{
-				format = alGetEnumValue("AL_FORMAT_QUAD8");
-				break;
-			}
-		case AUD_CHANNELS_SURROUND51:
-			if(m_useMC)
-			{
-				format = alGetEnumValue("AL_FORMAT_51CHN8");
-				break;
-			}
-		case AUD_CHANNELS_SURROUND61:
-			if(m_useMC)
-			{
-				format = alGetEnumValue("AL_FORMAT_61CHN8");
-				break;
-			}
-		case AUD_CHANNELS_SURROUND71:
-			if(m_useMC)
-			{
-				format = alGetEnumValue("AL_FORMAT_71CHN8");
-				break;
-			}
-		default:
-			valid = false;
-		}
-		break;
 	case AUD_FORMAT_S16:
 		switch(specs.channels)
 		{
@@ -591,23 +557,16 @@ AUD_Handle* AUD_OpenALDevice::play(AUD_IFactory* factory, bool keep)
 	if(reader == NULL)
 		AUD_THROW(AUD_ERROR_READER);
 
-	AUD_Specs specs;
-
-	specs = reader->getSpecs();
+	AUD_DeviceSpecs specs = m_specs;
+	specs.specs = reader->getSpecs();
 
 	// check format
-	bool valid = true;
+	bool valid = specs.channels != AUD_CHANNELS_INVALID;
 
-	if(specs.format == AUD_FORMAT_INVALID)
-		valid = false;
-	else if(specs.format == AUD_FORMAT_S24 ||
-			specs.format == AUD_FORMAT_S32 ||
-			specs.format == AUD_FORMAT_FLOAT32 ||
-			specs.format == AUD_FORMAT_FLOAT64)
+	if(m_converter)
 	{
 		m_converter->setReader(reader);
 		reader = m_converter->createReader();
-		specs = reader->getSpecs();
 	}
 
 	// create the handle
@@ -618,7 +577,7 @@ AUD_Handle* AUD_OpenALDevice::play(AUD_IFactory* factory, bool keep)
 	sound->isBuffered = false;
 	sound->data_end = false;
 
-	valid &= getFormat(sound->format, specs);
+	valid &= getFormat(sound->format, specs.specs);
 
 	if(!valid)
 	{
@@ -647,7 +606,8 @@ AUD_Handle* AUD_OpenALDevice::play(AUD_IFactory* factory, bool keep)
 				length = m_buffersize;
 				reader->read(length, buf);
 				alBufferData(sound->buffers[i], sound->format, buf,
-							 length * AUD_SAMPLE_SIZE(specs), specs.rate);
+							 length * AUD_DEVICE_SAMPLE_SIZE(specs),
+							 specs.rate);
 				if(alGetError() != AL_NO_ERROR)
 					AUD_THROW(AUD_ERROR_OPENAL);
 			}
@@ -875,14 +835,16 @@ bool AUD_OpenALDevice::seek(AUD_Handle* handle, float position)
 				{
 					sample_t* buf;
 					int length;
-					AUD_Specs specs = alhandle->reader->getSpecs();
+					AUD_DeviceSpecs specs = m_specs;
+					specs.specs = alhandle->reader->getSpecs();
 
 					for(int i = 0; i < AUD_OPENAL_CYCLE_BUFFERS; i++)
 					{
 						length = m_buffersize;
 						alhandle->reader->read(length, buf);
 						alBufferData(alhandle->buffers[i], alhandle->format,
-									 buf, length * AUD_SAMPLE_SIZE(specs),
+									 buf,
+									 length * AUD_DEVICE_SAMPLE_SIZE(specs),
 									 specs.rate);
 
 						if(alGetError() != AL_NO_ERROR)
@@ -1021,6 +983,7 @@ bool AUD_OpenALDevice::setCapability(int capability, void *value)
 			// load the factory into an OpenAL buffer
 			if(factory)
 			{
+				// check if the factory is already buffered
 				lock();
 				for(AUD_BFIterator i = m_bufferedFactories->begin();
 					i != m_bufferedFactories->end(); i++)
@@ -1040,32 +1003,25 @@ bool AUD_OpenALDevice::setCapability(int capability, void *value)
 				if(reader == NULL)
 					return false;
 
-				AUD_Specs specs;
-
-				specs = reader->getSpecs();
+				AUD_DeviceSpecs specs = m_specs;
+				specs.specs = reader->getSpecs();
 
 				// determine format
 				bool valid = reader->getType() == AUD_TYPE_BUFFER;
 
 				if(valid)
 				{
-					if(specs.format == AUD_FORMAT_INVALID)
-						valid = false;
-					else if(specs.format == AUD_FORMAT_S24 ||
-							specs.format == AUD_FORMAT_S32 ||
-							specs.format == AUD_FORMAT_FLOAT32 ||
-							specs.format == AUD_FORMAT_FLOAT64)
+					if(m_converter)
 					{
 						m_converter->setReader(reader);
 						reader = m_converter->createReader();
-						specs = reader->getSpecs();
 					}
 				}
 
 				ALenum format;
 
 				if(valid)
-					valid = getFormat(format, specs);
+					valid = getFormat(format, specs.specs);
 
 				if(!valid)
 				{
@@ -1094,7 +1050,7 @@ bool AUD_OpenALDevice::setCapability(int capability, void *value)
 
 						reader->read(length, buf);
 						alBufferData(bf->buffer, format, buf,
-									 length * AUD_SAMPLE_SIZE(specs),
+									 length * AUD_DEVICE_SAMPLE_SIZE(specs),
 									 specs.rate);
 						if(alGetError() != AL_NO_ERROR)
 							AUD_THROW(AUD_ERROR_OPENAL);

intern/audaspace/OpenAL/AUD_OpenALDevice.h

@@ -56,7 +56,7 @@ private:
 	/**
 	 * The specification of the device.
 	 */
-	AUD_Specs m_specs;
+	AUD_DeviceSpecs m_specs;
 
 	/**
 	 * Whether the device has the AL_EXT_MCFORMATS extension.
@@ -64,8 +64,8 @@ private:
 	bool m_useMC;
 
 	/**
-	 * The converter factory for readers with wrong input format.
-	 */
+	* The converter factory for readers with wrong input format.
+	*/
 	AUD_ConverterFactory* m_converter;
 
 	/**
@@ -118,7 +118,7 @@ private:
 	/**
 	 * Gets the format according to the specs.
 	 * \param format The variable to put the format into.
-	 * \param specs The specs to read the format from.
+	 * \param specs The specs to read the channel count from.
 	 * \return Whether the format is valid or not.
 	 */
 	bool getFormat(ALenum &format, AUD_Specs specs);
@@ -132,7 +132,8 @@ public:
 	 * \note The buffersize will be multiplicated by three for this device.
 	 * \exception AUD_Exception Thrown if the audio device cannot be opened.
 	 */
-	AUD_OpenALDevice(AUD_Specs specs, int buffersize = AUD_DEFAULT_BUFFER_SIZE);
+	AUD_OpenALDevice(AUD_DeviceSpecs specs,
+					 int buffersize = AUD_DEFAULT_BUFFER_SIZE);
 
 	/**
 	 * Streaming thread main function.
@@ -141,7 +142,7 @@ public:
 
 	virtual ~AUD_OpenALDevice();
 
-	virtual AUD_Specs getSpecs();
+	virtual AUD_DeviceSpecs getSpecs();
 	virtual AUD_Handle* play(AUD_IFactory* factory, bool keep = false);
 	virtual bool pause(AUD_Handle* handle);
 	virtual bool resume(AUD_Handle* handle);

intern/audaspace/SDL/AUD_SDLDevice.cpp

@@ -23,7 +23,6 @@
  * ***** END LGPL LICENSE BLOCK *****
  */
 
-#include "AUD_SDLMixer.h"
 #include "AUD_SDLDevice.h"
 #include "AUD_IReader.h"
 
@@ -31,10 +30,10 @@ void AUD_SDLDevice::SDL_mix(void *data, Uint8* buffer, int length)
 {
 	AUD_SDLDevice* device = (AUD_SDLDevice*)data;
 
-	device->mix((sample_t*)buffer, length/AUD_SAMPLE_SIZE(device->m_specs));
+	device->mix((data_t*)buffer,length/AUD_DEVICE_SAMPLE_SIZE(device->m_specs));
 }
 
-AUD_SDLDevice::AUD_SDLDevice(AUD_Specs specs, int buffersize)
+AUD_SDLDevice::AUD_SDLDevice(AUD_DeviceSpecs specs, int buffersize)
 {
 	if(specs.channels == AUD_CHANNELS_INVALID)
 		specs.channels = AUD_CHANNELS_STEREO;
@@ -69,9 +68,6 @@ AUD_SDLDevice::AUD_SDLDevice(AUD_Specs specs, int buffersize)
 	else
 		AUD_THROW(AUD_ERROR_SDL);
 
-	m_mixer = new AUD_SDLMixer(); AUD_NEW("mixer")
-	m_mixer->setSpecs(m_specs);
-
 	create();
 }
 

intern/audaspace/SDL/AUD_SDLDevice.h

@@ -55,7 +55,8 @@ public:
 	 * \note The specification really used for opening the device may differ.
 	 * \exception AUD_Exception Thrown if the audio device cannot be opened.
 	 */
-	AUD_SDLDevice(AUD_Specs specs, int buffersize = AUD_DEFAULT_BUFFER_SIZE);
+	AUD_SDLDevice(AUD_DeviceSpecs specs,
+				  int buffersize = AUD_DEFAULT_BUFFER_SIZE);
 
 	/**
 	 * Closes the SDL audio device.

intern/audaspace/SDL/AUD_SDLMixer.cpp

@@ -1,83 +0,0 @@
-/*
- * $Id$
- *
- * ***** BEGIN LGPL LICENSE BLOCK *****
- *
- * Copyright 2009 Jörg Hermann Müller
- *
- * This file is part of AudaSpace.
- *
- * AudaSpace is free software: you can redistribute it and/or modify
- * it under the terms of the GNU Lesser General Public License as published by
- * the Free Software Foundation, either version 3 of the License, or
- * (at your option) any later version.
- *
- * AudaSpace 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 Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public License
- * along with AudaSpace.  If not, see <http://www.gnu.org/licenses/>.
- *
- * ***** END LGPL LICENSE BLOCK *****
- */
-
-#include "AUD_SDLMixer.h"
-#include "AUD_SDLMixerFactory.h"
-
-#include <SDL.h>
-
-AUD_SDLMixer::AUD_SDLMixer()
-{
-	m_factory = NULL;
-}
-
-AUD_SDLMixer::~AUD_SDLMixer()
-{
-	if(m_factory)
-	{
-		delete m_factory; AUD_DELETE("factory")
-	}
-}
-
-AUD_IReader* AUD_SDLMixer::prepare(AUD_IReader* reader)
-{
-	m_factory->setReader(reader);
-	return m_factory->createReader();
-}
-
-void AUD_SDLMixer::setSpecs(AUD_Specs specs)
-{
-	m_samplesize = AUD_SAMPLE_SIZE(specs);
-	if(m_factory)
-	{
-		delete m_factory; AUD_DELETE("factory")
-	}
-	m_factory = new AUD_SDLMixerFactory(specs); AUD_NEW("factory")
-}
-
-void AUD_SDLMixer::add(sample_t* buffer, AUD_Specs specs, int length,
-					   float volume)
-{
-	AUD_SDLMixerBuffer buf;
-	buf.buffer = buffer;
-	buf.length = length;
-	buf.volume = volume;
-	m_buffers.push_back(buf);
-}
-
-void AUD_SDLMixer::superpose(sample_t* buffer, int length, float volume)
-{
-	AUD_SDLMixerBuffer buf;
-
-	while(!m_buffers.empty())
-	{
-		buf = m_buffers.front();
-		m_buffers.pop_front();
-		SDL_MixAudio((Uint8*)buffer,
-					 (Uint8*)buf.buffer,
-					 buf.length * m_samplesize,
-					 (int)(SDL_MIX_MAXVOLUME * volume * buf.volume));
-	}
-}

intern/audaspace/SDL/AUD_SDLMixerReader.cpp

@@ -1,216 +0,0 @@
-/*
- * $Id$
- *
- * ***** BEGIN LGPL LICENSE BLOCK *****
- *
- * Copyright 2009 Jörg Hermann Müller
- *
- * This file is part of AudaSpace.
- *
- * AudaSpace is free software: you can redistribute it and/or modify
- * it under the terms of the GNU Lesser General Public License as published by
- * the Free Software Foundation, either version 3 of the License, or
- * (at your option) any later version.
- *
- * AudaSpace 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 Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public License
- * along with AudaSpace.  If not, see <http://www.gnu.org/licenses/>.
- *
- * ***** END LGPL LICENSE BLOCK *****
- */
-
-#include "AUD_SDLMixerReader.h"
-#include "AUD_Buffer.h"
-
-#include <cstring>
-
-inline Uint16 AUD_TO_SDL(AUD_SampleFormat format)
-{
-	// SDL only supports 8 and 16 bit audio
-	switch(format)
-	{
-	case AUD_FORMAT_U8:
-		return AUDIO_U8;
-	case AUD_FORMAT_S16:
-		return AUDIO_S16SYS;
-	default:
-		AUD_THROW(AUD_ERROR_SDL);
-	}
-}
-
-// greatest common divisor
-inline int gcd(int a, int b)
-{
-	int c;
-
-	// make sure a is the bigger
-	if(b > a)
-	{
-		c = b;
-		b = a;
-		a = c;
-	}
-
-	// greetings from Euclides
-	while(b != 0)
-	{
-		c = a % b;
-		a = b;
-		b = c;
-	}
-	return a;
-}
-
-AUD_SDLMixerReader::AUD_SDLMixerReader(AUD_IReader* reader,
-											 AUD_Specs specs)
-{
-	if(reader == NULL)
-		AUD_THROW(AUD_ERROR_READER);
-
-	m_reader = reader;
-	m_tspecs = specs;
-	m_sspecs = reader->getSpecs();
-
-	try
-	{
-		// SDL only supports 8 and 16 bit sample formats
-		if(SDL_BuildAudioCVT(&m_cvt,
-							 AUD_TO_SDL(m_sspecs.format),
-							 m_sspecs.channels,
-							 m_sspecs.rate,
-							 AUD_TO_SDL(specs.format),
-							 specs.channels,
-							 specs.rate) == -1)
-			AUD_THROW(AUD_ERROR_SDL);
-	}
-	catch(AUD_Exception)
-	{
-		delete m_reader; AUD_DELETE("reader")
-		throw;
-	}
-
-	m_eor = false;
-	m_rsposition = 0;
-	m_rssize = 0;
-	m_ssize = m_sspecs.rate / gcd(specs.rate, m_sspecs.rate);
-	m_tsize = m_tspecs.rate * m_ssize / m_sspecs.rate;
-
-	m_buffer = new AUD_Buffer(); AUD_NEW("buffer")
-	m_rsbuffer = new AUD_Buffer(); AUD_NEW("buffer")
-}
-
-AUD_SDLMixerReader::~AUD_SDLMixerReader()
-{
-	delete m_reader; AUD_DELETE("reader")
-	delete m_buffer; AUD_DELETE("buffer")
-	delete m_rsbuffer; AUD_DELETE("buffer")
-}
-
-bool AUD_SDLMixerReader::isSeekable()
-{
-	return m_reader->isSeekable();
-}
-
-void AUD_SDLMixerReader::seek(int position)
-{
-	m_reader->seek(position * m_ssize / m_tsize);
-	m_eor = false;
-}
-
-int AUD_SDLMixerReader::getLength()
-{
-	return m_reader->getLength() * m_tsize / m_ssize;
-}
-
-int AUD_SDLMixerReader::getPosition()
-{
-	return m_reader->getPosition() * m_tsize / m_ssize;
-}
-
-AUD_Specs AUD_SDLMixerReader::getSpecs()
-{
-	return m_tspecs;
-}
-
-AUD_ReaderType AUD_SDLMixerReader::getType()
-{
-	return m_reader->getType();
-}
-
-bool AUD_SDLMixerReader::notify(AUD_Message &message)
-{
-	return m_reader->notify(message);
-}
-
-void AUD_SDLMixerReader::read(int & length, sample_t* & buffer)
-{
-	// sample count for the target buffer without getting a shift
-	int tns = length + m_tsize - length % m_tsize;
-	// sample count for the source buffer without getting a shift
-	int sns = tns * m_ssize / m_tsize;
-	// target sample size
-	int tss = AUD_SAMPLE_SIZE(m_tspecs);
-	// source sample size
-	int sss = AUD_SAMPLE_SIZE(m_sspecs);
-
-	// input is output buffer
-	int buf_size = AUD_MAX(tns*tss, sns*sss);
-
-	// resize if necessary
-	if(m_rsbuffer->getSize() < buf_size)
-		m_rsbuffer->resize(buf_size, true);
-
-	if(m_buffer->getSize() < length*tss)
-		m_buffer->resize(length*tss);
-
-	buffer = m_buffer->getBuffer();
-	int size;
-	int index = 0;
-	sample_t* buf;
-
-	while(index < length)
-	{
-		if(m_rsposition == m_rssize)
-		{
-			// no more data
-			if(m_eor)
-				length = index;
-			// mix
-			else
-			{
-				// read from source
-				size = sns;
-				m_reader->read(size, buf);
-
-				// prepare
-				m_cvt.buf = m_rsbuffer->getBuffer();
-				m_cvt.len = size*sss;
-				memcpy(m_cvt.buf, buf, size*sss);
-
-				// convert
-				SDL_ConvertAudio(&m_cvt);
-
-				// end of reader
-				if(size < sns)
-					m_eor = true;
-
-				m_rsposition = 0;
-				m_rssize = size * m_tsize / m_ssize;
-			}
-		}
-
-		// size to copy
-		size = AUD_MIN(m_rssize-m_rsposition, length-index);
-
-		// copy
-		memcpy(m_buffer->getBuffer() + index * tss,
-			   m_rsbuffer->getBuffer() + m_rsposition * tss,
-			   size*tss);
-		m_rsposition += size;
-		index += size;
-	}
-}

