deleted file mode 100644

@@ -1,527 +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)

-///

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

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

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

-///

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

-//

-// Last changed  : $Date: 2011-09-02 15:56:11 -0300 (sex, 02 set 2011) $

-// File revision : $Revision: 4 $

-//

-// $Id: RateTransposer.cpp 131 2011-09-02 18:56:11Z 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

-//

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

-

-#include <stdexcept>

-#include "RateTransposer.h"

-

-using namespace soundtouch;

-

-/// A linear samplerate transposer class that uses integer arithmetics.

-/// for the transposing.

-class RateTransposerInteger : public RateTransposer

-{

-protected:

-	int iSlopeCount;

-	int iRate;

-	SAMPLETYPE sPrevSampleL, sPrevSampleR;

-

-	virtual void resetRegisters();

-

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

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

-

-public:

-	RateTransposerInteger();

-	virtual ~RateTransposerInteger();

-

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

-	/// rate, larger faster rates.

-	virtual void setRate(float newRate);

-};

-

-/// A linear samplerate transposer class that uses floating point arithmetics

-/// for the transposing.

-class RateTransposerFloat : public RateTransposer

-{

-protected:

-	float fSlopeCount;

-	SAMPLETYPE sPrevSampleL, sPrevSampleR;

-

-	virtual void resetRegisters();

-

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

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

-

-public:

-	RateTransposerFloat();

-	virtual ~RateTransposerFloat();

-};

-

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

-// depending on if we've a MMX/SSE/etc-capable CPU available or not.

-void *RateTransposer::operator new(size_t)

-{

-	// Notice! don't use "new TDStretch" directly, use "newInstance" to create a new instance instead!

-	//assert(false);

-	//return NULL;

-	throw std::runtime_error("Don't use 'new RateTransposer', use 'newInstance' member instead!");

-}

-

-RateTransposer *RateTransposer::newInstance()

-{

-#ifdef SOUNDTOUCH_INTEGER_SAMPLES

-	return ::new RateTransposerInteger;

-#else

-	return ::new RateTransposerFloat;

-#endif

-}

-

-// Constructor

-RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer)

-{

-	this->numChannels = 2;

-	this->bUseAAFilter = true;

-	this->fRate = 0;

-

-	// Instantiates the anti-alias filter with default tap length

-	// of 32

-	this->pAAFilter.reset(new AAFilter(32));

-}

-

-RateTransposer::~RateTransposer()

-{

-}

-

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

-void RateTransposer::enableAAFilter(bool newMode)

-{

-	this->bUseAAFilter = newMode;

-}

-

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

-bool RateTransposer::isAAFilterEnabled() const

-{

-	return this->bUseAAFilter;

-}

-

-AAFilter *RateTransposer::getAAFilter()

-{

-	return this->pAAFilter.get();

-}

-

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

-// iRate, larger faster iRates.

-void RateTransposer::setRate(float newRate)

-{

-	double fCutoff;

-

-	this->fRate = newRate;

-

-	// design a new anti-alias filter

-	if (newRate > 1.0f)

-		fCutoff = 0.5f / newRate;

-	else

-		fCutoff = 0.5f * newRate;

-    this->pAAFilter->setCutoffFreq(fCutoff);

-}

-

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

-// the input of the object.

-void RateTransposer::putSamples(const SAMPLETYPE *samples, uint32_t nSamples)

-{

-	this->processSamples(samples, nSamples);

-}

-

-// Transposes up the sample rate, causing the observed playback 'rate' of the

-// sound to decrease

-void RateTransposer::upsample(const SAMPLETYPE *src, uint32_t nSamples)

-{

-	// If the parameter 'uRate' value is smaller than 'SCALE', first transpose

-	// the samples and then apply the anti-alias filter to remove aliasing.

