mirror of
https://github.com/RPCS3/soundtouch.git
synced 2024-11-09 12:22:51 +01:00
Restructured RateTransposer to allow hosting alternative resampling algorithms.
This commit is contained in:
oparviai
parent
8174f6bc10
commit
abfeb3fcc9
@ -162,6 +162,12 @@ public:
|
||||
/// Sets number of channels, 1 = mono, 2 = stereo.
|
||||
void setChannels(int numChannels);
|
||||
|
||||
/// Get number of channels
|
||||
int getChannels()
|
||||
{
|
||||
return channels;
|
||||
}
|
||||
|
||||
/// Returns nonzero if there aren't any samples available for outputting.
|
||||
virtual int isEmpty() const;
|
||||
|
||||
|
||||
@ -205,6 +205,31 @@ uint AAFilter::evaluate(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples
|
||||
}
|
||||
|
||||
|
||||
/// Applies the filter to the given src & dest pipes, so that processed amount of
|
||||
/// samples get removed from src, and produced amount added to dest
|
||||
/// Note : The amount of outputted samples is by value of 'filter length'
|
||||
/// smaller than the amount of input samples.
|
||||
uint AAFilter::evaluate(FIFOSampleBuffer &dest, FIFOSampleBuffer &src) const
|
||||
{
|
||||
SAMPLETYPE *pdest;
|
||||
const SAMPLETYPE *psrc;
|
||||
uint numSrcSamples;
|
||||
uint result;
|
||||
int numChannels = src.getChannels();
|
||||
|
||||
assert(numChannels == dest.getChannels());
|
||||
|
||||
numSrcSamples = src.numSamples();
|
||||
psrc = src.ptrBegin();
|
||||
pdest = dest.ptrEnd(numSrcSamples);
|
||||
result = pFIR->evaluate(pdest, psrc, numSrcSamples, numChannels);
|
||||
src.receiveSamples(result);
|
||||
dest.putSamples(result);
|
||||
|
||||
return result;
|
||||
}
|
||||
|
||||
|
||||
uint AAFilter::getLength() const
|
||||
{
|
||||
return pFIR->getLength();
|
||||
|
||||
@ -45,6 +45,7 @@
|
||||
#define AAFilter_H
|
||||
|
||||
#include "STTypes.h"
|
||||
#include "FIFOSampleBuffer.h"
|
||||
|
||||
namespace soundtouch
|
||||
{
|
||||
@ -84,6 +85,14 @@ public:
|
||||
const SAMPLETYPE *src,
|
||||
uint numSamples,
|
||||
uint numChannels) const;
|
||||
|
||||
/// Applies the filter to the given src & dest pipes, so that processed amount of
|
||||
/// samples get removed from src, and produced amount added to dest
|
||||
/// Note : The amount of outputted samples is by value of 'filter length'
|
||||
/// smaller than the amount of input samples.
|
||||
uint AAFilter::evaluate(FIFOSampleBuffer &dest,
|
||||
FIFOSampleBuffer &src) const;
|
||||
|
||||
};
|
||||
|
||||
}
|
||||
|
||||
@ -48,9 +48,31 @@
|
||||
using namespace soundtouch;
|
||||
|
||||
|
||||
|
||||
/// A linear samplerate transposer class that uses integer arithmetics.
|
||||
/// for the transposing.
|
||||
class RateTransposerInteger : public RateTransposer
|
||||
class LinearTransposerBase: public TransposerBase
|
||||
{
|
||||
protected:
|
||||
virtual int transposeStereo(SAMPLETYPE *dest,
|
||||
const SAMPLETYPE *src,
|
||||
uint numSamples) = 0;
|
||||
virtual int transposeMono(SAMPLETYPE *dest,
|
||||
const SAMPLETYPE *src,
|
||||
uint numSamples) = 0;
|
||||
virtual int transposeMulti(SAMPLETYPE *dest,
|
||||
const SAMPLETYPE *src,
|
||||
uint numSamples) = 0;
|
||||
public:
|
||||
virtual int transpose(FIFOSampleBuffer &dest, FIFOSampleBuffer &src);
|
||||
|
||||
static LinearTransposerBase *newInstance();
|
||||
};
|
||||
|
||||
|
||||
/// A linear samplerate transposer class that uses integer arithmetics.
|
||||
/// for the transposing.