intern/audaspace/SDL/AUD_SDLMixerReader.h

@@ -1,128 +0,0 @@
-/*
- * $Id$
- *
- * ***** BEGIN LGPL LICENSE BLOCK *****
- *
- * Copyright 2009 Jörg Hermann Müller
- *
- * This file is part of AudaSpace.
- *
- * AudaSpace is free software: you can redistribute it and/or modify
- * it under the terms of the GNU Lesser General Public License as published by
- * the Free Software Foundation, either version 3 of the License, or
- * (at your option) any later version.
- *
- * AudaSpace 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 Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public License
- * along with AudaSpace.  If not, see <http://www.gnu.org/licenses/>.
- *
- * ***** END LGPL LICENSE BLOCK *****
- */
-
-#ifndef AUD_SDLMIXERREADER
-#define AUD_SDLMIXERREADER
-
-#include "AUD_IReader.h"
-class AUD_Buffer;
-
-#include <SDL.h>
-
-/**
- * This class mixes a sound source with help of the SDL library.
- * Unfortunately SDL is only capable of 8 and 16 bit audio, mono and stereo, as
- * well as resampling only 2^n sample rate relationships where n is a natural
- * number.
- * \warning Although SDL can only resample 2^n sample rate relationships, this
- *          class doesn't check for compliance, so in case of other factors,
- *          the behaviour is undefined.
- */
-class AUD_SDLMixerReader : public AUD_IReader
-{
-private:
-	/**
-	 * The reader that is being mixed.
-	 */
-	AUD_IReader* m_reader;
-
-	/**
-	 * The current reading position in the resampling buffer.
-	 */
-	int m_rsposition;
-
-	/**
-	 * The count of mixed samples in the resampling buffer.
-	 */
-	int m_rssize;
-
-	/**
-	 * The smallest count of source samples to get a fractionless resampling
-	 * factor.
-	 */
-	int m_ssize;
-
-	/**
-	 * The smallest count of target samples to get a fractionless resampling
-	 * factor.
-	 */
-	int m_tsize;
-
-	/**
-	 * The sound output buffer.
-	 */
-	AUD_Buffer *m_buffer;
-
-	/**
-	 * The resampling buffer.
-	 */
-	AUD_Buffer *m_rsbuffer;
-
-	/**
-	 * The target specification.
-	 */
-	AUD_Specs m_tspecs;
-
-	/**
-	 * The sample specification of the source.
-	 */
-	AUD_Specs m_sspecs;
-
-	/**
-	 * Saves whether the end of the source has been reached.
-	 */
-	bool m_eor;
-
-	/**
-	 * The SDL_AudioCVT structure used for resampling.
-	 */
-	SDL_AudioCVT m_cvt;
-
-public:
-	/**
-	 * Creates a resampling reader.
-	 * \param reader The reader to mix.
-	 * \param specs The target specification.
-	 * \exception AUD_Exception Thrown if the source specification cannot be
-	 *            mixed to the target specification or if the reader is
-	 *            NULL.
-	 */
-	AUD_SDLMixerReader(AUD_IReader* reader, AUD_Specs specs);
-	/**
-	 * Destroys the reader.
-	 */
-	~AUD_SDLMixerReader();
-
-	virtual bool isSeekable();
-	virtual void seek(int position);
-	virtual int getLength();
-	virtual int getPosition();
-	virtual AUD_Specs getSpecs();
-	virtual AUD_ReaderType getType();
-	virtual bool notify(AUD_Message &message);
-	virtual void read(int & length, sample_t* & buffer);
-};
-
-#endif //AUD_SDLMIXERREADER

intern/audaspace/SRC/AUD_SRCResampleFactory.cpp

@@ -27,14 +27,14 @@
 #include "AUD_SRCResampleReader.h"
 
 AUD_SRCResampleFactory::AUD_SRCResampleFactory(AUD_IReader* reader,
-											   AUD_Specs specs) :
+											   AUD_DeviceSpecs specs) :
 		AUD_ResampleFactory(reader, specs) {}
 
 AUD_SRCResampleFactory::AUD_SRCResampleFactory(AUD_IFactory* factory,
-											   AUD_Specs specs) :
+											   AUD_DeviceSpecs specs) :
 		AUD_ResampleFactory(factory, specs) {}
 
-AUD_SRCResampleFactory::AUD_SRCResampleFactory(AUD_Specs specs) :
+AUD_SRCResampleFactory::AUD_SRCResampleFactory(AUD_DeviceSpecs specs) :
 		AUD_ResampleFactory(specs) {}
 
 AUD_IReader* AUD_SRCResampleFactory::createReader()
@@ -45,7 +45,7 @@ AUD_IReader* AUD_SRCResampleFactory::createReader()
 	{
 		if(reader->getSpecs().rate != m_specs.rate)
 		{
-			reader = new AUD_SRCResampleReader(reader, m_specs);
+			reader = new AUD_SRCResampleReader(reader, m_specs.specs);
 			AUD_NEW("reader")
 		}
 	}

intern/audaspace/SRC/AUD_SRCResampleFactory.h

@@ -31,14 +31,13 @@
 /**
  * This factory creates a resampling reader that uses libsamplerate for
  * resampling.
- * \note The format of the input must be float.
  */
 class AUD_SRCResampleFactory : public AUD_ResampleFactory
 {
 public:
-	AUD_SRCResampleFactory(AUD_IReader* reader, AUD_Specs specs);
-	AUD_SRCResampleFactory(AUD_IFactory* factory, AUD_Specs specs);
-	AUD_SRCResampleFactory(AUD_Specs specs);
+	AUD_SRCResampleFactory(AUD_IReader* reader, AUD_DeviceSpecs specs);
+	AUD_SRCResampleFactory(AUD_IFactory* factory, AUD_DeviceSpecs specs);
+	AUD_SRCResampleFactory(AUD_DeviceSpecs specs);
 
 	virtual AUD_IReader* createReader();
 };

intern/audaspace/SRC/AUD_SRCResampleReader.cpp

@@ -26,9 +26,9 @@
 #include "AUD_SRCResampleReader.h"
 #include "AUD_Buffer.h"
 
-#include <math.h>
+#include <cmath>
 #include <cstring>
-#include <stdio.h>
+#include <cstdio>
 
 static long src_callback(void *cb_data, float **data)
 {
@@ -41,15 +41,8 @@ AUD_SRCResampleReader::AUD_SRCResampleReader(AUD_IReader* reader,
 {
 	m_sspecs = reader->getSpecs();
 
-	if(m_sspecs.format != AUD_FORMAT_FLOAT32)
-	{
-		delete m_reader; AUD_DELETE("reader")
-		AUD_THROW(AUD_ERROR_READER);
-	}
-
 	m_tspecs = specs;
 	m_tspecs.channels = m_sspecs.channels;
-	m_tspecs.format = m_sspecs.format;
 	m_factor = (double)m_tspecs.rate / (double)m_sspecs.rate;
 
 	int error;
@@ -71,9 +64,9 @@ AUD_SRCResampleReader::AUD_SRCResampleReader(AUD_IReader* reader,
 
 AUD_SRCResampleReader::~AUD_SRCResampleReader()
 {
-	delete m_buffer; AUD_DELETE("buffer")
-
 	src_delete(m_src);
+
+	delete m_buffer; AUD_DELETE("buffer")
 }
 
 long AUD_SRCResampleReader::doCallback(float** data)
@@ -83,7 +76,7 @@ long AUD_SRCResampleReader::doCallback(float** data)
 
 	m_reader->read(length, buffer);
 
-	*data = (float*)buffer;
+	*data = buffer;
 	return length;
 }
 
@@ -110,10 +103,12 @@ AUD_Specs AUD_SRCResampleReader::getSpecs()
 
 void AUD_SRCResampleReader::read(int & length, sample_t* & buffer)
 {
-	if(m_buffer->getSize() < length * m_tspecs.channels * 4)
-		m_buffer->resize(length * m_tspecs.channels * 4);
+	int size = length * AUD_SAMPLE_SIZE(m_tspecs);
+
+	if(m_buffer->getSize() < size)
+		m_buffer->resize(size);
 
 	buffer = m_buffer->getBuffer();
 
-	length = src_callback_read(m_src, m_factor, length, (float*)buffer);
+	length = src_callback_read(m_src, m_factor, length, buffer);
 }

intern/audaspace/SRC/AUD_SRCResampleReader.h

@@ -32,13 +32,7 @@ class AUD_Buffer;
 #include <samplerate.h>
 
 /**
- * This class mixes a sound source with help of the SDL library.
- * Unfortunately SDL is only capable of 8 and 16 bit audio, mono and stereo, as
- * well as resampling only 2^n sample rate relationships where n is a natural
- * number.
- * \warning Although SDL can only resample 2^n sample rate relationships, this
- *          class doesn't check for compliance, so in case of other factors,
- *          the behaviour is undefined.
+ * This resampling reader uses libsamplerate for resampling.
  */
 class AUD_SRCResampleReader : public AUD_EffectReader
 {

intern/audaspace/ffmpeg/AUD_FFMPEGReader.cpp

@@ -34,26 +34,6 @@ extern "C" {
 #include <libavformat/avformat.h>
 }
 
-// This function transforms a FFMPEG SampleFormat to our own sample format
-static inline AUD_SampleFormat FFMPEG_TO_AUD(SampleFormat fmt)
-{
-	switch(fmt)
-	{
-	case SAMPLE_FMT_U8:
-		return AUD_FORMAT_U8;
-	case SAMPLE_FMT_S16:
-		return AUD_FORMAT_S16;
-	case SAMPLE_FMT_S32:
-		return AUD_FORMAT_S32;
-	case SAMPLE_FMT_FLT:
-		return AUD_FORMAT_FLOAT32;
-	case SAMPLE_FMT_DBL:
-		return AUD_FORMAT_FLOAT64;
-	default:
-		return AUD_FORMAT_INVALID;
-	}
-}
-
 int AUD_FFMPEGReader::decode(AVPacket* packet, AUD_Buffer* buffer)
 {
 	// save packet parameters
@@ -78,11 +58,11 @@ int AUD_FFMPEGReader::decode(AVPacket* packet, AUD_Buffer* buffer)
 		// read samples from the packet
 		data_size = buf_size - buf_pos;
 		/*read_length = avcodec_decode_audio3(m_codecCtx,
-			(int16_t*)(buffer->getBuffer()+buf_pos),
+			(int16_t*)(((data_t*)buffer->getBuffer())+buf_pos),
 			&data_size,
 			packet);*/
 		read_length = avcodec_decode_audio2(m_codecCtx,
-			(int16_t*)(buffer->getBuffer()+buf_pos),
+			(int16_t*)(((data_t*)buffer->getBuffer())+buf_pos),
 			&data_size,
 			audio_pkg_data,
 			audio_pkg_size);
@@ -101,10 +81,78 @@ int AUD_FFMPEGReader::decode(AVPacket* packet, AUD_Buffer* buffer)
 	return buf_pos;
 }
 
-AUD_FFMPEGReader::AUD_FFMPEGReader(const char* filename)
+void AUD_FFMPEGReader::init()
 {
 	m_position = 0;
 	m_pkgbuf_left = 0;
+
+	if(av_find_stream_info(m_formatCtx)<0)
+		AUD_THROW(AUD_ERROR_FFMPEG);
+
+	// find audio stream and codec
+	m_stream = -1;
+
+	for(unsigned int i = 0; i < m_formatCtx->nb_streams; i++)
+		if((m_formatCtx->streams[i]->codec->codec_type == CODEC_TYPE_AUDIO)
+			&& (m_stream < 0))
+		{
+			m_stream=i;
+			break;
+		}
+	if(m_stream == -1)
+		AUD_THROW(AUD_ERROR_FFMPEG);
+
+	m_codecCtx = m_formatCtx->streams[m_stream]->codec;
+
+	// get a decoder and open it
+	AVCodec *aCodec = avcodec_find_decoder(m_codecCtx->codec_id);
+	if(!aCodec)
+		AUD_THROW(AUD_ERROR_FFMPEG);
+
+	if(avcodec_open(m_codecCtx, aCodec)<0)
+		AUD_THROW(AUD_ERROR_FFMPEG);
+
+	// XXX this prints file information to stdout:
+	//dump_format(m_formatCtx, 0, NULL, 0);
+
+	m_specs.channels = (AUD_Channels) m_codecCtx->channels;
+
+	switch(m_codecCtx->sample_fmt)
+	{
+	case SAMPLE_FMT_U8:
+		m_convert = AUD_convert_u8_float;
+		m_specs.format = AUD_FORMAT_U8;
+		break;
+	case SAMPLE_FMT_S16:
+		m_convert = AUD_convert_s16_float;
+		m_specs.format = AUD_FORMAT_S16;
+		break;
+	case SAMPLE_FMT_S32:
+		m_convert = AUD_convert_s32_float;
+		m_specs.format = AUD_FORMAT_S32;
+		break;
+	case SAMPLE_FMT_FLT:
+		m_convert = AUD_convert_copy<float>;
+		m_specs.format = AUD_FORMAT_FLOAT32;
+		break;
+	case SAMPLE_FMT_DBL:
+		m_convert = AUD_convert_double_float;
+		m_specs.format = AUD_FORMAT_FLOAT64;
+		break;
+	default:
+		AUD_THROW(AUD_ERROR_FILE);
+	}
+
+	m_specs.rate = (AUD_SampleRate) m_codecCtx->sample_rate;
+
+	// last but not least if there hasn't been any error, create the buffers
+	m_buffer = new AUD_Buffer(); AUD_NEW("buffer")
+	m_pkgbuf = new AUD_Buffer(AVCODEC_MAX_AUDIO_FRAME_SIZE<<1);
+	AUD_NEW("buffer")
+}
+
+AUD_FFMPEGReader::AUD_FFMPEGReader(const char* filename)
+{
 	m_byteiocontext = NULL;
 
 	// open file
@@ -113,66 +161,28 @@ AUD_FFMPEGReader::AUD_FFMPEGReader(const char* filename)
 
 	try
 	{
-		if(av_find_stream_info(m_formatCtx)<0)
-			AUD_THROW(AUD_ERROR_FFMPEG);
-
-		// find audio stream and codec
-		m_stream = -1;
-
-		for(unsigned int i = 0; i < m_formatCtx->nb_streams; i++)
-			if((m_formatCtx->streams[i]->codec->codec_type == CODEC_TYPE_AUDIO)
-				&& (m_stream < 0))
-			{
-				m_stream=i;
-				break;
-			}
-		if(m_stream == -1)
-			AUD_THROW(AUD_ERROR_FFMPEG);
-
-		m_codecCtx = m_formatCtx->streams[m_stream]->codec;
-
-		// get a decoder and open it
-		AVCodec *aCodec = avcodec_find_decoder(m_codecCtx->codec_id);
-		if(!aCodec)
-			AUD_THROW(AUD_ERROR_FFMPEG);
-
-		if(avcodec_open(m_codecCtx, aCodec)<0)
-			AUD_THROW(AUD_ERROR_FFMPEG);
-
-		// XXX this prints file information to stdout:
-		//dump_format(m_formatCtx, 0, filename, 0);
-
-		m_specs.channels = (AUD_Channels) m_codecCtx->channels;
-		m_specs.format = FFMPEG_TO_AUD(m_codecCtx->sample_fmt);
-		m_specs.rate = (AUD_SampleRate) m_codecCtx->sample_rate;
+		init();
 	}
 	catch(AUD_Exception)
 	{
 		av_close_input_file(m_formatCtx);
 		throw;
 	}
-
-	// last but not least if there hasn't been any error, create the buffers
-	m_buffer = new AUD_Buffer(); AUD_NEW("buffer")
-	m_pkgbuf = new AUD_Buffer(AVCODEC_MAX_AUDIO_FRAME_SIZE<<1);
-	AUD_NEW("buffer")
 }
 
 AUD_FFMPEGReader::AUD_FFMPEGReader(AUD_Reference<AUD_Buffer> buffer)
 {
-	m_position = 0;
-	m_pkgbuf_left = 0;
 	m_byteiocontext = (ByteIOContext*)av_mallocz(sizeof(ByteIOContext));
 	AUD_NEW("byteiocontext")
 	m_membuffer = buffer;
 
-	if(init_put_byte(m_byteiocontext, buffer.get()->getBuffer(), buffer.get()->getSize(), 0,
-					 NULL, NULL, NULL, NULL) != 0)
+	if(init_put_byte(m_byteiocontext, (data_t*)buffer.get()->getBuffer(),
+					 buffer.get()->getSize(), 0, NULL, NULL, NULL, NULL) != 0)
 		AUD_THROW(AUD_ERROR_FILE);
 
 	AVProbeData probe_data;
 	probe_data.filename = "";
-	probe_data.buf = buffer.get()->getBuffer();
+	probe_data.buf = (data_t*)buffer.get()->getBuffer();
 	probe_data.buf_size = buffer.get()->getSize();
 	AVInputFormat* fmt = av_probe_input_format(&probe_data, 1);
 
@@ -182,38 +192,7 @@ AUD_FFMPEGReader::AUD_FFMPEGReader(AUD_Reference<AUD_Buffer> buffer)
 
 	try
 	{
-		if(av_find_stream_info(m_formatCtx)<0)
-			AUD_THROW(AUD_ERROR_FFMPEG);
-
-		// find audio stream and codec
-		m_stream = -1;
-
-		for(unsigned int i = 0; i < m_formatCtx->nb_streams; i++)
-			if((m_formatCtx->streams[i]->codec->codec_type == CODEC_TYPE_AUDIO)
-				&& (m_stream < 0))
-			{
-				m_stream=i;
-				break;
-			}
-		if(m_stream == -1)
-			AUD_THROW(AUD_ERROR_FFMPEG);
-
-		m_codecCtx = m_formatCtx->streams[m_stream]->codec;
-
-		// get a decoder and open it
-		AVCodec *aCodec = avcodec_find_decoder(m_codecCtx->codec_id);
-		if(!aCodec)
-			AUD_THROW(AUD_ERROR_FFMPEG);
-
-		if(avcodec_open(m_codecCtx, aCodec)<0)
-			AUD_THROW(AUD_ERROR_FFMPEG);
-
-		// XXX this prints stream information to stdout:
-		//dump_format(m_formatCtx, 0, NULL, 0);
-
-		m_specs.channels = (AUD_Channels) m_codecCtx->channels;
-		m_specs.format = FFMPEG_TO_AUD(m_codecCtx->sample_fmt);
-		m_specs.rate = (AUD_SampleRate) m_codecCtx->sample_rate;
+		init();
 	}
 	catch(AUD_Exception)
 	{
@@ -221,11 +200,6 @@ AUD_FFMPEGReader::AUD_FFMPEGReader(AUD_Reference<AUD_Buffer> buffer)
 		av_free(m_byteiocontext); AUD_DELETE("byteiocontext")
 		throw;
 	}
-
-	// last but not least if there hasn't been any error, create the buffers
-	m_buffer = new AUD_Buffer(); AUD_NEW("buffer")
-	m_pkgbuf = new AUD_Buffer(AVCODEC_MAX_AUDIO_FRAME_SIZE<<1);
-	AUD_NEW("buffer")
 }
 