-

-	// First check that there's enough room in 'storeBuffer'

-	// (+16 is to reserve some slack in the destination buffer)

-	uint32_t sizeTemp = static_cast<uint32_t>(nSamples / this->fRate + 16.0f);

-

-	// Transpose the samples, store the result into the end of "storeBuffer"

-	uint32_t count = this->transpose(this->storeBuffer.ptrEnd(sizeTemp), src, nSamples);

-	this->storeBuffer.putSamples(count);

-

-	// Apply the anti-alias filter to samples in "store output", output the

-	// result to "dest"

-	uint32_t num = this->storeBuffer.numSamples();

-	count = this->pAAFilter->evaluate(this->outputBuffer.ptrEnd(num), this->storeBuffer.ptrBegin(), num, this->numChannels);

-	this->outputBuffer.putSamples(count);

-

-	// Remove the processed samples from "storeBuffer"

-	this->storeBuffer.receiveSamples(count);

-}

-

-// Transposes down the sample rate, causing the observed playback 'rate' of the

-// sound to increase

-void RateTransposer::downsample(const SAMPLETYPE *src, uint32_t nSamples)

-{

-	// If the parameter 'uRate' value is larger than 'SCALE', first apply the

-	// anti-alias filter to remove high frequencies (prevent them from folding

-	// over the lover frequencies), then transpose. */

-

-	// Add the new samples to the end of the storeBuffer */

-	this->storeBuffer.putSamples(src, nSamples);

-

-	// Anti-alias filter the samples to prevent folding and output the filtered

-	// data to tempBuffer. Note : because of the FIR filter length, the

-	// filtering routine takes in 'filter_length' more samples than it outputs.

-	assert(this->tempBuffer.isEmpty());

-	uint32_t sizeTemp = this->storeBuffer.numSamples();

-

-	uint32_t count = this->pAAFilter->evaluate(this->tempBuffer.ptrEnd(sizeTemp), this->storeBuffer.ptrBegin(), sizeTemp, this->numChannels);

-

-	// Remove the filtered samples from 'storeBuffer'

-	this->storeBuffer.receiveSamples(count);

-

-	// Transpose the samples (+16 is to reserve some slack in the destination buffer)

-	sizeTemp = static_cast<uint32_t>(nSamples / this->fRate + 16.0f);

-	count = this->transpose(this->outputBuffer.ptrEnd(sizeTemp), this->tempBuffer.ptrBegin(), count);

-	this->outputBuffer.putSamples(count);

-}

-

-// 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 RateTransposer::processSamples(const SAMPLETYPE *src, uint32_t nSamples)

-{

-	if (!nSamples)

-		return;

-	assert(this->pAAFilter);

-

-	// If anti-alias filter is turned off, simply transpose without applying

-	// the filter

-	if (!bUseAAFilter)

-	{

-		uint32_t sizeReq = static_cast<uint32_t>(nSamples / this->fRate + 1.0f);

-		uint32_t count = this->transpose(this->outputBuffer.ptrEnd(sizeReq), src, nSamples);

-		this->outputBuffer.putSamples(count);

-		return;

-	}

-

-	// Transpose with anti-alias filter

-	if (this->fRate < 1.0f)

-		this->upsample(src, nSamples);

-	else

-		this->downsample(src, nSamples);

-}

-

-// Transposes the sample rate of the given samples using linear interpolation.

-// Returns the number of samples returned in the "dest" buffer

-uint32_t RateTransposer::transpose(SAMPLETYPE *dest, const SAMPLETYPE *src, uint32_t nSamples)

-{

-	if (this->numChannels == 2)

-		return this->transposeStereo(dest, src, nSamples);

-	else

-		return this->transposeMono(dest, src, nSamples);

-}

-

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

-void RateTransposer::setChannels(int32_t nChannels)

-{

-	assert(nChannels > 0);

-	if (this->numChannels == nChannels)

-		return;

-

-	assert(nChannels == 1 || nChannels == 2);

-	this->numChannels = nChannels;

-

-	this->storeBuffer.setChannels(this->numChannels);

-	this->tempBuffer.setChannels(this->numChannels);

-	this->outputBuffer.setChannels(this->numChannels);

-

-	// Inits the linear interpolation registers

-	this->resetRegisters();

-}

-

-// Clears all the samples in the object

-void RateTransposer::clear()

-{

-	this->outputBuffer.clear();

-	this->storeBuffer.clear();

-}

-

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

-bool RateTransposer::isEmpty() const

-{

-	bool res = FIFOProcessor::isEmpty();

-	if (!res)

-		return false;

-	return this->storeBuffer.isEmpty();

-}

-

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

-//

-// RateTransposerInteger - integer arithmetic implementation

-//

-

-/// fixed-point interpolation routine precision

-static const int SCALE = 65536;

-

-// Constructor

-RateTransposerInteger::RateTransposerInteger() : RateTransposer()

-{

-	// Notice: use local function calling syntax for sake of clarity,

-	// to indicate the fact that C++ constructor can't call virtual functions.