|
||||
class LinearTransposerInteger : public LinearTransposerBase
|
||||
{
|
||||
protected:
|
||||
int iSlopeCount;
|
||||
@ -67,19 +89,18 @@ protected:
|
||||
uint numSamples);
|
||||
virtual int transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uint numSamples);
|
||||
public:
|
||||
RateTransposerInteger();
|
||||
virtual ~RateTransposerInteger();
|
||||
LinearTransposerInteger();
|
||||
virtual ~LinearTransposerInteger();
|
||||
|
||||
/// 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
|
||||
class LinearTransposerFloat : public LinearTransposerBase
|
||||
{
|
||||
protected:
|
||||
float fSlopeCount;
|
||||
@ -96,28 +117,53 @@ protected:
|
||||
virtual int transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples);
|
||||
|
||||
public:
|
||||
RateTransposerFloat();
|
||||
virtual ~RateTransposerFloat();
|
||||
LinearTransposerFloat();
|
||||
virtual ~LinearTransposerFloat();
|
||||
};
|
||||
|
||||
|
||||
TransposerBase::TransposerBase()
|
||||
{
|
||||
numChannels = 0;
|
||||
rate = 1.0f;
|
||||
}
|
||||
|
||||
|
||||
TransposerBase::~TransposerBase()
|
||||
{
|
||||
}
|
||||
|
||||
|
||||
void TransposerBase::setChannels(int channels)
|
||||
{
|
||||
numChannels = channels;
|
||||
resetRegisters();
|
||||
}
|
||||
|
||||
|
||||
void TransposerBase::setRate(float newRate)
|
||||
{
|
||||
rate = newRate;
|
||||
}
|
||||
|
||||
|
||||
// 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 s)
|
||||
{
|
||||
ST_THROW_RT_ERROR("Error in RateTransoser::new: don't use \"new TDStretch\" directly, use \"newInstance\" to create a new instance instead!");
|
||||
return newInstance();
|
||||
}
|
||||
*/
|
||||
|
||||
|
||||
RateTransposer *RateTransposer::newInstance()
|
||||
TransposerBase *TransposerBase::newInstance()
|
||||
{
|
||||
#ifdef SOUNDTOUCH_INTEGER_SAMPLES
|
||||
return ::new RateTransposerInteger;
|
||||
return ::new LinearTransposerInteger;
|
||||
#else
|
||||
return ::new RateTransposerFloat;
|
||||
return ::new LinearTransposerFloat;
|
||||
#endif
|
||||
}
|
||||
|
||||
@ -125,13 +171,12 @@ RateTransposer *RateTransposer::newInstance()
|
||||
// Constructor
|
||||
RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer)
|
||||
{
|
||||
numChannels = 2;
|
||||
bUseAAFilter = TRUE;
|
||||
fRate = 0;
|
||||
|
||||
// Instantiates the anti-alias filter with default tap length
|
||||
// of 32
|
||||
pAAFilter = new AAFilter(32);
|
||||
pTransposer = TransposerBase::newInstance();
|
||||
}
|
||||
|
||||
|
||||
@ -139,6 +184,7 @@ RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer)
|
||||
RateTransposer::~RateTransposer()
|
||||
{
|
||||
delete pAAFilter;
|
||||
delete pTransposer;
|
||||
}
|
||||
|
||||
|
||||
@ -170,7 +216,7 @@ void RateTransposer::setRate(float newRate)
|
||||
{
|
||||
double fCutoff;
|
||||
|
||||
fRate = newRate;
|
||||
pTransposer->setRate(newRate);
|
||||
|
||||
// design a new anti-alias filter
|
||||
if (newRate > 1.0f)
|
||||
@ -192,11 +238,11 @@ void RateTransposer::setRate(float newRate)
|
||||
// It's allowed for 'output' and 'input' parameters to point to the same
|
||||
// memory position.
|
||||
/*
|
||||
void RateTransposer::flushStoreBuffer()
|
||||
void RateTransposer::flushinputBuffer()
|
||||
{
|
||||
if (storeBuffer.isEmpty()) return;
|
||||
if (inputBuffer.isEmpty()) return;
|
||||
|
||||
outputBuffer.moveSamples(storeBuffer);
|
||||
outputBuffer.moveSamples(inputBuffer);
|
||||
}
|
||||
*/
|
||||
|
||||
@ -209,70 +255,6 @@ void RateTransposer::putSamples(const SAMPLETYPE *samples, uint nSamples)
|
||||
}
|
||||
|
||||
|
||||
|
||||
// Transposes up the sample rate, causing the observed playback 'rate' of the
|
||||
// sound to decrease
|
||||
void RateTransposer::upsample(const SAMPLETYPE *src, uint nSamples)
|
||||
{
|
||||
uint 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.