 AUD_FFMPEGReader::~AUD_FFMPEGReader()
@@ -316,7 +290,7 @@ int AUD_FFMPEGReader::getPosition()
 
 AUD_Specs AUD_FFMPEGReader::getSpecs()
 {
-	return m_specs;
+	return m_specs.specs;
 }
 
 AUD_ReaderType AUD_FFMPEGReader::getType()
@@ -336,11 +310,11 @@ void AUD_FFMPEGReader::read(int & length, sample_t* & buffer)
 	int data_size = 0;
 	int pkgbuf_pos;
 	int left = length;
-	int sample_size = AUD_SAMPLE_SIZE(m_specs);
+	int sample_size = AUD_DEVICE_SAMPLE_SIZE(m_specs);
 
 	// resize output buffer if necessary
-	if(m_buffer->getSize() < length*sample_size)
-		m_buffer->resize(length*sample_size);
+	if(m_buffer->getSize() < length * AUD_SAMPLE_SIZE(m_specs))
+		m_buffer->resize(length * AUD_SAMPLE_SIZE(m_specs));
 
 	buffer = m_buffer->getBuffer();
 	pkgbuf_pos = m_pkgbuf_left;
@@ -350,8 +324,9 @@ void AUD_FFMPEGReader::read(int & length, sample_t* & buffer)
 	if(pkgbuf_pos > 0)
 	{
 		data_size = AUD_MIN(pkgbuf_pos, left * sample_size);
-		memcpy(buffer, m_pkgbuf->getBuffer(), data_size);
-		buffer += data_size;
+		m_convert((data_t*) buffer, (data_t*) m_pkgbuf->getBuffer(),
+				  data_size / AUD_FORMAT_SIZE(m_specs.format));
+		buffer += data_size / AUD_FORMAT_SIZE(m_specs.format);
 		left -= data_size/sample_size;
 	}
 
@@ -366,8 +341,9 @@ void AUD_FFMPEGReader::read(int & length, sample_t* & buffer)
 
 			// copy to output buffer
 			data_size = AUD_MIN(pkgbuf_pos, left * sample_size);
-			memcpy(buffer, m_pkgbuf->getBuffer(), data_size);
-			buffer += data_size;
+			m_convert((data_t*) buffer, (data_t*) m_pkgbuf->getBuffer(),
+					  data_size / AUD_FORMAT_SIZE(m_specs.format));
+			buffer += data_size / AUD_FORMAT_SIZE(m_specs.format);
 			left -= data_size/sample_size;
 		}
 		av_free_packet(&packet);
@@ -376,7 +352,8 @@ void AUD_FFMPEGReader::read(int & length, sample_t* & buffer)
 	if(pkgbuf_pos > data_size)
 	{
 		m_pkgbuf_left = pkgbuf_pos-data_size;
-		memmove(m_pkgbuf->getBuffer(), m_pkgbuf->getBuffer()+data_size,
+		memmove(m_pkgbuf->getBuffer(),
+				((data_t*)m_pkgbuf->getBuffer())+data_size,
 				pkgbuf_pos-data_size);
 	}
 

intern/audaspace/ffmpeg/AUD_FFMPEGReader.h

@@ -26,6 +26,7 @@
 #ifndef AUD_FFMPEGREADER
 #define AUD_FFMPEGREADER
 
+#include "AUD_ConverterFunctions.h"
 #include "AUD_IReader.h"
 #include "AUD_Reference.h"
 class AUD_Buffer;
@@ -59,7 +60,7 @@ private:
 	/**
 	 * The specification of the audio data.
 	 */
-	AUD_Specs m_specs;
+	AUD_DeviceSpecs m_specs;
 
 	/**
 	 * The buffer for package reading.
@@ -91,6 +92,11 @@ private:
 	 */
 	int m_stream;
 
+	/**
+	 * Converter function.
+	 */
+	AUD_convert_f m_convert;
+
 	/**
 	 * The memory file to read from, only saved to keep the buffer alive.
 	 */
@@ -104,6 +110,11 @@ private:
 	 */
 	int decode(AVPacket* packet, AUD_Buffer* buffer);
 
+	/**
+	 * Initializes the object.
+	 */
+	void init();
+
 public:
 	/**
 	 * Creates a new reader.

intern/audaspace/fftw/AUD_BandPassFactory.cpp

@@ -63,16 +63,8 @@ AUD_IReader* AUD_BandPassFactory::createReader()
 
 	if(reader != 0)
 	{
-		if(reader->getSpecs().format == AUD_FORMAT_FLOAT32)
-		{
-			reader = new AUD_BandPassReader(reader, m_low, m_high);
-			AUD_NEW("reader")
-		}
-		else
-		{
-			delete reader; AUD_DELETE("reader")
-			return 0;
-		}
+		reader = new AUD_BandPassReader(reader, m_low, m_high);
+		AUD_NEW("reader")
 	}
 
 	return reader;

intern/audaspace/fftw/AUD_BandPassReader.cpp

@@ -77,7 +77,6 @@ void AUD_BandPassReader::read(int & length, sample_t* & buffer)
 			}
 
 			m_length = length;
-			printf("WINDOW: %d\n", m_length);
 
 			if(m_length * sizeof(double) > m_in->getSize())
 			{
@@ -95,16 +94,15 @@ void AUD_BandPassReader::read(int & length, sample_t* & buffer)
 											  FFTW_ESTIMATE);
 		}
 
-		float* source = (float*) buffer;
 		double* target = (double*) m_in->getBuffer();
-		float* target2 = (float*) m_buffer->getBuffer();
+		sample_t* target2 = m_buffer->getBuffer();
 		fftw_complex* complex = (fftw_complex*) m_out->getBuffer();
 		float frequency;
 
 		for(int channel = 0; channel < specs.channels; channel++)
 		{
 			for(int i = 0; i < m_length; i++)
-				target[i] = source[i * specs.channels + channel];
+				target[i] = buffer[i * specs.channels + channel];
 
 			fftw_execute(m_forward);
 

intern/audaspace/intern/AUD_Buffer.cpp

@@ -27,14 +27,14 @@
 #include "AUD_Space.h"
 
 #include <cstring>
-#include <stdlib.h>
+#include <cstdlib>
 
 #define AUD_ALIGN(a) (a + 16 - ((long)a & 15))
 
 AUD_Buffer::AUD_Buffer(int size)
 {
 	m_size = size;
-	m_buffer = (sample_t*) malloc(size+16); AUD_NEW("buffer")
+	m_buffer = (data_t*) malloc(size+16); AUD_NEW("buffer")
 }
 
 AUD_Buffer::~AUD_Buffer()
@@ -44,7 +44,7 @@ AUD_Buffer::~AUD_Buffer()
 
 sample_t* AUD_Buffer::getBuffer()
 {
-	return AUD_ALIGN(m_buffer);
+	return (sample_t*) AUD_ALIGN(m_buffer);
 }
 
 int AUD_Buffer::getSize()
@@ -54,7 +54,7 @@ int AUD_Buffer::getSize()
 
 void AUD_Buffer::resize(int size, bool keep)
 {
-	sample_t* buffer = (sample_t*) malloc(size+16); AUD_NEW("buffer")
+	data_t* buffer = (data_t*) malloc(size+16); AUD_NEW("buffer")
 
 	// copy old data over if wanted
 	if(keep)

intern/audaspace/intern/AUD_Buffer.h

@@ -39,7 +39,7 @@ private:
 	int m_size;
 
 	/// The pointer to the buffer memory.
-	sample_t* m_buffer;
+	data_t* m_buffer;
 
 public:
 	/**

intern/audaspace/intern/AUD_BufferReader.cpp

@@ -79,11 +79,11 @@ void AUD_BufferReader::read(int & length, sample_t* & buffer)
 {
 	int sample_size = AUD_SAMPLE_SIZE(m_specs);
 
-	buffer = m_buffer.get()->getBuffer()+m_position*sample_size;
+	buffer = m_buffer.get()->getBuffer() + m_position * m_specs.channels;
 
 	// in case the end of the buffer is reached
-	if(m_buffer.get()->getSize() < (m_position+length)*sample_size)
-		length = m_buffer.get()->getSize()/sample_size-m_position;
+	if(m_buffer.get()->getSize() < (m_position + length) * sample_size)
+		length = m_buffer.get()->getSize() / sample_size - m_position;
 
 	if(length < 0)
 		length = 0;

intern/audaspace/intern/AUD_C-API.cpp

@@ -23,6 +23,9 @@
  * ***** END LGPL LICENSE BLOCK *****
  */
 
+#include <cstdlib>
+#include <cstring>
+
 #include "AUD_NULLDevice.h"
 #include "AUD_I3DDevice.h"
 #include "AUD_FileFactory.h"
@@ -32,13 +35,18 @@
 #include "AUD_PingPongFactory.h"
 #include "AUD_LoopFactory.h"
 #include "AUD_RectifyFactory.h"
+#include "AUD_EnvelopeFactory.h"
+#include "AUD_LinearResampleFactory.h"
+#include "AUD_LowpassFactory.h"
+#include "AUD_HighpassFactory.h"
+#include "AUD_ChannelMapperFactory.h"
+#include "AUD_Buffer.h"
 #include "AUD_ReadDevice.h"
 #include "AUD_SourceCaps.h"
 #include "AUD_IReader.h"
 
 #ifdef WITH_SDL
 #include "AUD_SDLDevice.h"
-#include "AUD_FloatMixer.h"
 #endif
 
 #ifdef WITH_OPENAL
@@ -55,7 +63,7 @@ extern "C" {
 }
 #endif
 
-#include <assert.h>
+#include <cassert>
 
 typedef AUD_IFactory AUD_Sound;
 typedef AUD_ReadDevice AUD_Device;
@@ -71,7 +79,7 @@ static AUD_IDevice* AUD_device = NULL;
 static int AUD_available_devices[4];
 static AUD_I3DDevice* AUD_3ddevice = NULL;
 
-int AUD_init(AUD_DeviceType device, AUD_Specs specs, int buffersize)
+int AUD_init(AUD_DeviceType device, AUD_DeviceSpecs specs, int buffersize)
 {
 #ifdef WITH_FFMPEG
 	av_register_all();
@@ -90,12 +98,8 @@ int AUD_init(AUD_DeviceType device, AUD_Specs specs, int buffersize)
 			break;
 #ifdef WITH_SDL
 		case AUD_SDL_DEVICE:
-			{
-				dev = new AUD_SDLDevice(specs, buffersize);
-				AUD_FloatMixer* mixer = new AUD_FloatMixer();
-				((AUD_SDLDevice*)dev)->setMixer(mixer);
-				break;
-			}
+			dev = new AUD_SDLDevice(specs, buffersize);
+			break;
 #endif
 #ifdef WITH_OPENAL
 		case AUD_OPENAL_DEVICE:
@@ -177,7 +181,6 @@ AUD_SoundInfo AUD_getInfo(AUD_Sound* sound)
 	else
 	{
 		info.specs.channels = AUD_CHANNELS_INVALID;
-		info.specs.format = AUD_FORMAT_INVALID;
 		info.specs.rate = AUD_RATE_INVALID;
 		info.length = 0.0;
 	}
@@ -519,7 +522,7 @@ int AUD_setSoundPitch(AUD_Handle* handle, float pitch)
 	return false;
 }
 
-AUD_Device* AUD_openReadDevice(AUD_Specs specs)
+AUD_Device* AUD_openReadDevice(AUD_DeviceSpecs specs)
 {
 	try
 	{
@@ -578,7 +581,7 @@ int AUD_setDeviceSoundVolume(AUD_Device* device, AUD_Handle* handle,
 	return false;
 }
 
-int AUD_readDevice(AUD_Device* device, sample_t* buffer, int length)
+int AUD_readDevice(AUD_Device* device, data_t* buffer, int length)
 {
 	assert(device);
 	assert(buffer);
@@ -605,3 +608,43 @@ void AUD_closeReadDevice(AUD_Device* device)
 	{
 	}
 }
+
+float* AUD_readSoundBuffer(const char* filename, float low, float high,
+						   float attack, float release, float threshold,
+						   int samplerate, int* length)
+{
+	AUD_Buffer buffer;
+	AUD_DeviceSpecs specs;
+	specs.channels = AUD_CHANNELS_MONO;
+	specs.rate = (AUD_SampleRate)samplerate;
+
+	AUD_FileFactory file(filename);
+	AUD_ChannelMapperFactory mapper(&file, specs);
+	AUD_LowpassFactory lowpass(&mapper, high);
+	AUD_HighpassFactory highpass(&lowpass, low);
+	AUD_EnvelopeFactory envelope(&highpass, attack, release, threshold, 0.1f);
+	AUD_LinearResampleFactory resampler(&envelope, specs);
+
+	AUD_IReader* reader = resampler.createReader();
+
+	if(reader == NULL)
+		return NULL;
+
+	int len;
+	int position = 0;
+	sample_t* readbuffer;
+	do
+	{
+		len = samplerate;
+		buffer.resize((position + len) * sizeof(float), true);
+		reader->read(len, readbuffer);
+		memcpy(buffer.getBuffer() + position, readbuffer, len * sizeof(float));
+		position += len;
+	} while(len != 0);
+	delete reader;
+
+	float* result = (float*)malloc(position * sizeof(float));
+	memcpy(result, buffer.getBuffer(), position * sizeof(float));
+	*length = position;
+	return result;
+}

intern/audaspace/intern/AUD_C-API.h

@@ -59,7 +59,7 @@ typedef struct
  * \param buffersize The buffersize for the device.
  * \return Whether the device has been initialized.
  */
-extern int AUD_init(AUD_DeviceType device, AUD_Specs specs, int buffersize);
+extern int AUD_init(AUD_DeviceType device, AUD_DeviceSpecs specs, int buffersize);
 
 /**
  * Returns a integer list with available sound devices. The last one is always
@@ -304,7 +304,7 @@ extern int AUD_setSoundPitch(AUD_Handle* handle, float pitch);
  * \param specs The specification of the audio data.
  * \return A device handle.
  */
-extern AUD_Device* AUD_openReadDevice(AUD_Specs specs);
+extern AUD_Device* AUD_openReadDevice(AUD_DeviceSpecs specs);
 
 /**
  * Sets the main volume of a device.
@@ -342,7 +342,7 @@ extern int AUD_setDeviceSoundVolume(AUD_Device* device,
  *         played back currently, in that case the buffer is filled with
  *         silence.
  */
-extern int AUD_readDevice(AUD_Device* device, sample_t* buffer, int length);
+extern int AUD_readDevice(AUD_Device* device, data_t* buffer, int length);
 
 /**
  * Closes a read device.
@@ -350,6 +350,14 @@ extern int AUD_readDevice(AUD_Device* device, sample_t* buffer, int length);
  */
 extern void AUD_closeReadDevice(AUD_Device* device);
 
+/**
+ * Reads a sound file into a newly created float buffer.
+ * The sound is therefore bandpassed, rectified and resampled.
+ */
+extern float* AUD_readSoundBuffer(const char* filename, float low, float high,
+								  float attack, float release, float threshold,
+								  int samplerate, int* length);
+
 #ifdef __cplusplus
 }
 #endif

intern/audaspace/intern/AUD_ChannelMapperFactory.cpp

@@ -29,20 +29,20 @@
 #include <cstring>
 
 AUD_ChannelMapperFactory::AUD_ChannelMapperFactory(AUD_IReader* reader,
-										   AUD_Specs specs) :
+												   AUD_DeviceSpecs specs) :
 		AUD_MixerFactory(reader, specs)
 {
 	memset(m_mapping, 0, sizeof(m_mapping));
 }
 
 AUD_ChannelMapperFactory::AUD_ChannelMapperFactory(AUD_IFactory* factory,
-										   AUD_Specs specs) :
+												   AUD_DeviceSpecs specs) :
 		AUD_MixerFactory(factory, specs)
 {
 	memset(m_mapping, 0, sizeof(m_mapping));
 }
 
-AUD_ChannelMapperFactory::AUD_ChannelMapperFactory(AUD_Specs specs) :
+AUD_ChannelMapperFactory::AUD_ChannelMapperFactory(AUD_DeviceSpecs specs) :
 		AUD_MixerFactory(specs)
 {
 	memset(m_mapping, 0, sizeof(m_mapping));

intern/audaspace/intern/AUD_ChannelMapperFactory.h

@@ -41,9 +41,9 @@ private:
 	float **m_mapping[9];
 
 public:
-	AUD_ChannelMapperFactory(AUD_IReader* reader, AUD_Specs specs);
-	AUD_ChannelMapperFactory(AUD_IFactory* factory, AUD_Specs specs);
-	AUD_ChannelMapperFactory(AUD_Specs specs);
+	AUD_ChannelMapperFactory(AUD_IReader* reader, AUD_DeviceSpecs specs);
+	AUD_ChannelMapperFactory(AUD_IFactory* factory, AUD_DeviceSpecs specs);
+	AUD_ChannelMapperFactory(AUD_DeviceSpecs specs);
 
 	virtual ~AUD_ChannelMapperFactory();
 

intern/audaspace/intern/AUD_ChannelMapperReader.cpp

@@ -32,12 +32,6 @@ AUD_ChannelMapperReader::AUD_ChannelMapperReader(AUD_IReader* reader,
 {
 	m_specs = reader->getSpecs();
 