-	RateTransposerInteger::resetRegisters();

-	RateTransposerInteger::setRate(1.0f);

-}

-

-RateTransposerInteger::~RateTransposerInteger()

-{

-}

-

-void RateTransposerInteger::resetRegisters()

-{

-	this->iSlopeCount = 0;

-	this->sPrevSampleL = this->sPrevSampleR = 0;

-}

-

-// Transposes the sample rate of the given samples using linear interpolation.

-// 'Mono' version of the routine. Returns the number of samples returned in

-// the "dest" buffer

-uint32_t RateTransposerInteger::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint32_t nSamples)

-{

-	if (!nSamples)

-		return 0; // no samples, no work

-

-	unsigned used = 0, i = 0;

-

-	// Process the last sample saved from the previous call first...

-	LONG_SAMPLETYPE temp, vol1;

-	while (this->iSlopeCount <= SCALE)

-	{

-		vol1 = SCALE - this->iSlopeCount;

-		temp = vol1 * this->sPrevSampleL + this->iSlopeCount * src[0];

-		dest[i] = static_cast<SAMPLETYPE>(temp / SCALE);

-		++i;

-		this->iSlopeCount += this->iRate;

-	}

-	// now always (iSlopeCount > SCALE)

-	this->iSlopeCount -= SCALE;

-

-	while (1)

-	{

-		while (this->iSlopeCount > SCALE)

-		{

-			this->iSlopeCount -= SCALE;

-			++used;

-			if (used >= nSamples - 1)

-				goto end;

-		}

-		vol1 = SCALE - this->iSlopeCount;

-		temp = src[used] * vol1 + this->iSlopeCount * src[used + 1];

-		dest[i] = static_cast<SAMPLETYPE>(temp / SCALE);

-		++i;

-		this->iSlopeCount += this->iRate;

-	}

-end:

-	// Store the last sample for the next round

-	this->sPrevSampleL = src[nSamples - 1];

-

-	return i;

-}

-

-// Transposes the sample rate of the given samples using linear interpolation.

-// 'Stereo' version of the routine. Returns the number of samples returned in

-// the "dest" buffer

-uint32_t RateTransposerInteger::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint32_t nSamples)

-{

-	if (!nSamples)

-		return 0; // no samples, no work

-

-	unsigned used = 0, i = 0;

-

-	// Process the last sample saved from the sPrevSampleLious call first...

-	LONG_SAMPLETYPE temp, vol1;

-	while (this->iSlopeCount <= SCALE)

-	{

-		vol1 = SCALE - this->iSlopeCount;

-		temp = vol1 * this->sPrevSampleL + this->iSlopeCount * src[0];

-		dest[2 * i] = static_cast<SAMPLETYPE>(temp / SCALE);

-		temp = vol1 * this->sPrevSampleR + this->iSlopeCount * src[1];

-		dest[2 * i + 1] = static_cast<SAMPLETYPE>(temp / SCALE);

-		++i;

-		this->iSlopeCount += this->iRate;

-	}

-	// now always (iSlopeCount > SCALE)

-	this->iSlopeCount -= SCALE;

-

-	while (1)

-	{

-		while (this->iSlopeCount > SCALE)

-		{

-			this->iSlopeCount -= SCALE;

-			++used;

-			if (used >= nSamples - 1)

-				goto end;

-		}

-		unsigned srcPos = 2 * used;

-		vol1 = SCALE - this->iSlopeCount;

-		temp = src[srcPos] * vol1 + this->iSlopeCount * src[srcPos + 2];

-		dest[2 * i] = static_cast<SAMPLETYPE>(temp / SCALE);

-		temp = src[srcPos + 1] * vol1 + this->iSlopeCount * src[srcPos + 3];

-		dest[2 * i + 1] = static_cast<SAMPLETYPE>(temp / SCALE);

-		++i;

-		this->iSlopeCount += this->iRate;

-	}

-end:

-	// Store the last sample for the next round

-	this->sPrevSampleL = src[2 * nSamples - 2];

-	this->sPrevSampleR = src[2 * nSamples - 1];

-

-	return i;

-}

-

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

-// iRate, larger faster iRates.

