////////////////////////////////////////////////////////////////////////////////
///
/// 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"
#include "AAFilter.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.get());
// 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;
}