-	if(m_specs.format != AUD_FORMAT_FLOAT32)
-	{
-		delete m_reader; AUD_DELETE("reader")
-		AUD_THROW(AUD_ERROR_READER);
-	}
-
 	int channels = -1;
 	m_rch = m_specs.channels;
 	while(mapping[++channels] != 0);
@@ -89,9 +83,9 @@ void AUD_ChannelMapperReader::read(int & length, sample_t* & buffer)
 	if(m_buffer->getSize() < length * 4 * channels)
 		m_buffer->resize(length * 4 * channels);
 
-	float* in = (float*)buffer;
-	float* out = (float*)m_buffer->getBuffer();
-	float sum;
+	sample_t* in = buffer;
+	sample_t* out = m_buffer->getBuffer();
+	sample_t sum;
 
 	for(int i = 0; i < length; i++)
 	{

intern/audaspace/intern/AUD_ConverterFactory.cpp

@@ -27,14 +27,14 @@
 #include "AUD_ConverterReader.h"
 
 AUD_ConverterFactory::AUD_ConverterFactory(AUD_IReader* reader,
-										   AUD_Specs specs) :
+										   AUD_DeviceSpecs specs) :
 		AUD_MixerFactory(reader, specs) {}
 
 AUD_ConverterFactory::AUD_ConverterFactory(AUD_IFactory* factory,
-										   AUD_Specs specs) :
+										   AUD_DeviceSpecs specs) :
 		AUD_MixerFactory(factory, specs) {}
 
-AUD_ConverterFactory::AUD_ConverterFactory(AUD_Specs specs) :
+AUD_ConverterFactory::AUD_ConverterFactory(AUD_DeviceSpecs specs) :
 		AUD_MixerFactory(specs) {}
 
 AUD_IReader* AUD_ConverterFactory::createReader()
@@ -43,7 +43,7 @@ AUD_IReader* AUD_ConverterFactory::createReader()
 
 	if(reader != 0)
 	{
-		if(reader->getSpecs().format != m_specs.format)
+		if(m_specs.format != AUD_FORMAT_FLOAT32)
 		{
 			reader = new AUD_ConverterReader(reader, m_specs);
 			AUD_NEW("reader")

intern/audaspace/intern/AUD_ConverterFactory.h

@@ -35,9 +35,9 @@
 class AUD_ConverterFactory : public AUD_MixerFactory
 {
 public:
-	AUD_ConverterFactory(AUD_IReader* reader, AUD_Specs specs);
-	AUD_ConverterFactory(AUD_IFactory* factory, AUD_Specs specs);
-	AUD_ConverterFactory(AUD_Specs specs);
+	AUD_ConverterFactory(AUD_IReader* reader, AUD_DeviceSpecs specs);
+	AUD_ConverterFactory(AUD_IFactory* factory, AUD_DeviceSpecs specs);
+	AUD_ConverterFactory(AUD_DeviceSpecs specs);
 
 	virtual AUD_IReader* createReader();
 };

intern/audaspace/intern/AUD_ConverterFunctions.cpp

@@ -36,14 +36,14 @@
 #define AUD_FLT_MAX		1.0
 #define AUD_FLT_MIN		-1.0
 
-void AUD_convert_u8_s16(sample_t* target, sample_t* source, int length)
+void AUD_convert_u8_s16(data_t* target, data_t* source, int length)
 {
 	int16_t* t = (int16_t*) target;
 	for(int i = 0; i < length; i++)
 		t[i] = (((int16_t)source[i]) - AUD_U8_0) << 8;
 }
 
-void AUD_convert_u8_s24_be(sample_t* target, sample_t* source, int length)
+void AUD_convert_u8_s24_be(data_t* target, data_t* source, int length)
 {
 	for(int i = 0; i < length; i++)
 	{
@@ -53,7 +53,7 @@ void AUD_convert_u8_s24_be(sample_t* target, sample_t* source, int length)
 	}
 }
 
-void AUD_convert_u8_s24_le(sample_t* target, sample_t* source, int length)
+void AUD_convert_u8_s24_le(data_t* target, data_t* source, int length)
 {
 	for(int i = 0; i < length; i++)
 	{
@@ -63,35 +63,35 @@ void AUD_convert_u8_s24_le(sample_t* target, sample_t* source, int length)
 	}
 }
 
-void AUD_convert_u8_s32(sample_t* target, sample_t* source, int length)
+void AUD_convert_u8_s32(data_t* target, data_t* source, int length)
 {
 	int32_t* t = (int32_t*) target;
 	for(int i = 0; i < length; i++)
 		t[i] = (((int32_t)source[i]) - AUD_U8_0) << 24;
 }
 
-void AUD_convert_u8_float(sample_t* target, sample_t* source, int length)
+void AUD_convert_u8_float(data_t* target, data_t* source, int length)
 {
 	float* t = (float*) target;
 	for(int i = 0; i < length; i++)
 		t[i] = (((int32_t)source[i]) - AUD_U8_0) / ((float)AUD_U8_0);
 }
 
-void AUD_convert_u8_double(sample_t* target, sample_t* source, int length)
+void AUD_convert_u8_double(data_t* target, data_t* source, int length)
 {
 	double* t = (double*) target;
 	for(int i = 0; i < length; i++)
 		t[i] = (((int32_t)source[i]) - AUD_U8_0) / ((double)AUD_U8_0);
 }
 
-void AUD_convert_s16_u8(sample_t* target, sample_t* source, int length)
+void AUD_convert_s16_u8(data_t* target, data_t* source, int length)
 {
 	int16_t* s = (int16_t*) source;
 	for(int i = 0; i < length; i++)
 		target[i] = (unsigned char)((s[i] >> 8) + AUD_U8_0);
 }
 
-void AUD_convert_s16_s24_be(sample_t* target, sample_t* source, int length)
+void AUD_convert_s16_s24_be(data_t* target, data_t* source, int length)
 {
 	int16_t* s = (int16_t*) source;
 	for(int i = 0; i < length; i++)
@@ -102,7 +102,7 @@ void AUD_convert_s16_s24_be(sample_t* target, sample_t* source, int length)
 	}
 }
 
-void AUD_convert_s16_s24_le(sample_t* target, sample_t* source, int length)
+void AUD_convert_s16_s24_le(data_t* target, data_t* source, int length)
 {
 	int16_t* s = (int16_t*) source;
 	for(int i = 0; i < length; i++)
@@ -113,7 +113,7 @@ void AUD_convert_s16_s24_le(sample_t* target, sample_t* source, int length)
 	}
 }
 
-void AUD_convert_s16_s32(sample_t* target, sample_t* source, int length)
+void AUD_convert_s16_s32(data_t* target, data_t* source, int length)
 {
 	int16_t* s = (int16_t*) source;
 	int32_t* t = (int32_t*) target;
@@ -121,7 +121,7 @@ void AUD_convert_s16_s32(sample_t* target, sample_t* source, int length)
 		t[i] = ((int32_t)s[i]) << 16;
 }
 
-void AUD_convert_s16_float(sample_t* target, sample_t* source, int length)
+void AUD_convert_s16_float(data_t* target, data_t* source, int length)
 {
 	int16_t* s = (int16_t*) source;
 	float* t = (float*) target;
@@ -129,7 +129,7 @@ void AUD_convert_s16_float(sample_t* target, sample_t* source, int length)
 		t[i] = s[i] / AUD_S16_FLT;
 }
 
-void AUD_convert_s16_double(sample_t* target, sample_t* source, int length)
+void AUD_convert_s16_double(data_t* target, data_t* source, int length)
 {
 	int16_t* s = (int16_t*) source;
 	double* t = (double*) target;
@@ -137,52 +137,52 @@ void AUD_convert_s16_double(sample_t* target, sample_t* source, int length)
 		t[i] = s[i] / AUD_S16_FLT;
 }
 
-void AUD_convert_s24_u8_be(sample_t* target, sample_t* source, int length)
+void AUD_convert_s24_u8_be(data_t* target, data_t* source, int length)
 {
 	for(int i = 0; i < length; i++)
 		target[i] = source[i*3] ^ AUD_U8_0;
 }
 
-void AUD_convert_s24_u8_le(sample_t* target, sample_t* source, int length)
+void AUD_convert_s24_u8_le(data_t* target, data_t* source, int length)
 {
 	for(int i = 0; i < length; i++)
 		target[i] = source[i*3+2] ^ AUD_U8_0;
 }
 
-void AUD_convert_s24_s16_be(sample_t* target, sample_t* source, int length)
+void AUD_convert_s24_s16_be(data_t* target, data_t* source, int length)
 {
 	int16_t* t = (int16_t*) target;
 	for(int i = 0; i < length; i++)
 		t[i] = source[i*3] << 8 | source[i*3+1];
 }
 
-void AUD_convert_s24_s16_le(sample_t* target, sample_t* source, int length)
+void AUD_convert_s24_s16_le(data_t* target, data_t* source, int length)
 {
 	int16_t* t = (int16_t*) target;
 	for(int i = 0; i < length; i++)
 		t[i] = source[i*3+2] << 8 | source[i*3+1];
 }
 
-void AUD_convert_s24_s24(sample_t* target, sample_t* source, int length)
+void AUD_convert_s24_s24(data_t* target, data_t* source, int length)
 {
 	memcpy(target, source, length * 3);
 }
 
-void AUD_convert_s24_s32_be(sample_t* target, sample_t* source, int length)
+void AUD_convert_s24_s32_be(data_t* target, data_t* source, int length)
 {
 	int32_t* t = (int32_t*) target;
 	for(int i = 0; i < length; i++)
 		t[i] = source[i*3] << 24 | source[i*3+1] << 16 | source[i*3+2] << 8;
 }
 
-void AUD_convert_s24_s32_le(sample_t* target, sample_t* source, int length)
+void AUD_convert_s24_s32_le(data_t* target, data_t* source, int length)
 {
 	int32_t* t = (int32_t*) target;
 	for(int i = 0; i < length; i++)
 		t[i] = source[i*3+2] << 24 | source[i*3+1] << 16 | source[i*3] << 8;
 }
 
-void AUD_convert_s24_float_be(sample_t* target, sample_t* source, int length)
+void AUD_convert_s24_float_be(data_t* target, data_t* source, int length)
 {
 	float* t = (float*) target;
 	int32_t s;
@@ -193,7 +193,7 @@ void AUD_convert_s24_float_be(sample_t* target, sample_t* source, int length)
 	}
 }
 
-void AUD_convert_s24_float_le(sample_t* target, sample_t* source, int length)
+void AUD_convert_s24_float_le(data_t* target, data_t* source, int length)
 {
 	float* t = (float*) target;
 	int32_t s;
@@ -204,7 +204,7 @@ void AUD_convert_s24_float_le(sample_t* target, sample_t* source, int length)
 	}
 }
 
-void AUD_convert_s24_double_be(sample_t* target, sample_t* source, int length)
+void AUD_convert_s24_double_be(data_t* target, data_t* source, int length)
 {
 	double* t = (double*) target;
 	int32_t s;
@@ -215,7 +215,7 @@ void AUD_convert_s24_double_be(sample_t* target, sample_t* source, int length)
 	}
 }
 
-void AUD_convert_s24_double_le(sample_t* target, sample_t* source, int length)
+void AUD_convert_s24_double_le(data_t* target, data_t* source, int length)
 {
 	double* t = (double*) target;
 	int32_t s;
@@ -226,14 +226,14 @@ void AUD_convert_s24_double_le(sample_t* target, sample_t* source, int length)
 	}
 }
 
-void AUD_convert_s32_u8(sample_t* target, sample_t* source, int length)
+void AUD_convert_s32_u8(data_t* target, data_t* source, int length)
 {
 	int16_t* s = (int16_t*) source;
 	for(int i = 0; i < length; i++)
 		target[i] = (unsigned char)((s[i] >> 24) + AUD_U8_0);
 }
 
-void AUD_convert_s32_s16(sample_t* target, sample_t* source, int length)
+void AUD_convert_s32_s16(data_t* target, data_t* source, int length)
 {
 	int16_t* t = (int16_t*) target;
 	int32_t* s = (int32_t*) source;
@@ -241,7 +241,7 @@ void AUD_convert_s32_s16(sample_t* target, sample_t* source, int length)
 		t[i] = s[i] >> 16;
 }
 
-void AUD_convert_s32_s24_be(sample_t* target, sample_t* source, int length)
+void AUD_convert_s32_s24_be(data_t* target, data_t* source, int length)
 {
 	int32_t* s = (int32_t*) source;
 	for(int i = 0; i < length; i++)
@@ -252,7 +252,7 @@ void AUD_convert_s32_s24_be(sample_t* target, sample_t* source, int length)
 	}
 }
 
-void AUD_convert_s32_s24_le(sample_t* target, sample_t* source, int length)
+void AUD_convert_s32_s24_le(data_t* target, data_t* source, int length)
 {
 	int16_t* s = (int16_t*) source;
 	for(int i = 0; i < length; i++)
@@ -263,7 +263,7 @@ void AUD_convert_s32_s24_le(sample_t* target, sample_t* source, int length)
 	}
 }
 
-void AUD_convert_s32_float(sample_t* target, sample_t* source, int length)
+void AUD_convert_s32_float(data_t* target, data_t* source, int length)
 {
 	int32_t* s = (int32_t*) source;
 	float* t = (float*) target;
@@ -271,7 +271,7 @@ void AUD_convert_s32_float(sample_t* target, sample_t* source, int length)
 		t[i] = s[i] / AUD_S32_FLT;
 }
 
-void AUD_convert_s32_double(sample_t* target, sample_t* source, int length)
+void AUD_convert_s32_double(data_t* target, data_t* source, int length)
 {
 	int32_t* s = (int32_t*) source;
 	double* t = (double*) target;
@@ -279,7 +279,7 @@ void AUD_convert_s32_double(sample_t* target, sample_t* source, int length)
 		t[i] = s[i] / AUD_S32_FLT;
 }
 
-void AUD_convert_float_u8(sample_t* target, sample_t* source, int length)
+void AUD_convert_float_u8(data_t* target, data_t* source, int length)
 {
 	float* s = (float*) source;
 	float t;
@@ -295,7 +295,7 @@ void AUD_convert_float_u8(sample_t* target, sample_t* source, int length)
 	}
 }
 
-void AUD_convert_float_s16(sample_t* target, sample_t* source, int length)
+void AUD_convert_float_s16(data_t* target, data_t* source, int length)
 {
 	int16_t* t = (int16_t*) target;
 	float* s = (float*) source;
@@ -310,7 +310,7 @@ void AUD_convert_float_s16(sample_t* target, sample_t* source, int length)
 	}
 }
 
-void AUD_convert_float_s24_be(sample_t* target, sample_t* source, int length)
+void AUD_convert_float_s24_be(data_t* target, data_t* source, int length)
 {
 	int32_t t;
 	float* s = (float*) source;
@@ -328,7 +328,7 @@ void AUD_convert_float_s24_be(sample_t* target, sample_t* source, int length)
 	}
 }
 
-void AUD_convert_float_s24_le(sample_t* target, sample_t* source, int length)
+void AUD_convert_float_s24_le(data_t* target, data_t* source, int length)
 {
 	int32_t t;
 	float* s = (float*) source;
@@ -346,7 +346,7 @@ void AUD_convert_float_s24_le(sample_t* target, sample_t* source, int length)
 	}
 }
 
-void AUD_convert_float_s32(sample_t* target, sample_t* source, int length)
+void AUD_convert_float_s32(data_t* target, data_t* source, int length)
 {
 	int32_t* t = (int32_t*) target;
 	float* s = (float*) source;
@@ -361,7 +361,7 @@ void AUD_convert_float_s32(sample_t* target, sample_t* source, int length)
 	}
 }
 
-void AUD_convert_float_double(sample_t* target, sample_t* source, int length)
+void AUD_convert_float_double(data_t* target, data_t* source, int length)
 {
 	float* s = (float*) source;
 	double* t = (double*) target;
@@ -369,7 +369,7 @@ void AUD_convert_float_double(sample_t* target, sample_t* source, int length)
 		t[i] = s[i];
 }
 
-void AUD_convert_double_u8(sample_t* target, sample_t* source, int length)
+void AUD_convert_double_u8(data_t* target, data_t* source, int length)
 {
 	double* s = (double*) source;
 	double t;
@@ -385,7 +385,7 @@ void AUD_convert_double_u8(sample_t* target, sample_t* source, int length)
 	}
 }
 
-void AUD_convert_double_s16(sample_t* target, sample_t* source, int length)
+void AUD_convert_double_s16(data_t* target, data_t* source, int length)
 {
 	int16_t* t = (int16_t*) target;
 	double* s = (double*) source;
@@ -400,7 +400,7 @@ void AUD_convert_double_s16(sample_t* target, sample_t* source, int length)
 	}
 }
 
-void AUD_convert_double_s24_be(sample_t* target, sample_t* source, int length)
+void AUD_convert_double_s24_be(data_t* target, data_t* source, int length)
 {
 	int32_t t;
 	double* s = (double*) source;
@@ -418,7 +418,7 @@ void AUD_convert_double_s24_be(sample_t* target, sample_t* source, int length)
 	}
 }
 
-void AUD_convert_double_s24_le(sample_t* target, sample_t* source, int length)
+void AUD_convert_double_s24_le(data_t* target, data_t* source, int length)
 {
 	int32_t t;
 	double* s = (double*) source;
@@ -436,7 +436,7 @@ void AUD_convert_double_s24_le(sample_t* target, sample_t* source, int length)
 	}
 }
 
-void AUD_convert_double_s32(sample_t* target, sample_t* source, int length)
+void AUD_convert_double_s32(data_t* target, data_t* source, int length)
 {
 	int32_t* t = (int32_t*) target;
 	double* s = (double*) source;
@@ -451,92 +451,10 @@ void AUD_convert_double_s32(sample_t* target, sample_t* source, int length)
 	}
 }
 