-void RateTransposerInteger::setRate(float newRate)

-{

-	this->iRate = static_cast<int>(newRate * SCALE + 0.5f);

-	RateTransposer::setRate(newRate);

-}

-

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

-//

-// RateTransposerFloat - floating point arithmetic implementation

-//

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

-

-// Constructor

-RateTransposerFloat::RateTransposerFloat() : RateTransposer()

-{

-	// Notice: use local function calling syntax for sake of clarity,

-	// to indicate the fact that C++ constructor can't call virtual functions.

-	RateTransposerFloat::resetRegisters();

-	RateTransposerFloat::setRate(1.0f);

-}

-

-RateTransposerFloat::~RateTransposerFloat()

-{

-}

-

-void RateTransposerFloat::resetRegisters()

-{

-	this->fSlopeCount = 0;

-	this->sPrevSampleL = this->sPrevSampleR = 0;

-}

-

-// Transposes the sample rate of the given samples using linear interpolation.

-// 'Mono' version of the routine. Returns the number of samples returned in

-// the "dest" buffer

-uint32_t RateTransposerFloat::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint32_t nSamples)

-{

-	unsigned used = 0, i = 0;

-

-	// Process the last sample saved from the previous call first...

-	while (this->fSlopeCount <= 1.0f)

-	{

-		dest[i] = static_cast<SAMPLETYPE>((1.0f - this->fSlopeCount) * this->sPrevSampleL + this->fSlopeCount * src[0]);

-		++i;

-		this->fSlopeCount += this->fRate;

-	}

-	this->fSlopeCount -= 1.0f;

-

-	if (nSamples > 1)

-	{

-		while (1)

-		{

-			while (this->fSlopeCount > 1.0f)

-			{

-				this->fSlopeCount -= 1.0f;

-				++used;

-				if (used >= nSamples - 1)

-					goto end;

-			}

-			dest[i] = static_cast<SAMPLETYPE>((1.0f - this->fSlopeCount) * src[used] + this->fSlopeCount * src[used + 1]);

-			++i;

-			this->fSlopeCount += this->fRate;

-		}

-	}

-end:

-	// Store the last sample for the next round

-	this->sPrevSampleL = src[nSamples - 1];

-

-	return i;

-}

-

-// Transposes the sample rate of the given samples using linear interpolation.

-// 'Mono' version of the routine. Returns the number of samples returned in

-// the "dest" buffer

-uint32_t RateTransposerFloat::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint32_t nSamples)

-{

-	if (!nSamples)

-		return 0; // no samples, no work

-

-	unsigned used = 0, i = 0;

-

-	// Process the last sample saved from the sPrevSampleLious call first...

-	while (this->fSlopeCount <= 1.0f)

-	{

-		dest[2 * i] = static_cast<SAMPLETYPE>((1.0f - this->fSlopeCount) * this->sPrevSampleL + this->fSlopeCount * src[0]);

-		dest[2 * i + 1] = static_cast<SAMPLETYPE>((1.0f - this->fSlopeCount) * this->sPrevSampleR + this->fSlopeCount * src[1]);

-		++i;

-		this->fSlopeCount += this->fRate;

-	}

-	// now always (iSlopeCount > 1.0f)

-	this->fSlopeCount -= 1.0f;

-

-	if (nSamples > 1)

-	{

-		while (1)

-		{

-			while (this->fSlopeCount > 1.0f)

-			{

-				this->fSlopeCount -= 1.0f;

-				++used;

-				if (used >= nSamples - 1)

-					goto end;

-			}

-			unsigned srcPos = 2 * used;

-

-			dest[2 * i] = static_cast<SAMPLETYPE>((1.0f - this->fSlopeCount) * src[srcPos] + this->fSlopeCount * src[srcPos + 2]);

-			dest[2 * i + 1] = static_cast<SAMPLETYPE>((1.0f - this->fSlopeCount) * src[srcPos + 1] + this->fSlopeCount * src[srcPos + 3]);

-			++i;

-			this->fSlopeCount += this->fRate;

-		}

-	}

-end:

-	// Store the last sample for the next round

-	this->sPrevSampleL = src[2 * nSamples - 2];

-	this->sPrevSampleR = src[2 * nSamples - 1];

-

-	return i;

-}

@@ -40,7 +40,6 @@

 #include <stdexcept>

 #include "RateTransposer.h"

-#include "AAFilter.h"

 using namespace soundtouch;

@@ -137,7 +136,7 @@ AAFilter *RateTransposer::getAAFilter()

 	return this->pAAFilter.get();

 }

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

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

 // iRate, larger faster iRates.

