deleted file mode 100644

@@ -1,142 +0,0 @@

-////////////////////////////////////////////////////////////////////////////////

-///

-/// Sample rate transposer. Changes sample rate by using linear interpolation

-/// together with anti-alias filtering (first order interpolation with anti-

-/// alias filtering should be quite adequate for this application).

-///

-/// Use either of the derived classes of 'RateTransposerInteger' or

-/// 'RateTransposerFloat' for corresponding integer/floating point tranposing

-/// algorithm implementation.

-///

-/// Author        : Copyright (c) Olli Parviainen

-/// Author e-mail : oparviai 'at' iki.fi

-/// SoundTouch WWW: http://www.surina.net/soundtouch

-///

-////////////////////////////////////////////////////////////////////////////////

-//

-// Last changed  : $Date: 2009-02-21 13:00:14 -0300 (s�b, 21 fev 2009) $

-// File revision : $Revision: 4 $

-//

-// $Id: RateTransposer.h 63 2009-02-21 16:00:14Z oparviai $

-//

-////////////////////////////////////////////////////////////////////////////////

-//

-// License :

-//

-//  SoundTouch audio processing library

-//  Copyright (c) Olli Parviainen

-//

-//  This library 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 2.1 of the License, or (at your option) any later version.

-//

-//  This library 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 this library; if not, write to the Free Software

-//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

-//

-////////////////////////////////////////////////////////////////////////////////

-

-#pragma once

-

-#include "AAFilter.h"

-#include "FIFOSampleBuffer.h"

-

-namespace soundtouch

-{

-

-/// A common linear samplerate transposer class.

-///

-/// Note: Use function "RateTransposer::newInstance()" to create a new class

-/// instance instead of the "new" operator; that function automatically

-/// chooses a correct implementation depending on if integer or floating

-/// arithmetics are to be used.

-class RateTransposer : public FIFOProcessor

-{

-protected:

-	/// Anti-alias filter object

-	std::unique_ptr<AAFilter> pAAFilter;

-

-	float fRate;

-

-	int32_t numChannels;

-

-	/// Buffer for collecting samples to feed the anti-alias filter between

-	/// two batches

-	FIFOSampleBuffer storeBuffer;

-

-	/// Buffer for keeping samples between transposing & anti-alias filter

-	FIFOSampleBuffer tempBuffer;

-

-	/// Output sample buffer

-	FIFOSampleBuffer outputBuffer;

-

-	bool bUseAAFilter;

-

-	virtual void resetRegisters() = 0;

-

-	virtual uint32_t transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint32_t numSamples) = 0;

-	virtual uint32_t transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint32_t numSamples) = 0;

-	uint32_t transpose(SAMPLETYPE *dest, const SAMPLETYPE *src, uint32_t numSamples);

-

-	void downsample(const SAMPLETYPE *src, uint32_t numSamples);

-	void upsample(const SAMPLETYPE *src, uint32_t numSamples);

-

-	/// Transposes sample rate by applying anti-alias filter to prevent folding.

-	/// Returns amount of samples returned in the "dest" buffer.

-	/// The maximum amount of samples that can be returned at a time is set by

-	/// the 'set_returnBuffer_size' function.

-	void processSamples(const SAMPLETYPE *src, uint32_t numSamples);

-

-public:

-	RateTransposer();

-	virtual ~RateTransposer();

-

-	/// Operator 'new' is overloaded so that it automatically creates a suitable instance

-	/// depending on if we're to use integer or floating point arithmetics.

-	static void *operator new(size_t s);

-

-	/// Use this function instead of "new" operator to create a new instance of this class.

-	/// This function automatically chooses a correct implementation, depending on if

-	/// integer ot floating point arithmetics are to be used.

-	static RateTransposer *newInstance();

-

-	/// Returns the output buffer object

-	FIFOSamplePipe *getOutput() { return &this->outputBuffer; }

-

-	/// Returns the store buffer object

-	FIFOSamplePipe *getStore() { return &this->storeBuffer; }

-

-	/// Return anti-alias filter object

-	AAFilter *getAAFilter();

-

-	/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable

-	void enableAAFilter(bool newMode);

-

-	/// Returns nonzero if anti-alias filter is enabled.

-	bool isAAFilterEnabled() const;

-

-	/// Sets new target rate. Normal rate = 1.0, smaller values represent slower

-	/// rate, larger faster rates.

-	virtual void setRate(float newRate);

-

-	/// Sets the number of channels, 1 = mono, 2 = stereo

-	void setChannels(int32_t channels);

-

-	/// Adds 'numSamples' pcs of samples from the 'samples' memory position into

-	/// the input of the object.