-void AUD_convert_double_float(sample_t* target, sample_t* source, int length)
+void AUD_convert_double_float(data_t* target, data_t* source, int length)
 {
 	double* s = (double*) source;
 	float* t = (float*) target;
 	for(int i = 0; i < length; i++)
 		t[i] = s[i];
 }
-
-void AUD_volume_adjust_u8(sample_t* target, sample_t* source,
-						  int count, float volume)
-{
-	for(int i=0; i<count; i++)
-		target[i] = (unsigned char)((source[i]-0x0080) * volume + 0x80);
-}
-
-void AUD_volume_adjust_s24_le(sample_t* target, sample_t* source,
-							  int count, float volume)
-{
-	count *= 3;
-	int value;
-
-	for(int i=0; i<count; i+=3)
-	{
-		value = source[i+2] << 16 | source[i+1] << 8 | source[i];
-		value |= (((value & 0x800000) >> 23) * 255) << 24;
-		value *= volume;
-		target[i+2] = value >> 16;
-		target[i+1] = value >> 8;
-		target[i] = value;
-	}
-}
-
-void AUD_volume_adjust_s24_be(sample_t* target, sample_t* source,
-							  int count, float volume)
-{
-	count *= 3;
-	int value;
-
-	for(int i=0; i < count; i+=3)
-	{
-		value = source[i] << 16 | source[i+1] << 8 | source[i+2];
-		value |= (((value & 0x800000) >> 23) * 255) << 24;
-		value *= volume;
-		target[i] = value >> 16;
-		target[i+1] = value >> 8;
-		target[i+2] = value;
-	}
-}
-
-void AUD_rectify_u8(sample_t* target, sample_t* source, int count)
-{
-	for(int i=0; i<count; i++)
-		target[i] = source[i] < 0x80 ? 0x0100 - source[i] : source[i];
-}
-
-void AUD_rectify_s24_le(sample_t* target, sample_t* source, int count)
-{
-	count *= 3;
-	int value;
-
-	for(int i=0; i<count; i+=3)
-	{
-		value = source[i+2] << 16 | source[i+1] << 8 | source[i];
-		value |= (((value & 0x800000) >> 23) * 255) << 24;
-		if(value < 0)
-			value = -value;
-		target[i+2] = value >> 16;
-		target[i+1] = value >> 8;
-		target[i] = value;
-	}
-}
-
-void AUD_rectify_s24_be(sample_t* target, sample_t* source, int count)
-{
-	count *= 3;
-	int value;
-
-	for(int i=0; i < count; i+=3)
-	{
-		value = source[i] << 16 | source[i+1] << 8 | source[i+2];
-		value |= (((value & 0x800000) >> 23) * 255) << 24;
-		if(value < 0)
-			value = -value;
-		target[i] = value >> 16;
-		target[i+1] = value >> 8;
-		target[i+2] = value;
-	}
-}
-

intern/audaspace/intern/AUD_ConverterFunctions.h

@@ -41,133 +41,94 @@
 #include <stdint.h>
 #endif
 
-typedef void (*AUD_convert_f)(sample_t* target, sample_t* source, int length);
-
-typedef void (*AUD_volume_adjust_f)(sample_t* target, sample_t* source,
-									int count, float volume);
-
-typedef void (*AUD_rectify_f)(sample_t* target, sample_t* source, int count);
+typedef void (*AUD_convert_f)(data_t* target, data_t* source, int length);
 
 template <class T>
-void AUD_convert_copy(sample_t* target, sample_t* source, int length)
+void AUD_convert_copy(data_t* target, data_t* source, int length)
 {
 	memcpy(target, source, length*sizeof(T));
 }
 
-void AUD_convert_u8_s16(sample_t* target, sample_t* source, int length);
+void AUD_convert_u8_s16(data_t* target, data_t* source, int length);
 
-void AUD_convert_u8_s24_be(sample_t* target, sample_t* source, int length);
+void AUD_convert_u8_s24_be(data_t* target, data_t* source, int length);
 
-void AUD_convert_u8_s24_le(sample_t* target, sample_t* source, int length);
+void AUD_convert_u8_s24_le(data_t* target, data_t* source, int length);
 
-void AUD_convert_u8_s32(sample_t* target, sample_t* source, int length);
+void AUD_convert_u8_s32(data_t* target, data_t* source, int length);
 
-void AUD_convert_u8_float(sample_t* target, sample_t* source, int length);
+void AUD_convert_u8_float(data_t* target, data_t* source, int length);
 
-void AUD_convert_u8_double(sample_t* target, sample_t* source, int length);
+void AUD_convert_u8_double(data_t* target, data_t* source, int length);
 
-void AUD_convert_s16_u8(sample_t* target, sample_t* source, int length);
+void AUD_convert_s16_u8(data_t* target, data_t* source, int length);
 
-void AUD_convert_s16_s24_be(sample_t* target, sample_t* source, int length);
+void AUD_convert_s16_s24_be(data_t* target, data_t* source, int length);
 
-void AUD_convert_s16_s24_le(sample_t* target, sample_t* source, int length);
+void AUD_convert_s16_s24_le(data_t* target, data_t* source, int length);
 
-void AUD_convert_s16_s32(sample_t* target, sample_t* source, int length);
+void AUD_convert_s16_s32(data_t* target, data_t* source, int length);
 
-void AUD_convert_s16_float(sample_t* target, sample_t* source, int length);
+void AUD_convert_s16_float(data_t* target, data_t* source, int length);
 
-void AUD_convert_s16_double(sample_t* target, sample_t* source, int length);
+void AUD_convert_s16_double(data_t* target, data_t* source, int length);
 
-void AUD_convert_s24_u8_be(sample_t* target, sample_t* source, int length);
+void AUD_convert_s24_u8_be(data_t* target, data_t* source, int length);
 
-void AUD_convert_s24_u8_le(sample_t* target, sample_t* source, int length);
+void AUD_convert_s24_u8_le(data_t* target, data_t* source, int length);
 
-void AUD_convert_s24_s16_be(sample_t* target, sample_t* source, int length);
+void AUD_convert_s24_s16_be(data_t* target, data_t* source, int length);
 
-void AUD_convert_s24_s16_le(sample_t* target, sample_t* source, int length);
+void AUD_convert_s24_s16_le(data_t* target, data_t* source, int length);
 
-void AUD_convert_s24_s24(sample_t* target, sample_t* source, int length);
+void AUD_convert_s24_s24(data_t* target, data_t* source, int length);
 
-void AUD_convert_s24_s32_be(sample_t* target, sample_t* source, int length);
+void AUD_convert_s24_s32_be(data_t* target, data_t* source, int length);
 
-void AUD_convert_s24_s32_le(sample_t* target, sample_t* source, int length);
+void AUD_convert_s24_s32_le(data_t* target, data_t* source, int length);
 
-void AUD_convert_s24_float_be(sample_t* target, sample_t* source, int length);
+void AUD_convert_s24_float_be(data_t* target, data_t* source, int length);
 
-void AUD_convert_s24_float_le(sample_t* target, sample_t* source, int length);
+void AUD_convert_s24_float_le(data_t* target, data_t* source, int length);
 
-void AUD_convert_s24_double_be(sample_t* target, sample_t* source, int length);
+void AUD_convert_s24_double_be(data_t* target, data_t* source, int length);
 
-void AUD_convert_s24_double_le(sample_t* target, sample_t* source, int length);
+void AUD_convert_s24_double_le(data_t* target, data_t* source, int length);
 
-void AUD_convert_s32_u8(sample_t* target, sample_t* source, int length);
+void AUD_convert_s32_u8(data_t* target, data_t* source, int length);
 
-void AUD_convert_s32_s16(sample_t* target, sample_t* source, int length);
+void AUD_convert_s32_s16(data_t* target, data_t* source, int length);
 
-void AUD_convert_s32_s24_be(sample_t* target, sample_t* source, int length);
+void AUD_convert_s32_s24_be(data_t* target, data_t* source, int length);
 
-void AUD_convert_s32_s24_le(sample_t* target, sample_t* source, int length);
+void AUD_convert_s32_s24_le(data_t* target, data_t* source, int length);
 
-void AUD_convert_s32_float(sample_t* target, sample_t* source, int length);
+void AUD_convert_s32_float(data_t* target, data_t* source, int length);
 
-void AUD_convert_s32_double(sample_t* target, sample_t* source, int length);
+void AUD_convert_s32_double(data_t* target, data_t* source, int length);
 
-void AUD_convert_float_u8(sample_t* target, sample_t* source, int length);
+void AUD_convert_float_u8(data_t* target, data_t* source, int length);
 
-void AUD_convert_float_s16(sample_t* target, sample_t* source, int length);
+void AUD_convert_float_s16(data_t* target, data_t* source, int length);
 
-void AUD_convert_float_s24_be(sample_t* target, sample_t* source, int length);
+void AUD_convert_float_s24_be(data_t* target, data_t* source, int length);
 
-void AUD_convert_float_s24_le(sample_t* target, sample_t* source, int length);
+void AUD_convert_float_s24_le(data_t* target, data_t* source, int length);
 
-void AUD_convert_float_s32(sample_t* target, sample_t* source, int length);
+void AUD_convert_float_s32(data_t* target, data_t* source, int length);
 
-void AUD_convert_float_double(sample_t* target, sample_t* source, int length);
+void AUD_convert_float_double(data_t* target, data_t* source, int length);
 
-void AUD_convert_double_u8(sample_t* target, sample_t* source, int length);
+void AUD_convert_double_u8(data_t* target, data_t* source, int length);
 
-void AUD_convert_double_s16(sample_t* target, sample_t* source, int length);
+void AUD_convert_double_s16(data_t* target, data_t* source, int length);
 
-void AUD_convert_double_s24_be(sample_t* target, sample_t* source, int length);
+void AUD_convert_double_s24_be(data_t* target, data_t* source, int length);
 
-void AUD_convert_double_s24_le(sample_t* target, sample_t* source, int length);
+void AUD_convert_double_s24_le(data_t* target, data_t* source, int length);
 
-void AUD_convert_double_s32(sample_t* target, sample_t* source, int length);
+void AUD_convert_double_s32(data_t* target, data_t* source, int length);
 
-void AUD_convert_double_float(sample_t* target, sample_t* source, int length);
-
-template <class T>
-void AUD_volume_adjust(sample_t* target, sample_t* source,
-					   int count, float volume)
-{
-	T* t = (T*)target;
-	T* s = (T*)source;
-	for(int i=0; i < count; i++)
-		t[i] = (T)(s[i] * volume);
-}
-
-void AUD_volume_adjust_u8(sample_t* target, sample_t* source,
-						  int count, float volume);
-
-void AUD_volume_adjust_s24_le(sample_t* target, sample_t* source,
-							  int count, float volume);
-
-void AUD_volume_adjust_s24_be(sample_t* target, sample_t* source,
-							  int count, float volume);
-
-template <class T>
-void AUD_rectify(sample_t* target, sample_t* source, int count)
-{
-	T* t = (T*)target;
-	T* s = (T*)source;
-	for(int i=0; i < count; i++)
-		t[i] = s[i] < 0 ? -s[i] : s[i];
-}
-
-void AUD_rectify_u8(sample_t* target, sample_t* source, int count);
-
-void AUD_rectify_s24_le(sample_t* target, sample_t* source, int count);
-
-void AUD_rectify_s24_be(sample_t* target, sample_t* source, int count);
+void AUD_convert_double_float(data_t* target, data_t* source, int length);
 
 #endif //AUD_CONVERTERFUNCTIONS

intern/audaspace/intern/AUD_ConverterReader.cpp

@@ -26,205 +26,37 @@
 #include "AUD_ConverterReader.h"
 #include "AUD_Buffer.h"
 
-AUD_ConverterReader::AUD_ConverterReader(AUD_IReader* reader, AUD_Specs specs) :
+AUD_ConverterReader::AUD_ConverterReader(AUD_IReader* reader,
+										 AUD_DeviceSpecs specs) :
 		AUD_EffectReader(reader)
 {
-	m_specs = reader->getSpecs();
+	m_specs.specs = reader->getSpecs();
 
 	int bigendian = 1;
 	bigendian = (((char*)&bigendian)[0]) ? 0: 1; // 1 if Big Endian
 
-	switch(m_specs.format)
+	switch(specs.format)
 	{
 	case AUD_FORMAT_U8:
-		switch(specs.format)
-		{
-		case AUD_FORMAT_U8:
-			m_convert = AUD_convert_copy<unsigned char>;
-			break;
-		case AUD_FORMAT_S16:
-			m_convert = AUD_convert_u8_s16;
-			break;
-		case AUD_FORMAT_S24:
-			if(bigendian)
-				m_convert = AUD_convert_u8_s24_be;
-			else
-				m_convert = AUD_convert_u8_s24_le;
-			break;
-		case AUD_FORMAT_S32:
-			m_convert = AUD_convert_u8_s32;
-			break;
-		case AUD_FORMAT_FLOAT32:
-			m_convert = AUD_convert_u8_float;
-			break;
-		case AUD_FORMAT_FLOAT64:
-			m_convert = AUD_convert_u8_double;
-			break;
-		default:
-			break;
-		}
+		m_convert = AUD_convert_float_u8;
 		break;
 	case AUD_FORMAT_S16:
-		switch(specs.format)
-		{
-		case AUD_FORMAT_U8:
-			m_convert = AUD_convert_s16_u8;
-			break;
-		case AUD_FORMAT_S16:
-			m_convert = AUD_convert_copy<int16_t>;
-			break;
-		case AUD_FORMAT_S24:
-			if(bigendian)
-				m_convert = AUD_convert_s16_s24_be;
-			else
-				m_convert = AUD_convert_s16_s24_le;
-			break;
-		case AUD_FORMAT_S32:
-			m_convert = AUD_convert_s16_s32;
-			break;
-		case AUD_FORMAT_FLOAT32:
-			m_convert = AUD_convert_s16_float;
-			break;
-		case AUD_FORMAT_FLOAT64:
-			m_convert = AUD_convert_s16_double;
-			break;
-		default:
-			break;
-		}
+		m_convert = AUD_convert_float_s16;
 		break;
 	case AUD_FORMAT_S24:
 		if(bigendian)
-			switch(specs.format)
-			{
-			case AUD_FORMAT_U8:
-				m_convert = AUD_convert_u8_s24_be;
-				break;
-			case AUD_FORMAT_S16:
-				m_convert = AUD_convert_s16_s24_be;
-				break;
-			case AUD_FORMAT_S24:
-				m_convert = AUD_convert_s24_s24;
-				break;
-			case AUD_FORMAT_S32:
-				m_convert = AUD_convert_s32_s24_be;
-				break;
-			case AUD_FORMAT_FLOAT32:
-				m_convert = AUD_convert_float_s24_be;
-				break;
-			case AUD_FORMAT_FLOAT64:
-				m_convert = AUD_convert_double_s24_be;
-				break;
-			default:
-				break;
-			}
+			m_convert = AUD_convert_float_s24_be;
 		else
-			switch(specs.format)
-			{
-			case AUD_FORMAT_U8:
-				m_convert = AUD_convert_u8_s24_le;
-				break;
-			case AUD_FORMAT_S16:
-				m_convert = AUD_convert_s16_s24_le;
-				break;
-			case AUD_FORMAT_S24:
-				m_convert = AUD_convert_s24_s24;
-				break;
-			case AUD_FORMAT_S32:
-				m_convert = AUD_convert_s32_s24_le;
-				break;
-			case AUD_FORMAT_FLOAT32:
-				m_convert = AUD_convert_float_s24_le;
-				break;
-			case AUD_FORMAT_FLOAT64:
-				m_convert = AUD_convert_double_s24_le;
-				break;
-			default:
-				break;
-			}
+			m_convert = AUD_convert_float_s24_le;
 		break;
 	case AUD_FORMAT_S32:
-		switch(specs.format)
-		{
-		case AUD_FORMAT_U8:
-			m_convert = AUD_convert_s32_u8;
-			break;
-		case AUD_FORMAT_S16:
-			m_convert = AUD_convert_s32_s16;
-			break;
-		case AUD_FORMAT_S24:
-			if(bigendian)
-				m_convert = AUD_convert_s32_s24_be;
-			else
-				m_convert = AUD_convert_s32_s24_le;
-			break;
-		case AUD_FORMAT_S32:
-			m_convert = AUD_convert_copy<int32_t>;
-			break;
-		case AUD_FORMAT_FLOAT32:
-			m_convert = AUD_convert_s32_float;
-			break;
-		case AUD_FORMAT_FLOAT64:
-			m_convert = AUD_convert_s32_double;
-			break;
-		default:
-			break;
-		}
+		m_convert = AUD_convert_float_s32;
 		break;
 	case AUD_FORMAT_FLOAT32:
-		switch(specs.format)
-		{
-		case AUD_FORMAT_U8:
-			m_convert = AUD_convert_float_u8;
-			break;
-		case AUD_FORMAT_S16:
-			m_convert = AUD_convert_float_s16;
-			break;
-		case AUD_FORMAT_S24:
-			if(bigendian)
-				m_convert = AUD_convert_float_s24_be;
-			else
-				m_convert = AUD_convert_float_s24_le;
-			break;
-		case AUD_FORMAT_S32:
-			m_convert = AUD_convert_float_s32;
-			break;
-		case AUD_FORMAT_FLOAT32:
-			m_convert = AUD_convert_copy<float>;
-			break;
-		case AUD_FORMAT_FLOAT64:
-			m_convert = AUD_convert_float_double;
-			break;
-		default:
-			break;
-		}
+		m_convert = AUD_convert_copy<float>;
 		break;
 	case AUD_FORMAT_FLOAT64:
-		switch(specs.format)
-		{
-		case AUD_FORMAT_U8:
-			m_convert = AUD_convert_double_u8;
-			break;
-		case AUD_FORMAT_S16:
-			m_convert = AUD_convert_double_s16;
-			break;
-		case AUD_FORMAT_S24:
-			if(bigendian)
-				m_convert = AUD_convert_double_s24_be;
-			else
-				m_convert = AUD_convert_double_s24_le;
-			break;
-		case AUD_FORMAT_S32:
-			m_convert = AUD_convert_double_s32;
-			break;
-		case AUD_FORMAT_FLOAT32:
-			m_convert = AUD_convert_double_float;
-			break;
-		case AUD_FORMAT_FLOAT64:
-			m_convert = AUD_convert_copy<double>;
-			break;
-		default:
-			break;
-		}
+		m_convert = AUD_convert_float_double;
 		break;
 	default:
 		break;
@@ -242,7 +74,7 @@ AUD_ConverterReader::~AUD_ConverterReader()
 
 AUD_Specs AUD_ConverterReader::getSpecs()
 {
-	return m_specs;
+	return m_specs.specs;
 }
 
 void AUD_ConverterReader::read(int & length, sample_t* & buffer)
@@ -254,7 +86,8 @@ void AUD_ConverterReader::read(int & length, sample_t* & buffer)
 	if(m_buffer->getSize() < length*samplesize)
 		m_buffer->resize(length*samplesize);
 