 void RateTransposer::setRate(float newRate)

 {

@@ -192,7 +191,7 @@ void RateTransposer::downsample(const SAMPLETYPE *src, uint32_t nSamples)

 	// If the parameter 'uRate' value is larger than 'SCALE', first apply the

 	// anti-alias filter to remove high frequencies (prevent them from folding

 	// over the lover frequencies), then transpose. */

-	

+

 	// Add the new samples to the end of the storeBuffer */

 	this->storeBuffer.putSamples(src, nSamples);

@@ -407,7 +406,7 @@ end:

 	return i;

 }

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

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

 // iRate, larger faster iRates.

 void RateTransposerInteger::setRate(float newRate)

 {

@@ -221,7 +221,7 @@ void RateTransposer::processSamples(const SAMPLETYPE *src, uint32_t nSamples)

 {

 	if (!nSamples)

 		return;

-	assert(this->pAAFilter.get());

+	assert(this->pAAFilter);

 	// If anti-alias filter is turned off, simply transpose without applying

 	// the filter

@@ -325,7 +325,7 @@ uint32_t RateTransposerInteger::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE

 	LONG_SAMPLETYPE temp, vol1;

 	while (this->iSlopeCount <= SCALE)

 	{

-		vol1 = static_cast<LONG_SAMPLETYPE>(SCALE - this->iSlopeCount);

+		vol1 = SCALE - this->iSlopeCount;

 		temp = vol1 * this->sPrevSampleL + this->iSlopeCount * src[0];

 		dest[i] = static_cast<SAMPLETYPE>(temp / SCALE);

 		++i;

@@ -343,7 +343,7 @@ uint32_t RateTransposerInteger::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE

 			if (used >= nSamples - 1)

 				goto end;

 		}

-		vol1 = static_cast<LONG_SAMPLETYPE>(SCALE - this->iSlopeCount);

+		vol1 = SCALE - this->iSlopeCount;

 		temp = src[used] * vol1 + this->iSlopeCount * src[used + 1];

 		dest[i] = static_cast<SAMPLETYPE>(temp / SCALE);

 		++i;

@@ -370,7 +370,7 @@ uint32_t RateTransposerInteger::transposeStereo(SAMPLETYPE *dest, const SAMPLETY

 	LONG_SAMPLETYPE temp, vol1;

 	while (this->iSlopeCount <= SCALE)

 	{

-		vol1 = static_cast<LONG_SAMPLETYPE>(SCALE - this->iSlopeCount);

+		vol1 = SCALE - this->iSlopeCount;

 		temp = vol1 * this->sPrevSampleL + this->iSlopeCount * src[0];

 		dest[2 * i] = static_cast<SAMPLETYPE>(temp / SCALE);

 		temp = vol1 * this->sPrevSampleR + this->iSlopeCount * src[1];

@@ -391,7 +391,7 @@ uint32_t RateTransposerInteger::transposeStereo(SAMPLETYPE *dest, const SAMPLETY

 				goto end;

 		}

 		unsigned srcPos = 2 * used;

-		vol1 = static_cast<LONG_SAMPLETYPE>(SCALE - this->iSlopeCount);

+		vol1 = SCALE - this->iSlopeCount;

 		temp = src[srcPos] * vol1 + this->iSlopeCount * src[srcPos + 2];

 		dest[2 * i] = static_cast<SAMPLETYPE>(temp / SCALE);

 		temp = src[srcPos + 1] * vol1 + this->iSlopeCount * src[srcPos + 3];

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

 ///

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

 //

-// Last changed  : $Date: 2006/03/19 10:05:49 $

-// File revision : $Revision: 1.13 $

+// Last changed  : $Date: 2011-09-02 15:56:11 -0300 (sex, 02 set 2011) $

+// File revision : $Revision: 4 $

 //

-// $Id: RateTransposer.cpp,v 1.13 2006/03/19 10:05:49 Olli Exp $

+// $Id: RateTransposer.cpp 131 2011-09-02 18:56:11Z oparviai $

 //

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

 //

@@ -50,7 +50,7 @@ class RateTransposerInteger : public RateTransposer

 {

 protected:

 	int iSlopeCount;

-	uint32_t uRate;

+	int iRate;

 	SAMPLETYPE sPrevSampleL, sPrevSampleR;

 	virtual void resetRegisters();

@@ -73,7 +73,6 @@ class RateTransposerFloat : public RateTransposer

 {

 protected:

 	float fSlopeCount;

-	float fRateStep;

 	SAMPLETYPE sPrevSampleL, sPrevSampleR;

 	virtual void resetRegisters();

@@ -86,406 +85,336 @@ public:

 	virtual ~RateTransposerFloat();

 };

-/*#ifndef min

-#define min(a,b) ((a > b) ? b : a)

-#define max(a,b) ((a < b) ? b : a)

-#endif*/

-

-

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

 // depending on if we've a MMX/SSE/etc-capable CPU available or not.