-	void putSamples(const SAMPLETYPE *samples, uint32_t numSamples);

-

-	/// Clears all the samples in the object

-	void clear();

-

-	/// Returns nonzero if there aren't any samples available for outputting.

-	bool isEmpty() const;

-};

-

-}

@@ -42,8 +42,7 @@

 //

 ////////////////////////////////////////////////////////////////////////////////

-#ifndef RateTransposer_H

-#define RateTransposer_H

+#pragma once

 #include "AAFilter.h"

 #include "FIFOSampleBuffer.h"

@@ -141,5 +140,3 @@ public:

 };

 }

-

-#endif

@@ -46,7 +46,6 @@

 #define RateTransposer_H

 #include "AAFilter.h"

-#include "FIFOSamplePipe.h"

 #include "FIFOSampleBuffer.h"

 namespace soundtouch

@@ -14,10 +14,10 @@

 ///

 ////////////////////////////////////////////////////////////////////////////////

 //

-// Last changed  : $Date: 2006/02/05 16:44:06 $

-// File revision : $Revision: 1.10 $

+// Last changed  : $Date: 2009-02-21 13:00:14 -0300 (s�b, 21 fev 2009) $

+// File revision : $Revision: 4 $

 //

-// $Id: RateTransposer.h,v 1.10 2006/02/05 16:44:06 Olli Exp $

+// $Id: RateTransposer.h 63 2009-02-21 16:00:14Z oparviai $

 //

 ////////////////////////////////////////////////////////////////////////////////

 //

@@ -49,8 +49,6 @@

 #include "FIFOSamplePipe.h"

 #include "FIFOSampleBuffer.h"

-#include "STTypes.h"

-

 namespace soundtouch

 {

@@ -64,11 +62,11 @@ class RateTransposer : public FIFOProcessor

 {

 protected:

 	/// Anti-alias filter object

-	AAFilter *pAAFilter;

+	std::unique_ptr<AAFilter> pAAFilter;

 	float fRate;

-	uint32_t uChannels;

+	int32_t numChannels;

 	/// Buffer for collecting samples to feed the anti-alias filter between

 	/// two batches

@@ -82,16 +80,12 @@ protected:

 	bool bUseAAFilter;

-	void init();

-

 	virtual void resetRegisters() = 0;

 	virtual uint32_t transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint32_t numSamples) = 0;

 	virtual uint32_t transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint32_t numSamples) = 0;

 	uint32_t transpose(SAMPLETYPE *dest, const SAMPLETYPE *src, uint32_t numSamples);

-	void flushStoreBuffer();

-

 	void downsample(const SAMPLETYPE *src, uint32_t numSamples);

 	void upsample(const SAMPLETYPE *src, uint32_t numSamples);

@@ -107,7 +101,7 @@ public:

 	/// Operator 'new' is overloaded so that it automatically creates a suitable instance

 	/// depending on if we're to use integer or floating point arithmetics.

-	void *operator new(size_t s);

+	static void *operator new(size_t s);

 	/// Use this function instead of "new" operator to create a new instance of this class.

 	/// This function automatically chooses a correct implementation, depending on if

@@ -115,13 +109,13 @@ public:

 	static RateTransposer *newInstance();

 	/// Returns the output buffer object

-	FIFOSamplePipe *getOutput() { return &outputBuffer; };

+	FIFOSamplePipe *getOutput() { return &this->outputBuffer; }

 	/// Returns the store buffer object

-	FIFOSamplePipe *getStore() { return &storeBuffer; };

+	FIFOSamplePipe *getStore() { return &this->storeBuffer; }

 	/// Return anti-alias filter object

-	AAFilter *getAAFilter() const;

+	AAFilter *getAAFilter();

 	/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable

 	void enableAAFilter(bool newMode);

@@ -134,7 +128,7 @@ public:

 	virtual void setRate(float newRate);

 	/// Sets the number of channels, 1 = mono, 2 = stereo

-	void setChannels(uint32_t channels);

+	void setChannels(int32_t channels);

 	/// Adds 'numSamples' pcs of samples from the 'samples' memory position into

 	/// the input of the object.

new file mode 100644

@@ -0,0 +1,152 @@

+////////////////////////////////////////////////////////////////////////////////

+///

+/// Sample rate transposer. Changes sample rate by using linear interpolation

+/// together with anti-alias filtering (first order interpolation with anti-

+/// alias filtering should be quite adequate for this application).

+///

+/// Use either of the derived classes of 'RateTransposerInteger' or

+/// 'RateTransposerFloat' for corresponding integer/floating point tranposing

+/// algorithm implementation.