-	m_convert(m_buffer->getBuffer(), buffer, length*m_specs.channels);
+	m_convert((data_t*)m_buffer->getBuffer(), (data_t*)buffer,
+			  length * m_specs.channels);
 
 	buffer = m_buffer->getBuffer();
 }

intern/audaspace/intern/AUD_ConverterReader.h

@@ -44,7 +44,7 @@ private:
 	/**
 	 * The target specification.
 	 */
-	AUD_Specs m_specs;
+	AUD_DeviceSpecs m_specs;
 
 	/**
 	 * Converter function.
@@ -58,7 +58,7 @@ public:
 	 * \param specs The target specification.
 	 * \exception AUD_Exception Thrown if the reader is NULL.
 	 */
-	AUD_ConverterReader(AUD_IReader* reader, AUD_Specs specs);
+	AUD_ConverterReader(AUD_IReader* reader, AUD_DeviceSpecs specs);
 	/**
 	 * Destroys the reader.
 	 */

intern/audaspace/intern/AUD_FloatMixer.h

@@ -1,100 +0,0 @@
-/*
- * $Id$
- *
- * ***** BEGIN LGPL LICENSE BLOCK *****
- *
- * Copyright 2009 Jörg Hermann Müller
- *
- * This file is part of AudaSpace.
- *
- * AudaSpace is free software: you can redistribute it and/or modify
- * it under the terms of the GNU Lesser General Public License as published by
- * the Free Software Foundation, either version 3 of the License, or
- * (at your option) any later version.
- *
- * AudaSpace 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 Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public License
- * along with AudaSpace.  If not, see <http://www.gnu.org/licenses/>.
- *
- * ***** END LGPL LICENSE BLOCK *****
- */
-
-#ifndef AUD_FLOATMIXER
-#define AUD_FLOATMIXER
-
-#include "AUD_IMixer.h"
-#include "AUD_ConverterFunctions.h"
-class AUD_ConverterFactory;
-class AUD_SRCResampleFactory;
-class AUD_ChannelMapperFactory;
-class AUD_Buffer;
-#include <list>
-
-struct AUD_FloatMixerBuffer
-{
-	sample_t* buffer;
-	int length;
-	float volume;
-};
-
-/**
- * This class is able to mix two audiosignals with floats.
- */
-class AUD_FloatMixer : public AUD_IMixer
-{
-private:
-	/**
-	 * The converter factory that converts all readers for superposition.
-	 */
-	AUD_ConverterFactory* m_converter;
-
-	/**
-	 * The resampling factory that resamples all readers for superposition.
-	 */
-	AUD_SRCResampleFactory* m_resampler;
-
-	/**
-	 * The channel mapper factory that maps all readers for superposition.
-	 */
-	AUD_ChannelMapperFactory* m_mapper;
-
-	/**
-	 * The list of buffers to superpose.
-	 */
-	std::list<AUD_FloatMixerBuffer> m_buffers;
-
-	/**
-	 * The output specification.
-	 */
-	AUD_Specs m_specs;
-
-	/**
-	 * The temporary mixing buffer.
-	 */
-	AUD_Buffer* m_buffer;
-
-	/**
-	 * Converter function.
-	 */
-	AUD_convert_f m_convert;
-
-public:
-	/**
-	 * Creates the mixer.
-	 */
-	AUD_FloatMixer();
-
-	virtual ~AUD_FloatMixer();
-
-	virtual AUD_IReader* prepare(AUD_IReader* reader);
-	virtual void setSpecs(AUD_Specs specs);
-	virtual void add(sample_t* buffer, AUD_Specs specs, int length,
-					 float volume);
-	virtual void superpose(sample_t* buffer, int length, float volume);
-};
-
-#endif //AUD_FLOATMIXER

intern/audaspace/intern/AUD_IDevice.h

@@ -53,7 +53,7 @@ public:
 	/**
 	 * Returns the specification of the device.
 	 */
-	virtual AUD_Specs getSpecs()=0;
+	virtual AUD_DeviceSpecs getSpecs()=0;
 
 	/**
 	 * Plays a sound source.

intern/audaspace/intern/AUD_LinearResampleFactory.cpp

@@ -23,42 +23,30 @@
  * ***** END LGPL LICENSE BLOCK *****
  */
 
-#include "AUD_SDLMixerFactory.h"
-#include "AUD_SDLMixerReader.h"
+#include "AUD_LinearResampleFactory.h"
+#include "AUD_LinearResampleReader.h"
 
-#include <cstring>
+AUD_LinearResampleFactory::AUD_LinearResampleFactory(AUD_IReader* reader,
+													 AUD_DeviceSpecs specs) :
+		AUD_ResampleFactory(reader, specs) {}
 
-AUD_SDLMixerFactory::AUD_SDLMixerFactory(AUD_IReader* reader, AUD_Specs specs) :
-		AUD_MixerFactory(reader, specs) {}
+AUD_LinearResampleFactory::AUD_LinearResampleFactory(AUD_IFactory* factory,
+													 AUD_DeviceSpecs specs) :
+		AUD_ResampleFactory(factory, specs) {}
 
-AUD_SDLMixerFactory::AUD_SDLMixerFactory(AUD_IFactory* factory, AUD_Specs specs) :
-		AUD_MixerFactory(factory, specs) {}
+AUD_LinearResampleFactory::AUD_LinearResampleFactory(AUD_DeviceSpecs specs) :
+		AUD_ResampleFactory(specs) {}
 
-AUD_SDLMixerFactory::AUD_SDLMixerFactory(AUD_Specs specs) :
-		AUD_MixerFactory(specs) {}
-
-AUD_IReader* AUD_SDLMixerFactory::createReader()
+AUD_IReader* AUD_LinearResampleFactory::createReader()
 {
 	AUD_IReader* reader = getReader();
 
 	if(reader != 0)
 	{
-		AUD_Specs specs = reader->getSpecs();
-		if(memcmp(&m_specs, &specs, sizeof(AUD_Specs)) != 0)
+		if(reader->getSpecs().rate != m_specs.rate)
 		{
-			try
-			{
-				reader = new AUD_SDLMixerReader(reader, m_specs);
-				AUD_NEW("reader")
-			}
-			catch(AUD_Exception e)
-			{
-				// return 0 in case SDL cannot mix the source
-				if(e.error != AUD_ERROR_SDL)
-					throw;
-				else
-					reader = NULL;
-			}
+			reader = new AUD_LinearResampleReader(reader, m_specs.specs);
+			AUD_NEW("reader")
 		}
 	}
 	return reader;

intern/audaspace/intern/AUD_LinearResampleFactory.h

@@ -23,23 +23,22 @@
  * ***** END LGPL LICENSE BLOCK *****
  */
 
-#ifndef AUD_SDLMIXERFACTORY
-#define AUD_SDLMIXERFACTORY
+#ifndef AUD_LINEARRESAMPLEFACTORY
+#define AUD_LINEARRESAMPLEFACTORY
 
-#include "AUD_MixerFactory.h"
+#include "AUD_ResampleFactory.h"
 
 /**
- * This factory creates a resampling reader that uses SDL's resampling
- * functionality which unfortunately is very very very limited.
+ * This factory creates a resampling reader that does simple linear resampling.
  */
-class AUD_SDLMixerFactory : public AUD_MixerFactory
+class AUD_LinearResampleFactory : public AUD_ResampleFactory
 {
 public:
-	AUD_SDLMixerFactory(AUD_IReader* reader, AUD_Specs specs);
-	AUD_SDLMixerFactory(AUD_IFactory* factory, AUD_Specs specs);
-	AUD_SDLMixerFactory(AUD_Specs specs);
+	AUD_LinearResampleFactory(AUD_IReader* reader, AUD_DeviceSpecs specs);
+	AUD_LinearResampleFactory(AUD_IFactory* factory, AUD_DeviceSpecs specs);
+	AUD_LinearResampleFactory(AUD_DeviceSpecs specs);
 
 	virtual AUD_IReader* createReader();
 };
 
-#endif //AUD_SDLMIXERFACTORY
+#endif //AUD_LINEARRESAMPLEFACTORY

intern/audaspace/intern/AUD_LinearResampleReader.cpp

@@ -0,0 +1,138 @@
+/*
+ * $Id$
+ *
+ * ***** BEGIN LGPL LICENSE BLOCK *****
+ *
+ * Copyright 2009 Jörg Hermann Müller
+ *
+ * This file is part of AudaSpace.
+ *
+ * AudaSpace is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU Lesser General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * AudaSpace 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 Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public License
+ * along with AudaSpace.  If not, see <http://www.gnu.org/licenses/>.
+ *
+ * ***** END LGPL LICENSE BLOCK *****
+ */
+
+#include "AUD_LinearResampleReader.h"
+#include "AUD_Buffer.h"
+
+#include <cmath>
+#include <cstring>
+#include <cstdio>
+
+AUD_LinearResampleReader::AUD_LinearResampleReader(AUD_IReader* reader,
+												   AUD_Specs specs) :
+		AUD_EffectReader(reader)
+{
+	m_sspecs = reader->getSpecs();
+
+	m_tspecs = specs;
+	m_tspecs.channels = m_sspecs.channels;
+	m_factor = (float)m_tspecs.rate / (float)m_sspecs.rate;
+
+	m_position = 0;
+	m_sposition = 0;
+
+	m_buffer = new AUD_Buffer(); AUD_NEW("buffer")
+	m_cache = new AUD_Buffer(2 * AUD_SAMPLE_SIZE(specs)); AUD_NEW("buffer")
+}
+
+AUD_LinearResampleReader::~AUD_LinearResampleReader()
+{
+	delete m_buffer; AUD_DELETE("buffer")
+	delete m_cache; AUD_DELETE("buffer")
+}
+
+void AUD_LinearResampleReader::seek(int position)
+{
+	m_position = position;
+	m_sposition = floor(position / m_factor);
+	m_reader->seek(m_sposition);
+}
+
+int AUD_LinearResampleReader::getLength()
+{
+	return m_reader->getLength() * m_factor;
+}
+
+int AUD_LinearResampleReader::getPosition()
+{
+	return m_position;
+}
+
+AUD_Specs AUD_LinearResampleReader::getSpecs()
+{
+	return m_tspecs;
+}
+
+void AUD_LinearResampleReader::read(int & length, sample_t* & buffer)
+{
+	int size = length * AUD_SAMPLE_SIZE(m_tspecs);
+
+	if(m_buffer->getSize() < size)
+		m_buffer->resize(size);
+
+	int need = ceil((m_position + length) / m_factor) + 1 - m_sposition;
+	int len = need;
+	sample_t* buf;
+	buffer = m_buffer->getBuffer();
+
+	m_reader->read(len, buf);
+
+	if(len < need)
+		length = floor((m_sposition + len - 1) * m_factor) - m_position;
+
+	float spos;
+	sample_t low, high;
+	int channels = m_sspecs.channels;
+
+	for(int channel = 0; channel < channels; channel++)
+	{
+		for(int i = 0; i < length; i++)
+		{
+			spos = (m_position + i) / m_factor - m_sposition;
+
+			if((floor(spos) < -2) || (ceil(spos) >= len))
+			{
+				fprintf(stderr, "FATAL ERROR: REPORT THIS TO neXyon!\n");
+//				exit(1);
+			}
+
+			if(floor(spos) < 0)
+			{
+				low = m_cache->getBuffer()[(int)(floor(spos) + 2) * channels
+										   + channel];
+				if(ceil(spos) < 0)
+					high = m_cache->getBuffer()[(int)(ceil(spos) + 2)
+												* channels + channel];
+				else
+					high = buf[(int)ceil(spos) * channels + channel];
+			}
+			else
+			{
+					low = buf[(int)floor(spos) * channels + channel];
+					high = buf[(int)ceil(spos) * channels + channel];
+			}
+			buffer[i * channels + channel] = low + (spos - floor(spos)) *
+												   (high - low);
+		}
+	}
+
+	if(len > 1)
+		memcpy(m_cache->getBuffer(), buf + (len - 2) * channels, 2 * channels);
+	else if(len == 1)
+		memcpy(m_cache->getBuffer() + 1 * channels, buf, channels);
+
+	m_sposition += len;
+	m_position += length;
+}

intern/audaspace/intern/AUD_LinearResampleReader.h

@@ -0,0 +1,94 @@
+/*
+ * $Id$
+ *
+ * ***** BEGIN LGPL LICENSE BLOCK *****
+ *
+ * Copyright 2009 Jörg Hermann Müller
+ *
+ * This file is part of AudaSpace.
+ *
+ * AudaSpace is free software: you can redistribute it and/or modify
+ * it under the terms of the GNU Lesser General Public License as published by
+ * the Free Software Foundation, either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * AudaSpace 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 Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public License
+ * along with AudaSpace.  If not, see <http://www.gnu.org/licenses/>.
+ *
+ * ***** END LGPL LICENSE BLOCK *****
+ */
+
+#ifndef AUD_LINEARRESAMPLEREADER
+#define AUD_LINEARRESAMPLEREADER
+
+#include "AUD_EffectReader.h"
+class AUD_Buffer;
+
+/**
+ * This resampling reader uses libsamplerate for resampling.
+ */
+class AUD_LinearResampleReader : public AUD_EffectReader
+{
+private:
+	/**
+	 * The resampling factor.
+	 */
+	float m_factor;
+
+	/**
+	 * The current position.
+	 */
+	int m_position;
+
+	/**
+	 * The current reading source position.
+	 */
+	int m_sposition;
+
+	/**
+	 * The sound output buffer.
+	 */
+	AUD_Buffer *m_buffer;
+
+	/**
+	 * The input caching buffer.
+	 */
+	AUD_Buffer *m_cache;
+
+	/**
+	 * The target specification.
+	 */
+	AUD_Specs m_tspecs;
+
+	/**
+	 * The sample specification of the source.
+	 */
+	AUD_Specs m_sspecs;
+
+public:
+	/**
+	 * Creates a resampling reader.
+	 * \param reader The reader to mix.
+	 * \param specs The target specification.
+	 * \exception AUD_Exception Thrown if the reader is NULL.
+	 */
+	AUD_LinearResampleReader(AUD_IReader* reader, AUD_Specs specs);
+
+	/**
+	 * Destroys the reader.
+	 */
+	~AUD_LinearResampleReader();
+
+	virtual void seek(int position);
+	virtual int getLength();
+	virtual int getPosition();
+	virtual AUD_Specs getSpecs();
+	virtual void read(int & length, sample_t* & buffer);
+};
+
+#endif //AUD_LINEARRESAMPLEREADER

intern/audaspace/intern/AUD_Mixer.cpp

@@ -23,8 +23,7 @@
  * ***** END LGPL LICENSE BLOCK *****
  */
 
-#include "AUD_FloatMixer.h"
-#include "AUD_ConverterFactory.h"
+#include "AUD_Mixer.h"
 #include "AUD_SRCResampleFactory.h"
 #include "AUD_ChannelMapperFactory.h"
 #include "AUD_IReader.h"
@@ -32,23 +31,19 @@
 
 #include <cstring>
 
-AUD_FloatMixer::AUD_FloatMixer()
+AUD_Mixer::AUD_Mixer()
 {
 	m_buffer = new AUD_Buffer(); AUD_NEW("buffer")
 
-	m_converter = NULL;
 	m_resampler = NULL;
 	m_mapper = NULL;
 }
 
-AUD_FloatMixer::~AUD_FloatMixer()
+AUD_Mixer::~AUD_Mixer()
 {
 	delete m_buffer; AUD_DELETE("buffer")
 
-	if(m_converter)
-	{
-		delete m_converter; AUD_DELETE("factory")
-	}
+
 	if(m_resampler)
 	{
 		delete m_resampler; AUD_DELETE("factory")
@@ -59,11 +54,8 @@ AUD_FloatMixer::~AUD_FloatMixer()
 	}
 }
 
-AUD_IReader* AUD_FloatMixer::prepare(AUD_IReader* reader)
+AUD_IReader* AUD_Mixer::prepare(AUD_IReader* reader)
 {
-	m_converter->setReader(reader);
-	reader = m_converter->createReader();
-
 	m_resampler->setReader(reader);
 	reader = m_resampler->createReader();
 
@@ -76,14 +68,10 @@ AUD_IReader* AUD_FloatMixer::prepare(AUD_IReader* reader)
 	return reader;
 }
 
-void AUD_FloatMixer::setSpecs(AUD_Specs specs)
+void AUD_Mixer::setSpecs(AUD_DeviceSpecs specs)
 {
 	m_specs = specs;
 
-	if(m_converter)
-	{
-		delete m_converter; AUD_DELETE("factory")
-	}
 	if(m_resampler)
 	{
 		delete m_resampler; AUD_DELETE("factory")
@@ -93,9 +81,6 @@ void AUD_FloatMixer::setSpecs(AUD_Specs specs)
 		delete m_mapper; AUD_DELETE("factory")
 	}
 
-	specs.format = AUD_FORMAT_FLOAT32;
-
-	m_converter = new AUD_ConverterFactory(specs); AUD_NEW("factory")
 	m_resampler = new AUD_SRCResampleFactory(specs); AUD_NEW("factory")
 	m_mapper = new AUD_ChannelMapperFactory(specs); AUD_NEW("factory")
 
@@ -130,27 +115,26 @@ void AUD_FloatMixer::setSpecs(AUD_Specs specs)
 	}
 }
 
-void AUD_FloatMixer::add(sample_t* buffer, AUD_Specs specs, int length,
-						 float volume)
+void AUD_Mixer::add(sample_t* buffer, int length, float volume)
 {
-	AUD_FloatMixerBuffer buf;
+	AUD_MixerBuffer buf;
 	buf.buffer = buffer;
 	buf.length = length;
 	buf.volume = volume;
 	m_buffers.push_back(buf);
 }
 
-void AUD_FloatMixer::superpose(sample_t* buffer, int length, float volume)
+void AUD_Mixer::superpose(data_t* buffer, int length, float volume)
 {
-	AUD_FloatMixerBuffer buf;
+	AUD_MixerBuffer buf;
 
 	int channels = m_specs.channels;
 
 	if(m_buffer->getSize() < length * channels * 4)
 		m_buffer->resize(length * channels * 4);
 