-void * RateTransposer::operator new(size_t)

+void *RateTransposer::operator new(size_t)

 {

-    // Notice! don't use "new TDStretch" directly, use "newInstance" to create a new instance instead!

-    //assert(false);

-    //return NULL;

+	// Notice! don't use "new TDStretch" directly, use "newInstance" to create a new instance instead!

+	//assert(false);

+	//return NULL;

 	throw std::runtime_error("Don't use 'new RateTransposer', use 'newInstance' member instead!");

 }

-

 RateTransposer *RateTransposer::newInstance()

 {

-#ifdef INTEGER_SAMPLES

-    return ::new RateTransposerInteger;

+#ifdef SOUNDTOUCH_INTEGER_SAMPLES

+	return ::new RateTransposerInteger;

 #else

-    return ::new RateTransposerFloat;

+	return ::new RateTransposerFloat;

 #endif

 }

-

 // Constructor

 RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer)

 {

-    uChannels = 2;

-    bUseAAFilter = true;

+	this->numChannels = 2;

+	this->bUseAAFilter = true;

+	this->fRate = 0;

-    // Instantiates the anti-alias filter with default tap length

-    // of 32

-    pAAFilter = new AAFilter(32);

+	// Instantiates the anti-alias filter with default tap length

+	// of 32

+	this->pAAFilter.reset(new AAFilter(32));

 }

-

-

 RateTransposer::~RateTransposer()

 {

-    delete pAAFilter;

 }

-

-

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

-void RateTransposer::enableAAFilter(const bool newMode)

+void RateTransposer::enableAAFilter(bool newMode)

 {

-    bUseAAFilter = newMode;

+	this->bUseAAFilter = newMode;

 }

-

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

 bool RateTransposer::isAAFilterEnabled() const

 {

-    return bUseAAFilter;

+	return this->bUseAAFilter;

 }

-

-AAFilter *RateTransposer::getAAFilter() const

+AAFilter *RateTransposer::getAAFilter()

 {

-    return pAAFilter;

+	return this->pAAFilter.get();

 }

-

-

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

-// uRate, larger faster uRates.

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

+// iRate, larger faster iRates.

 void RateTransposer::setRate(float newRate)

 {

-    float fCutoff;

-

-    fRate = newRate;

-

-    // design a new anti-alias filter

-    if (newRate > 1.0f)

-    {

-        fCutoff = 0.5f / newRate;

-    }

-    else

-    {

-        fCutoff = 0.5f * newRate;

-    }

-    pAAFilter->setCutoffFreq(fCutoff);

-}

+	double fCutoff;

+	this->fRate = newRate;

-// Outputs as many samples of the 'outputBuffer' as possible, and if there's

-// any room left, outputs also as many of the incoming samples as possible.

-// The goal is to drive the outputBuffer empty.

-//

-// It's allowed for 'output' and 'input' parameters to point to the same

-// memory position.

-void RateTransposer::flushStoreBuffer()

-{

-	if (storeBuffer.isEmpty())

-		return;

-

-	outputBuffer.moveSamples(storeBuffer);

+	// design a new anti-alias filter

+	if (newRate > 1.0f)

+		fCutoff = 0.5f / newRate;

+	else

+		fCutoff = 0.5f * newRate;

+    this->pAAFilter->setCutoffFreq(fCutoff);

 }

-

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

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

 // the input of the object.

-void RateTransposer::putSamples(const SAMPLETYPE *samples, uint32_t numsamples)

+void RateTransposer::putSamples(const SAMPLETYPE *samples, uint32_t nSamples)

 {

-    processSamples(samples, numsamples);

+	this->processSamples(samples, nSamples);

 }

-

-

 // Transposes up the sample rate, causing the observed playback 'rate' of the

 // sound to decrease

-void RateTransposer::upsample(const SAMPLETYPE *src, uint32_t numsamples)

+void RateTransposer::upsample(const SAMPLETYPE *src, uint32_t nSamples)

 {

-    int count, sizeTemp, num;

-

-    // If the parameter 'uRate' value is smaller than 'SCALE', first transpose

-    // the samples and then apply the anti-alias filter to remove aliasing.