+///

+/// Author        : Copyright (c) Olli Parviainen

+/// Author e-mail : oparviai 'at' iki.fi

+/// SoundTouch WWW: http://www.surina.net/soundtouch

+///

+////////////////////////////////////////////////////////////////////////////////

+//

+// Last changed  : $Date: 2006/02/05 16:44:06 $

+// File revision : $Revision: 1.10 $

+//

+// $Id: RateTransposer.h,v 1.10 2006/02/05 16:44:06 Olli Exp $

+//

+////////////////////////////////////////////////////////////////////////////////

+//

+// License :

+//

+//  SoundTouch audio processing library

+//  Copyright (c) Olli Parviainen

+//

+//  This library 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 2.1 of the License, or (at your option) any later version.

+//

+//  This library 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 this library; if not, write to the Free Software

+//  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA

+//

+////////////////////////////////////////////////////////////////////////////////

+

+#ifndef RateTransposer_H

+#define RateTransposer_H

+

+#include "AAFilter.h"

+#include "FIFOSamplePipe.h"

+#include "FIFOSampleBuffer.h"

+

+#include "STTypes.h"

+

+namespace soundtouch

+{

+

+/// A common linear samplerate transposer class.

+///

+/// Note: Use function "RateTransposer::newInstance()" to create a new class

+/// instance instead of the "new" operator; that function automatically

+/// chooses a correct implementation depending on if integer or floating

+/// arithmetics are to be used.

+class RateTransposer : public FIFOProcessor

+{

+protected:

+	/// Anti-alias filter object

+	AAFilter *pAAFilter;

+

+	float fRate;

+

+	uint32_t uChannels;

+

+	/// Buffer for collecting samples to feed the anti-alias filter between

+	/// two batches

+	FIFOSampleBuffer storeBuffer;

+

+	/// Buffer for keeping samples between transposing & anti-alias filter

+	FIFOSampleBuffer tempBuffer;

+

+	/// Output sample buffer

+	FIFOSampleBuffer outputBuffer;

+

+	bool bUseAAFilter;

+

+	void init();

+

+	virtual void resetRegisters() = 0;

+

+	virtual uint32_t transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint32_t numSamples) = 0;

+	virtual uint32_t transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint32_t numSamples) = 0;

+	uint32_t transpose(SAMPLETYPE *dest, const SAMPLETYPE *src, uint32_t numSamples);

+

+	void flushStoreBuffer();

+

+	void downsample(const SAMPLETYPE *src, uint32_t numSamples);

+	void upsample(const SAMPLETYPE *src, uint32_t numSamples);

+

+	/// Transposes sample rate by applying anti-alias filter to prevent folding.

+	/// Returns amount of samples returned in the "dest" buffer.

+	/// The maximum amount of samples that can be returned at a time is set by

+	/// the 'set_returnBuffer_size' function.

+	void processSamples(const SAMPLETYPE *src, uint32_t numSamples);

+

+public:

+	RateTransposer();

+	virtual ~RateTransposer();

+

+	/// Operator 'new' is overloaded so that it automatically creates a suitable instance

+	/// depending on if we're to use integer or floating point arithmetics.

+	void *operator new(size_t s);

+

+	/// Use this function instead of "new" operator to create a new instance of this class.

+	/// This function automatically chooses a correct implementation, depending on if

+	/// integer ot floating point arithmetics are to be used.

+	static RateTransposer *newInstance();

+

+	/// Returns the output buffer object

+	FIFOSamplePipe *getOutput() { return &outputBuffer; };

+

+	/// Returns the store buffer object

+	FIFOSamplePipe *getStore() { return &storeBuffer; };

+

+	/// Return anti-alias filter object

+	AAFilter *getAAFilter() const;

+

+	/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable

+	void enableAAFilter(bool newMode);

+

+	/// Returns nonzero if anti-alias filter is enabled.

+	bool isAAFilterEnabled() const;

+

+	/// Sets new target rate. Normal rate = 1.0, smaller values represent slower

+	/// rate, larger faster rates.

+	virtual void setRate(float newRate);

+

+	/// Sets the number of channels, 1 = mono, 2 = stereo

+	void setChannels(uint32_t channels);

+

+	/// Adds 'numSamples' pcs of samples from the 'samples' memory position into

+	/// the input of the object.

+	void putSamples(const SAMPLETYPE *samples, uint32_t numSamples);

+

+	/// Clears all the samples in the object

+	void clear();

+

+	/// Returns nonzero if there aren't any samples available for outputting.

+	bool isEmpty() const;

+};

+

+}

+

+#endif