-	float* out = (float*)m_buffer->getBuffer();
-	float* in;
+	sample_t* out = m_buffer->getBuffer();
+	sample_t* in;
 
 	memset(out, 0, length * channels * 4);
 
@@ -162,11 +146,11 @@ void AUD_FloatMixer::superpose(sample_t* buffer, int length, float volume)
 		m_buffers.pop_front();
 
 		end = buf.length*channels;
-		in = (float*) buf.buffer;
+		in = buf.buffer;
 
 		for(int i = 0; i < end; i++)
 			out[i] += in[i]*buf.volume * volume;
 	}
 
-	m_convert(buffer, (sample_t*) out, length * channels);
+	m_convert(buffer, (data_t*) out, length * channels);
 }

intern/audaspace/intern/AUD_Mixer.h

@@ -23,47 +23,95 @@
  * ***** END LGPL LICENSE BLOCK *****
  */
 
-#ifndef AUD_IMIXER
-#define AUD_IMIXER
+#ifndef AUD_MIXER
+#define AUD_MIXER
 
-#include "AUD_Space.h"
+#include "AUD_ConverterFunctions.h"
+class AUD_ConverterFactory;
+class AUD_SRCResampleFactory;
+class AUD_ChannelMapperFactory;
+class AUD_Buffer;
 class AUD_IReader;
+#include <list>
+
+struct AUD_MixerBuffer
+{
+	sample_t* buffer;
+	int length;
+	float volume;
+};
 
 /**
- * This class is able to mix audiosignals of different format and channel count.
- * \note This class doesn't do resampling!
+ * This class is able to mix audiosignals of different channel count and sample
+ * rate and convert it to a specific output format.
+ * It uses a default ChannelMapperFactory and a SRCResampleFactory for
+ * the perparation.
  */
-class AUD_IMixer
+class AUD_Mixer
 {
+private:
+	/**
+	 * The resampling factory that resamples all readers for superposition.
+	 */
+	AUD_SRCResampleFactory* m_resampler;
+
+	/**
+	 * The channel mapper factory that maps all readers for superposition.
+	 */
+	AUD_ChannelMapperFactory* m_mapper;
+
+	/**
+	 * The list of buffers to superpose.
+	 */
+	std::list<AUD_MixerBuffer> m_buffers;
+
+	/**
+	 * The output specification.
+	 */
+	AUD_DeviceSpecs m_specs;
+
+	/**
+	 * The temporary mixing buffer.
+	 */
+	AUD_Buffer* m_buffer;
+
+	/**
+	 * Converter function.
+	 */
+	AUD_convert_f m_convert;
+
 public:
+	/**
+	 * Creates the mixer.
+	 */
+	AUD_Mixer();
+
 	/**
 	 * Destroys the mixer.
 	 */
-	virtual ~AUD_IMixer(){}
+	~AUD_Mixer();
 
 	/**
 	 * This funuction prepares a reader for playback.
 	 * \param reader The reader to prepare.
 	 * \return The reader that should be used for playback.
 	 */
-	virtual AUD_IReader* prepare(AUD_IReader* reader)=0;
+	AUD_IReader* prepare(AUD_IReader* reader);
 
 	/**
 	 * Sets the target specification for superposing.
 	 * \param specs The target specification.
 	 */
-	virtual void setSpecs(AUD_Specs specs)=0;
+	void setSpecs(AUD_DeviceSpecs specs);
 
 	/**
 	 * Adds a buffer for superposition.
 	 * \param buffer The buffer to superpose.
-	 * \param specs The specification of the buffer.
 	 * \param start The start sample of the buffer.
 	 * \param length The length of the buffer in samples.
 	 * \param volume The mixing volume. Must be a value between 0.0 and 1.0.
 	 */
-	virtual void add(sample_t* buffer, AUD_Specs specs, int length,
-					 float volume)=0;
+	void add(sample_t* buffer, int length, float volume);
 
 	/**
 	 * Superposes all added buffers into an output buffer.
@@ -71,7 +119,7 @@ public:
 	 * \param length The length of the buffer in samples.
 	 * \param volume The mixing volume. Must be a value between 0.0 and 1.0.
 	 */
-	virtual void superpose(sample_t* buffer, int length, float volume)=0;
+	void superpose(data_t* buffer, int length, float volume);
 };
 
-#endif //AUD_IMIXER
+#endif //AUD_MIXER

intern/audaspace/intern/AUD_MixerFactory.cpp

@@ -49,7 +49,7 @@ AUD_IReader* AUD_MixerFactory::getReader()
 }
 
 AUD_MixerFactory::AUD_MixerFactory(AUD_IReader* reader,
-											   AUD_Specs specs)
+								   AUD_DeviceSpecs specs)
 {
 	m_specs = specs;
 	m_reader = reader;
@@ -57,14 +57,14 @@ AUD_MixerFactory::AUD_MixerFactory(AUD_IReader* reader,
 }
 
 AUD_MixerFactory::AUD_MixerFactory(AUD_IFactory* factory,
-											   AUD_Specs specs)
+								   AUD_DeviceSpecs specs)
 {
 	m_specs = specs;
 	m_reader = 0;
 	m_factory = factory;
 }
 
-AUD_MixerFactory::AUD_MixerFactory(AUD_Specs specs)
+AUD_MixerFactory::AUD_MixerFactory(AUD_DeviceSpecs specs)
 {
 	m_specs = specs;
 	m_reader = 0;
@@ -79,12 +79,12 @@ AUD_MixerFactory::~AUD_MixerFactory()
 	}
 }
 
-AUD_Specs AUD_MixerFactory::getSpecs()
+AUD_DeviceSpecs AUD_MixerFactory::getSpecs()
 {
 	return m_specs;
 }
 
-void AUD_MixerFactory::setSpecs(AUD_Specs specs)
+void AUD_MixerFactory::setSpecs(AUD_DeviceSpecs specs)
 {
 	m_specs = specs;
 }

intern/audaspace/intern/AUD_MixerFactory.h

@@ -47,7 +47,7 @@ protected:
 	/**
 	 * The target specification for resampling.
 	 */
-	AUD_Specs m_specs;
+	AUD_DeviceSpecs m_specs;
 
 	/**
 	 * Returns the reader created out of the factory or taken from m_reader.
@@ -63,20 +63,20 @@ public:
 	 * \param reader The reader to mix.
 	 * \param specs The target specification.
 	 */
-	AUD_MixerFactory(AUD_IReader* reader, AUD_Specs specs);
+	AUD_MixerFactory(AUD_IReader* reader, AUD_DeviceSpecs specs);
 
 	/**
 	 * Creates a new factory.
 	 * \param factory The factory to create the readers to mix out of.
 	 * \param specs The target specification.
 	 */
-	AUD_MixerFactory(AUD_IFactory* factory, AUD_Specs specs);
+	AUD_MixerFactory(AUD_IFactory* factory, AUD_DeviceSpecs specs);
 
 	/**
 	 * Creates a new factory.
 	 * \param specs The target specification.
 	 */
-	AUD_MixerFactory(AUD_Specs specs);
+	AUD_MixerFactory(AUD_DeviceSpecs specs);
 
 	/**
 	 * Destroys the resampling factory.
@@ -86,13 +86,13 @@ public:
 	/**
 	 * Returns the target specification for resampling.
 	 */
-	AUD_Specs getSpecs();
+	AUD_DeviceSpecs getSpecs();
 
 	/**
 	 * Sets the target specification for resampling.
 	 * \param specs The specification.
 	 */
-	void setSpecs(AUD_Specs specs);
+	void setSpecs(AUD_DeviceSpecs specs);
 
 	/**
 	 * Sets the reader for resampling.

intern/audaspace/intern/AUD_NULLDevice.cpp

@@ -34,7 +34,7 @@ AUD_NULLDevice::AUD_NULLDevice()
 	m_specs.rate = AUD_RATE_INVALID;
 }
 
-AUD_Specs AUD_NULLDevice::getSpecs()
+AUD_DeviceSpecs AUD_NULLDevice::getSpecs()
 {
 	return m_specs;
 }

intern/audaspace/intern/AUD_NULLDevice.h

@@ -37,7 +37,7 @@ private:
 	/**
 	 * The specs of the device.
 	 */
-	AUD_Specs m_specs;
+	AUD_DeviceSpecs m_specs;
 
 public:
 	/**
@@ -45,7 +45,7 @@ public:
 	 */
 	AUD_NULLDevice();
 
-	virtual AUD_Specs getSpecs();
+	virtual AUD_DeviceSpecs getSpecs();
 	virtual AUD_Handle* play(AUD_IFactory* factory, bool keep = false);
 	virtual bool pause(AUD_Handle* handle);
 	virtual bool resume(AUD_Handle* handle);

intern/audaspace/intern/AUD_ReadDevice.cpp

@@ -23,17 +23,17 @@
  * ***** END LGPL LICENSE BLOCK *****
  */
 
-#include "AUD_FloatMixer.h"
+#include "AUD_Mixer.h"
 #include "AUD_ReadDevice.h"
 #include "AUD_IReader.h"
 
 #include <cstring>
 
-AUD_ReadDevice::AUD_ReadDevice(AUD_Specs specs)
+AUD_ReadDevice::AUD_ReadDevice(AUD_DeviceSpecs specs)
 {
 	m_specs = specs;
 
-	m_mixer = new AUD_FloatMixer(); AUD_NEW("mixer")
+	m_mixer = new AUD_Mixer(); AUD_NEW("mixer")
 	m_mixer->setSpecs(m_specs);
 
 	m_playing = false;
@@ -46,15 +46,15 @@ AUD_ReadDevice::~AUD_ReadDevice()
 	destroy();
 }
 
-bool AUD_ReadDevice::read(sample_t* buffer, int length)
+bool AUD_ReadDevice::read(data_t* buffer, int length)
 {
 	if(m_playing)
 		mix(buffer, length);
 	else
 		if(m_specs.format == AUD_FORMAT_U8)
-			memset(buffer, 0x80, length * AUD_SAMPLE_SIZE(m_specs));
+			memset(buffer, 0x80, length * AUD_DEVICE_SAMPLE_SIZE(m_specs));
 		else
-			memset(buffer, 0, length * AUD_SAMPLE_SIZE(m_specs));
+			memset(buffer, 0, length * AUD_DEVICE_SAMPLE_SIZE(m_specs));
 	return m_playing;
 }
 

intern/audaspace/intern/AUD_ReadDevice.h

@@ -47,7 +47,7 @@ public:
 	 * Creates a new read device.
 	 * \param specs The wanted audio specification.
 	 */
-	AUD_ReadDevice(AUD_Specs specs);
+	AUD_ReadDevice(AUD_DeviceSpecs specs);
 
 	/**
 	 * Closes the device.
@@ -62,7 +62,7 @@ public:
 	 *         played back currently, in that case the buffer is filled with
 	 *         silence.
 	 */
-	bool read(sample_t* buffer, int length);
+	bool read(data_t* buffer, int length);
 };
 
 #endif //AUD_READDEVICE

intern/audaspace/intern/AUD_SinusReader.cpp

@@ -26,7 +26,7 @@
 #include "AUD_SinusReader.h"
 #include "AUD_Buffer.h"
 
-#include <math.h>
+#include <cmath>
 
 #ifndef M_PI
 #define M_PI 3.14159265358979323846
@@ -69,8 +69,7 @@ AUD_Specs AUD_SinusReader::getSpecs()
 {
 	AUD_Specs specs;
 	specs.rate = m_sampleRate;
-	specs.format = AUD_FORMAT_S16;
-	specs.channels = AUD_CHANNELS_STEREO;
+	specs.channels = AUD_CHANNELS_MONO;
 	return specs;
 }
 
@@ -87,18 +86,16 @@ bool AUD_SinusReader::notify(AUD_Message &message)
 void AUD_SinusReader::read(int & length, sample_t* & buffer)
 {
 	// resize if necessary
-	if(m_buffer->getSize() < length*4)
-		m_buffer->resize(length*4);
+	if(m_buffer->getSize() < length * sizeof(sample_t))
+		m_buffer->resize(length * sizeof(sample_t));
 
 	// fill with sine data
-	short* buf = (short*) m_buffer->getBuffer();
-	for(int i=0; i < length; i++)
+	buffer = m_buffer->getBuffer();
+	for(int i = 0; i < length; i++)
 	{
-		buf[i*2] = sin((m_position + i) * 2.0 * M_PI * m_frequency /
-					   (float)m_sampleRate) * 32700;
-		buf[i*2+1] = buf[i*2];
+		buffer[i] = sin((m_position + i) * 2.0 * M_PI * m_frequency /
+						(float)m_sampleRate);
 	}
 
-	buffer = (sample_t*)buf;
 	m_position += length;
 }

intern/audaspace/intern/AUD_SoftwareDevice.cpp

@@ -25,7 +25,7 @@
 
 #include "AUD_SoftwareDevice.h"
 #include "AUD_IReader.h"
-#include "AUD_IMixer.h"
+#include "AUD_Mixer.h"
 #include "AUD_IFactory.h"
 #include "AUD_SourceCaps.h"
 
@@ -52,6 +52,8 @@ void AUD_SoftwareDevice::create()
 	m_pausedSounds = new std::list<AUD_SoftwareHandle*>(); AUD_NEW("list")
 	m_playback = false;
 	m_volume = 1.0;
+	m_mixer = new AUD_Mixer(); AUD_NEW("mixer")
+	m_mixer->setSpecs(m_specs);
 
 	pthread_mutexattr_t attr;
 	pthread_mutexattr_init(&attr);
@@ -90,7 +92,7 @@ void AUD_SoftwareDevice::destroy()
 	pthread_mutex_destroy(&m_mutex);
 }
 
-void AUD_SoftwareDevice::mix(sample_t* buffer, int length)
+void AUD_SoftwareDevice::mix(data_t* buffer, int length)
 {
 	lock();
 
@@ -98,7 +100,7 @@ void AUD_SoftwareDevice::mix(sample_t* buffer, int length)
 		AUD_SoftwareHandle* sound;
 		int len;
 		sample_t* buf;
-		int sample_size = AUD_SAMPLE_SIZE(m_specs);
+		int sample_size = AUD_DEVICE_SAMPLE_SIZE(m_specs);
 		std::list<AUD_SoftwareHandle*> stopSounds;
 
 		// for all sounds
@@ -114,7 +116,7 @@ void AUD_SoftwareDevice::mix(sample_t* buffer, int length)
 			len = length;
 			sound->reader->read(len, buf);
 
-			m_mixer->add(buf, sound->reader->getSpecs(), len, sound->volume);
+			m_mixer->add(buf, len, sound->volume);
 
 			// in case the end of the sound is reached
 			if(len < length)
@@ -159,14 +161,7 @@ bool AUD_SoftwareDevice::isValid(AUD_Handle* handle)
 	return false;
 }
 
-void AUD_SoftwareDevice::setMixer(AUD_IMixer* mixer)
-{
-	delete m_mixer; AUD_DELETE("mixer")
-	m_mixer = mixer;
-	mixer->setSpecs(m_specs);
-}
-
-AUD_Specs AUD_SoftwareDevice::getSpecs()
+AUD_DeviceSpecs AUD_SoftwareDevice::getSpecs()
 {
 	return m_specs;
 }

intern/audaspace/intern/AUD_SoftwareDevice.h

@@ -28,7 +28,7 @@
 
 #include "AUD_IDevice.h"
 struct AUD_SoftwareHandle;
-class AUD_IMixer;
+class AUD_Mixer;
 
 #include <list>
 #include <pthread.h>
@@ -47,12 +47,12 @@ protected:
 	/**
 	 * The specification of the device.
 	 */
-	AUD_Specs m_specs;
+	AUD_DeviceSpecs m_specs;
 
 	/**
-	 * The mixer. Will be deleted by the destroy function.
+	 * The mixer.
 	 */
-	AUD_IMixer* m_mixer;
+	AUD_Mixer* m_mixer;
 
 	/**
 	 * Initializes member variables.
@@ -69,7 +69,7 @@ protected:
 	 * \param buffer The target buffer.
 	 * \param length The length in samples to be filled.
 	 */
-	void mix(sample_t* buffer, int length);
+	void mix(data_t* buffer, int length);
 
 	/**
 	 * This function tells the device, to start or pause playback.
@@ -111,13 +111,7 @@ private:
 	bool isValid(AUD_Handle* handle);
 
 public:
-	/**
-	 * Sets a new mixer.
-	 * \param mixer The new mixer.
-	 */
-	void setMixer(AUD_IMixer* mixer);
-
-	virtual AUD_Specs getSpecs();
+	virtual AUD_DeviceSpecs getSpecs();
 	virtual AUD_Handle* play(AUD_IFactory* factory, bool keep = false);
 	virtual bool pause(AUD_Handle* handle);
 	virtual bool resume(AUD_Handle* handle);

intern/audaspace/intern/AUD_Space.h

@@ -28,8 +28,10 @@
 
 /// The size of a format in bytes.
 #define AUD_FORMAT_SIZE(format) (format & 0x0F)
+/// The size of a sample in the specified device format in bytes.
+#define AUD_DEVICE_SAMPLE_SIZE(specs) (specs.channels * (specs.format & 0x0F))
 /// The size of a sample in the specified format in bytes.
-#define AUD_SAMPLE_SIZE(specs) (specs.channels * (specs.format & 0x0F))
+#define AUD_SAMPLE_SIZE(specs) (specs.channels * sizeof(sample_t))
 /// Throws a AUD_Exception with the provided error code.
 #define AUD_THROW(exception) { AUD_Exception e; e.error = exception; throw e; }
 
@@ -233,22 +235,42 @@ typedef enum
 	AUD_3DSS_CONE_OUTER_GAIN		/// Cone outer gain.
 } AUD_3DSourceSetting;
 
-/// Sample pointer type.
-typedef unsigned char sample_t;
+/// Sample type.(float samples)
+typedef float sample_t;
 
-/// Specification of a sound source or device.
+/// Sample data type (format samples)
+typedef unsigned char data_t;
+
+/// Specification of a sound source.
 typedef struct
 {
 	/// Sample rate in Hz.
 	AUD_SampleRate rate;
 