+	// If the parameter 'uRate' value is smaller than 'SCALE', first transpose

+	// the samples and then apply the anti-alias filter to remove aliasing.

-    // First check that there's enough room in 'storeBuffer'

-    // (+16 is to reserve some slack in the destination buffer)

-    sizeTemp = (int)((float)numsamples / fRate + 16.0f);

+	// First check that there's enough room in 'storeBuffer'

+	// (+16 is to reserve some slack in the destination buffer)

+	uint32_t sizeTemp = static_cast<uint32_t>(nSamples / this->fRate + 16.0f);

-    // Transpose the samples, store the result into the end of "storeBuffer"

-    count = transpose(storeBuffer.ptrEnd(sizeTemp), src, numsamples);

-    storeBuffer.putSamples(count);

+	// Transpose the samples, store the result into the end of "storeBuffer"

+	uint32_t count = this->transpose(this->storeBuffer.ptrEnd(sizeTemp), src, nSamples);

+	this->storeBuffer.putSamples(count);

-    // Apply the anti-alias filter to samples in "store output", output the

-    // result to "dest"

-    num = storeBuffer.numSamples();

-    count = pAAFilter->evaluate(outputBuffer.ptrEnd(num),

-        storeBuffer.ptrBegin(), num, uChannels);

-    outputBuffer.putSamples(count);

+	// Apply the anti-alias filter to samples in "store output", output the

+	// result to "dest"

+	uint32_t num = this->storeBuffer.numSamples();

+	count = this->pAAFilter->evaluate(this->outputBuffer.ptrEnd(num), this->storeBuffer.ptrBegin(), num, this->numChannels);

+	this->outputBuffer.putSamples(count);

-    // Remove the processed samples from "storeBuffer"

-    storeBuffer.receiveSamples(count);

+	// Remove the processed samples from "storeBuffer"

+	this->storeBuffer.receiveSamples(count);

 }

-

 // Transposes down the sample rate, causing the observed playback 'rate' of the

 // sound to increase

-void RateTransposer::downsample(const SAMPLETYPE *src, uint32_t numsamples)

+void RateTransposer::downsample(const SAMPLETYPE *src, uint32_t nSamples)

 {

-    int count, sizeTemp;

-

-    // If the parameter 'uRate' value is larger than 'SCALE', first apply the

-    // anti-alias filter to remove high frequencies (prevent them from folding

-    // over the lover frequencies), then transpose. */

-

-    // Add the new samples to the end of the storeBuffer */

-    storeBuffer.putSamples(src, numsamples);

-

-    // Anti-alias filter the samples to prevent folding and output the filtered

-    // data to tempBuffer. Note : because of the FIR filter length, the

-    // filtering routine takes in 'filter_length' more samples than it outputs.

-    assert(tempBuffer.isEmpty());

-    sizeTemp = storeBuffer.numSamples();

-

-    count = pAAFilter->evaluate(tempBuffer.ptrEnd(sizeTemp),

-        storeBuffer.ptrBegin(), sizeTemp, uChannels);

-

-    // Remove the filtered samples from 'storeBuffer'

-    storeBuffer.receiveSamples(count);

-

-    // Transpose the samples (+16 is to reserve some slack in the destination buffer)

-    sizeTemp = (int)((float)numsamples / fRate + 16.0f);

-    count = transpose(outputBuffer.ptrEnd(sizeTemp), tempBuffer.ptrBegin(), count);

-    outputBuffer.putSamples(count);

+	// If the parameter 'uRate' value is larger than 'SCALE', first apply the

+	// anti-alias filter to remove high frequencies (prevent them from folding

+	// over the lover frequencies), then transpose. */

+	

+	// Add the new samples to the end of the storeBuffer */

+	this->storeBuffer.putSamples(src, nSamples);

+

+	// Anti-alias filter the samples to prevent folding and output the filtered

+	// data to tempBuffer. Note : because of the FIR filter length, the

+	// filtering routine takes in 'filter_length' more samples than it outputs.

+	assert(this->tempBuffer.isEmpty());

+	uint32_t sizeTemp = this->storeBuffer.numSamples();