-	/// Sample format.
-	AUD_SampleFormat format;
-
 	/// Channel count.
 	AUD_Channels channels;
 } AUD_Specs;
 
+/// Specification of a sound device.
+typedef struct
+{
+	/// Sample format.
+	AUD_SampleFormat format;
+
+	union
+	{
+		struct
+		{
+			/// Sample rate in Hz.
+			AUD_SampleRate rate;
+
+			/// Channel count.
+			AUD_Channels channels;
+		};
+		AUD_Specs specs;
+	};
+} AUD_DeviceSpecs;
+
 /// Exception structure.
 typedef struct
 {

intern/audaspace/intern/AUD_StreamBufferFactory.cpp

@@ -61,9 +61,9 @@ AUD_StreamBufferFactory::AUD_StreamBufferFactory(AUD_IFactory* factory)
 		// read more
 		length = size-index;
 		reader->read(length, buffer);
-		memcpy(m_buffer.get()->getBuffer()+index*sample_size,
+		memcpy(m_buffer.get()->getBuffer() + index * m_specs.channels,
 			   buffer,
-			   length*sample_size);
+			   length * sample_size);
 		size += AUD_BUFFER_RESIZE_BYTES / sample_size;
 		index += length;
 	}

intern/audaspace/jack/AUD_JackDevice.cpp

@@ -23,7 +23,7 @@
  * ***** END LGPL LICENSE BLOCK *****
  */
 
-#include "AUD_FloatMixer.h"
+#include "AUD_Mixer.h"
 #include "AUD_JackDevice.h"
 #include "AUD_IReader.h"
 #include "AUD_Buffer.h"
@@ -38,7 +38,7 @@ int AUD_JackDevice::jack_mix(jack_nframes_t length, void *data)
 	unsigned int samplesize = AUD_SAMPLE_SIZE(device->m_specs);
 	if(device->m_buffer->getSize() < samplesize * length)
 		device->m_buffer->resize(samplesize * length);
-	device->mix(device->m_buffer->getBuffer(), length);
+	device->mix((data_t*)device->m_buffer->getBuffer(), length);
 
 	float* in = (float*) device->m_buffer->getBuffer();
 	float* out;
@@ -60,7 +60,7 @@ void AUD_JackDevice::jack_shutdown(void *data)
 	device->m_valid = false;
 }
 
-AUD_JackDevice::AUD_JackDevice(AUD_Specs specs)
+AUD_JackDevice::AUD_JackDevice(AUD_DeviceSpecs specs)
 {
 	if(specs.channels == AUD_CHANNELS_INVALID)
 		specs.channels = AUD_CHANNELS_STEREO;
@@ -123,9 +123,6 @@ AUD_JackDevice::AUD_JackDevice(AUD_Specs specs)
 		free(ports);
 	}
 
-	m_mixer = new AUD_FloatMixer(); AUD_NEW("mixer")
-	m_mixer->setSpecs(m_specs);
-
 	m_valid = true;
 
 	create();

intern/audaspace/jack/AUD_JackDevice.h

@@ -81,7 +81,7 @@ public:
 	 * \param specs The wanted audio specification, where only the channel count is important.
 	 * \exception AUD_Exception Thrown if the audio device cannot be opened.
 	 */
-	AUD_JackDevice(AUD_Specs specs);
+	AUD_JackDevice(AUD_DeviceSpecs specs);
 
 	/**
 	 * Closes the Jack client.

intern/audaspace/sndfile/AUD_SndFileReader.cpp

@@ -28,31 +28,14 @@
 
 #include <cstring>
 
-// This function transforms a  SampleFormat to our own sample format
-static inline AUD_SampleFormat SNDFILE_TO_AUD(int fmt)
-{
-	switch(fmt & SF_FORMAT_SUBMASK)
-	{
-	// only read s16, s32 and double as they are
-	case SF_FORMAT_PCM_16:
-		return AUD_FORMAT_S16;
-	case SF_FORMAT_PCM_32:
-		return AUD_FORMAT_S32;
-	case SF_FORMAT_DOUBLE:
-		return AUD_FORMAT_FLOAT64;
-	// read all other formats as floats
-	default:
-		return AUD_FORMAT_FLOAT32;
-	}
-}
-
 sf_count_t AUD_SndFileReader::vio_get_filelen(void *user_data)
 {
 	AUD_SndFileReader* reader = (AUD_SndFileReader*)user_data;
 	return reader->m_membuffer.get()->getSize();
 }
 
-sf_count_t AUD_SndFileReader::vio_seek(sf_count_t offset, int whence, void *user_data)
+sf_count_t AUD_SndFileReader::vio_seek(sf_count_t offset, int whence,
+									   void *user_data)
 {
 	AUD_SndFileReader* reader = (AUD_SndFileReader*)user_data;
 
@@ -72,14 +55,16 @@ sf_count_t AUD_SndFileReader::vio_seek(sf_count_t offset, int whence, void *user
 	return reader->m_memoffset;
 }
 
-sf_count_t AUD_SndFileReader::vio_read(void *ptr, sf_count_t count, void *user_data)
+sf_count_t AUD_SndFileReader::vio_read(void *ptr, sf_count_t count,
+									   void *user_data)
 {
 	AUD_SndFileReader* reader = (AUD_SndFileReader*)user_data;
 
 	if(reader->m_memoffset + count > reader->m_membuffer.get()->getSize())
 		count = reader->m_membuffer.get()->getSize() - reader->m_memoffset;
 
-	memcpy(ptr, reader->m_membuffer.get()->getBuffer() + reader->m_memoffset, count);
+	memcpy(ptr, ((data_t*)reader->m_membuffer.get()->getBuffer()) +
+		   reader->m_memoffset, count);
 	reader->m_memoffset += count;
 
 	return count;
@@ -103,27 +88,11 @@ AUD_SndFileReader::AUD_SndFileReader(const char* filename)
 		AUD_THROW(AUD_ERROR_FILE);
 
 	m_specs.channels = (AUD_Channels) sfinfo.channels;
-	m_specs.format = SNDFILE_TO_AUD(sfinfo.format);
 	m_specs.rate = (AUD_SampleRate) sfinfo.samplerate;
 	m_length = sfinfo.frames;
 	m_seekable = sfinfo.seekable;
 	m_position = 0;
 
-	switch(m_specs.format)
-	{
-	case AUD_FORMAT_S16:
-		m_read = (sf_read_f) sf_readf_short;
-		break;
-	case AUD_FORMAT_S32:
-		m_read = (sf_read_f) sf_readf_int;
-		break;
-	case AUD_FORMAT_FLOAT64:
-		m_read = (sf_read_f) sf_readf_double;
-		break;
-	default:
-		m_read = (sf_read_f) sf_readf_float;
-	}
-
 	m_buffer = new AUD_Buffer(); AUD_NEW("buffer")
 }
 
@@ -147,27 +116,11 @@ AUD_SndFileReader::AUD_SndFileReader(AUD_Reference<AUD_Buffer> buffer)
 		AUD_THROW(AUD_ERROR_FILE);
 
 	m_specs.channels = (AUD_Channels) sfinfo.channels;
-	m_specs.format = SNDFILE_TO_AUD(sfinfo.format);
 	m_specs.rate = (AUD_SampleRate) sfinfo.samplerate;
 	m_length = sfinfo.frames;
 	m_seekable = sfinfo.seekable;
 	m_position = 0;
 
-	switch(m_specs.format)
-	{
-	case AUD_FORMAT_S16:
-		m_read = (sf_read_f) sf_readf_short;
-		break;
-	case AUD_FORMAT_S32:
-		m_read = (sf_read_f) sf_readf_int;
-		break;
-	case AUD_FORMAT_FLOAT64:
-		m_read = (sf_read_f) sf_readf_double;
-		break;
-	default:
-		m_read = (sf_read_f) sf_readf_float;
-	}
-
 	m_buffer = new AUD_Buffer(); AUD_NEW("buffer")
 }
 
@@ -227,7 +180,7 @@ void AUD_SndFileReader::read(int & length, sample_t* & buffer)
 
 	buffer = m_buffer->getBuffer();
 
-	length = m_read(m_sndfile, buffer, length);
+	length = sf_readf_float(m_sndfile, buffer, length);
 
 	m_position += length;
 }

intern/audaspace/sndfile/AUD_SndFileReader.h

@@ -70,11 +70,6 @@ private:
 	 */
 	SNDFILE* m_sndfile;
 
-	/**
-	 * The reading function.
-	 */
-	sf_read_f m_read;
-
 	/**
 	 * The virtual IO structure for memory file reading.
 	 */

source/blender/blenkernel/BKE_sound.h

@@ -73,7 +73,7 @@ void sound_update_playing(struct bContext *C);
 void sound_scrub(struct bContext *C);
 
 #ifdef AUD_CAPI
-AUD_Device* sound_mixdown(struct Scene *scene, AUD_Specs specs, int start, int end, float volume);
+AUD_Device* sound_mixdown(struct Scene *scene, AUD_DeviceSpecs specs, int start, int end, float volume);
 #endif
 
 void sound_stop_all(struct bContext *C);

source/blender/blenkernel/intern/sequencer.c

@@ -3792,7 +3792,7 @@ Sequence *sequencer_add_sound_strip(bContext *C, ListBase *seqbasep, SeqLoadInfo
 
 	info = AUD_getInfo(sound->handle);
 
-	if (info.specs.format == AUD_FORMAT_INVALID) {
+	if (info.specs.channels == AUD_CHANNELS_INVALID) {
 		sound_delete(C, sound);
 		//if(op)
 		//	BKE_report(op->reports, RPT_ERROR, "Unsupported audio format");

source/blender/blenkernel/intern/sound.c

@@ -40,7 +40,7 @@ void sound_disable()
 
 void sound_init()
 {
-	AUD_Specs specs;
+	AUD_DeviceSpecs specs;
 	int device, buffersize;
 
 	device = U.audiodevice;
@@ -455,7 +455,7 @@ void sound_scrub(struct bContext *C)
 	}
 }
 
-AUD_Device* sound_mixdown(struct Scene *scene, AUD_Specs specs, int start, int end, float volume)
+AUD_Device* sound_mixdown(struct Scene *scene, AUD_DeviceSpecs specs, int start, int end, float volume)
 {
 	AUD_Device* mixdown = AUD_openReadDevice(specs);
 	SoundHandle *handle;

source/blender/blenkernel/intern/writeffmpeg.c

@@ -832,7 +832,7 @@ int start_ffmpeg(struct Scene *scene, RenderData *rd, int rectx, int recty, Repo
 	if(ffmpeg_multiplex_audio && audio_stream)
 	{
 		AVCodecContext* c = get_codec_from_stream(audio_stream);
-		AUD_Specs specs;
+		AUD_DeviceSpecs specs;
 		specs.channels = c->channels;
 		specs.format = AUD_FORMAT_S16;
 		specs.rate = rd->ffcodecdata.audio_mixrate;

source/blender/editors/sound/sound_ops.c

@@ -81,7 +81,7 @@ static int open_exec(bContext *C, wmOperator *op)
 
 	info = AUD_getInfo(sound->handle);
 
-	if (info.specs.format == AUD_FORMAT_INVALID) {
+	if (info.specs.channels == AUD_CHANNELS_INVALID) {
 		sound_delete(C, sound);
 		BKE_report(op->reports, RPT_ERROR, "Unsupported audio format");
 		return OPERATOR_CANCELLED;

source/blender/editors/space_graph/graph_edit.c

@@ -36,6 +36,8 @@
 #include <config.h>
 #endif
 
+#include "AUD_C-API.h"
+
 #include "MEM_guardedalloc.h"
 
 #include "BLI_blenlib.h"
@@ -990,6 +992,139 @@ void GRAPH_OT_bake (wmOperatorType *ot)
 	// todo: add props for start/end frames
 }
 
+/* ******************** Sound Bake F-Curve Operator *********************** */
+/* This operator bakes the given sound to the selected F-Curves */
+
+/* ------------------- */
+
+/* Custom data storage passed to the F-Sample-ing function,
+ * which provides the necessary info for baking the sound
+ */
+typedef struct tSoundBakeInfo {
+	float* samples;
+	int length;
+	int cfra;
+} tSoundBakeInfo;
+
+/* ------------------- */
+
+/* Sampling callback used to determine the value from the sound to
+ * save in the F-Curve at the specified frame
+ */
+static float fcurve_samplingcb_sound (FCurve *fcu, void *data, float evaltime)
+{
+	tSoundBakeInfo *sbi= (tSoundBakeInfo *)data;
+
+	int position = evaltime - sbi->cfra;
+	if((position < 0) || (position >= sbi->length))
+		return 0.0f;
+
+	return sbi->samples[position];
+}
+
+/* ------------------- */
+
+static int graphkeys_sound_bake_exec(bContext *C, wmOperator *op)
+{
+	bAnimContext ac;
+	ListBase anim_data = {NULL, NULL};
+	bAnimListElem *ale;
+	int filter;
+
+	tSoundBakeInfo sbi;
+	Scene *scene= NULL;
+	int start, end;
+
+	char path[FILE_MAX];
+
+	/* get editor data */
+	if (ANIM_animdata_get_context(C, &ac) == 0)
+		return OPERATOR_CANCELLED;
+
+	RNA_string_get(op->ptr, "path", path);
+
+	scene= ac.scene;	/* current scene */
+
+	/* store necessary data for the baking steps */
+	sbi.samples = AUD_readSoundBuffer(path,
+									  RNA_float_get(op->ptr, "low"),
+									  RNA_float_get(op->ptr, "high"),
+									  RNA_float_get(op->ptr, "attack"),
+									  RNA_float_get(op->ptr, "release"),
+									  RNA_float_get(op->ptr, "threshold"),
+									  FPS, &sbi.length);
+
+	if (sbi.samples == NULL) {
+		BKE_report(op->reports, RPT_ERROR, "Unsupported audio format");
+		return OPERATOR_CANCELLED;
+	}
+
+	/* determine extents of the baking */
+	sbi.cfra = start = CFRA;
+	end = CFRA + sbi.length - 1;
+
+	/* filter anim channels */
+	filter= (ANIMFILTER_VISIBLE | ANIMFILTER_CURVEVISIBLE | ANIMFILTER_SEL | ANIMFILTER_FOREDIT | ANIMFILTER_CURVESONLY);
+	ANIM_animdata_filter(&ac, &anim_data, filter, ac.data, ac.datatype);
+
+	/* loop through all selected F-Curves, replacing its data with the sound samples */
+	for (ale= anim_data.first; ale; ale= ale->next) {
+		FCurve *fcu= (FCurve *)ale->key_data;
+
+		/* sample the sound */
+		fcurve_store_samples(fcu, &sbi, start, end, fcurve_samplingcb_sound);
+	}
+
+	/* free sample data */
+	free(sbi.samples);
+
+	/* admin and redraws */
+	BLI_freelistN(&anim_data);
+
+	/* validate keyframes after editing */
+	ANIM_editkeyframes_refresh(&ac);
+
+	/* set notifier that 'keyframes' have changed */
+	WM_event_add_notifier(C, NC_ANIMATION|ND_KEYFRAME_EDIT, NULL);
+
+	return OPERATOR_FINISHED;
+}
+
+static int graphkeys_sound_bake_invoke (bContext *C, wmOperator *op, wmEvent *event)
+{
+	bAnimContext ac;
+
+	/* verify editor data */
+	if (ANIM_animdata_get_context(C, &ac) == 0)
+		return OPERATOR_CANCELLED;
+
+	return WM_operator_filesel(C, op, event);
+}
+
+void GRAPH_OT_sound_bake (wmOperatorType *ot)
+{
+	/* identifiers */
+	ot->name= "Bake Sound to F-Curves";
+	ot->idname= "GRAPH_OT_sound_bake";
+	ot->description= "Bakes a sound wave to selected F-Curves.";
+
+	/* api callbacks */
+	ot->invoke= graphkeys_sound_bake_invoke;
+	ot->exec= graphkeys_sound_bake_exec;
+	ot->poll= graphop_selected_fcurve_poll;
+
+	/* flags */
+	ot->flag= OPTYPE_REGISTER|OPTYPE_UNDO;
+
+	/* properties */
+	WM_operator_properties_filesel(ot, FOLDERFILE|SOUNDFILE|MOVIEFILE, FILE_SPECIAL);
+	RNA_def_float(ot->srna, "low", 0.0f, 0.0, 100000.0, "Lowest frequency", "", 0.1, 1000.00);
+	RNA_def_float(ot->srna, "high", 100000.0, 0.0, 100000.0, "Highest frequency", "", 0.1, 1000.00);
+	RNA_def_float(ot->srna, "attack", 0.005, 0.0, 2.0, "Attack time", "", 0.01, 0.1);
+	RNA_def_float(ot->srna, "release", 0.2, 0.0, 5.0, "Release time", "", 0.01, 0.2);
+	RNA_def_float(ot->srna, "threshold", 0.0, 0.0, 1.0, "Threshold", "", 0.01, 0.1);
+}
+
 /* ******************** Sample Keyframes Operator *********************** */
 /* This operator 'bakes' the values of the curve into new keyframes between pairs
  * of selected keyframes. It is useful for creating keyframes for tweaking overlap.

source/blender/editors/space_graph/graph_intern.h

@@ -100,6 +100,7 @@ void GRAPH_OT_delete(struct wmOperatorType *ot);
 void GRAPH_OT_clean(struct wmOperatorType *ot);
 void GRAPH_OT_sample(struct wmOperatorType *ot);
 void GRAPH_OT_bake(struct wmOperatorType *ot);
+void GRAPH_OT_sound_bake(struct wmOperatorType *ot);
 void GRAPH_OT_smooth(struct wmOperatorType *ot);
 
 void GRAPH_OT_handle_type(struct wmOperatorType *ot);

source/blender/editors/space_graph/graph_ops.c

@@ -247,6 +247,7 @@ void graphedit_operatortypes(void)
 	WM_operatortype_append(GRAPH_OT_extrapolation_type);
 	WM_operatortype_append(GRAPH_OT_sample);
 	WM_operatortype_append(GRAPH_OT_bake);
+	WM_operatortype_append(GRAPH_OT_sound_bake);
 	WM_operatortype_append(GRAPH_OT_smooth);
 	WM_operatortype_append(GRAPH_OT_clean);
 	WM_operatortype_append(GRAPH_OT_delete);