|
||||
|
||||
// First check that there's enough room in 'storeBuffer'
|
||||
// (+16 is to reserve some slack in the destination buffer)
|
||||
sizeTemp = (uint)((float)nSamples / fRate + 16.0f);
|
||||
|
||||
// Transpose the samples, store the result into the end of "storeBuffer"
|
||||
count = transpose(storeBuffer.ptrEnd(sizeTemp), src, nSamples);
|
||||
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, (uint)numChannels);
|
||||
outputBuffer.putSamples(count);
|
||||
|
||||
// Remove the processed samples from "storeBuffer"
|
||||
storeBuffer.receiveSamples(count);
|
||||
}
|
||||
|
||||
|
||||
// Transposes down the sample rate, causing the observed playback 'rate' of the
|
||||
// sound to increase
|
||||
void RateTransposer::downsample(const SAMPLETYPE *src, uint nSamples)
|
||||
{
|
||||
uint 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, 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(tempBuffer.isEmpty());
|
||||
sizeTemp = storeBuffer.numSamples();
|
||||
|
||||
count = pAAFilter->evaluate(tempBuffer.ptrEnd(sizeTemp),
|
||||
storeBuffer.ptrBegin(), sizeTemp, (uint)numChannels);
|
||||
|
||||
if (count == 0) return;
|
||||
|
||||
// Remove the filtered samples from 'storeBuffer'
|
||||
storeBuffer.receiveSamples(count);
|
||||
|
||||
// Transpose the samples (+16 is to reserve some slack in the destination buffer)
|
||||
sizeTemp = (uint)((float)nSamples / fRate + 16.0f);
|
||||
count = transpose(outputBuffer.ptrEnd(sizeTemp), tempBuffer.ptrBegin(), count);
|
||||
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
|
||||
@ -280,52 +262,77 @@ void RateTransposer::downsample(const SAMPLETYPE *src, uint nSamples)
|
||||
void RateTransposer::processSamples(const SAMPLETYPE *src, uint nSamples)
|
||||
{
|
||||
uint count;
|
||||
uint sizeReq;
|
||||
|
||||
if (nSamples == 0) return;
|
||||
assert(pAAFilter);
|
||||
|
||||
// Store samples to input buffer
|
||||
inputBuffer.putSamples(src, nSamples);
|
||||
|
||||
// If anti-alias filter is turned off, simply transpose without applying
|
||||
// the filter
|
||||
if (bUseAAFilter == FALSE)
|
||||
{
|
||||
sizeReq = (uint)((float)nSamples / fRate + 1.0f);
|
||||
count = transpose(outputBuffer.ptrEnd(sizeReq), src, nSamples);
|
||||
outputBuffer.putSamples(count);
|
||||
count = pTransposer->transpose(outputBuffer, inputBuffer);
|
||||
return;
|
||||
}
|
||||
|
||||
assert(pAAFilter);
|
||||
|
||||
// Transpose with anti-alias filter
|
||||
if (fRate < 1.0f)
|
||||
if (pTransposer->rate < 1.0f)
|
||||
{
|
||||
upsample(src, nSamples);
|
||||
// If the parameter 'Rate' value is smaller than 1, first transpose
|
||||
// the samples and then apply the anti-alias filter to remove aliasing.
|
||||
|
||||
// Transpose the samples, store the result to end of "midBuffer"
|
||||
pTransposer->transpose(midBuffer, inputBuffer);
|
||||
|
||||
// Apply the anti-alias filter for transposed samples in midBuffer
|
||||
pAAFilter->evaluate(outputBuffer, midBuffer);
|
||||
}
|
||||
else
|
||||
{
|
||||
downsample(src, nSamples);
|
||||
// If the parameter 'Rate' value is larger than 1, first apply the
|
||||
// anti-alias filter to remove high frequencies (prevent them from folding
|
||||
// over the lover frequencies), then transpose.
|
||||
|
||||
// Apply the anti-alias filter for samples in inputBuffer
|
||||
pAAFilter->evaluate(midBuffer, inputBuffer);
|
||||
|
||||
// Transpose the AA-filtered samples in "midBuffer"
|
||||
pTransposer->transpose(outputBuffer, midBuffer);
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
// Transposes the sample rate of the given samples using linear interpolation.
|
||||
// Returns the number of samples returned in the "dest" buffer
|
||||
inline int RateTransposer::transpose(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
|
||||
int LinearTransposerBase::transpose(FIFOSampleBuffer &dest, FIFOSampleBuffer &src)
|
||||
{
|
||||
int numSrcSamples = src.numSamples();
|
||||
int sizeDemand = (int)((float)numSrcSamples / rate) + 8;
|
||||
int numOutput;
|
||||
SAMPLETYPE *psrc = src.ptrBegin();
|
||||
SAMPLETYPE *pdest = dest.ptrEnd(sizeDemand);
|
||||
|
||||
#ifndef USE_MULTICH_ALWAYS
|
||||
if (numChannels == 1)
|
||||
{
|
||||
return transposeMono(dest, src, nSamples);
|
||||
numOutput = transposeMono(pdest, psrc, numSrcSamples);
|
||||
}
|
||||
else if (numChannels == 2)
|
||||
{
|
||||
return transposeStereo(dest, src, nSamples);
|
||||
numOutput = transposeStereo(pdest, psrc, numSrcSamples);
|
||||
}
|
||||
else
|
||||
#endif // USE_MULTICH_ALWAYS
|
||||
{
|
||||
assert(numChannels > 0);
|
||||
return transposeMulti(dest, src, nSamples);
|
||||
numOutput = transposeMulti(pdest, psrc, numSrcSamples);
|
||||
}
|
||||
dest.putSamples(numOutput);
|
||||
src.receiveSamples(numOutput);
|
||||
return numOutput;
|
||||
}
|
||||
|
||||
|
||||
@ -333,17 +340,13 @@ inline int RateTransposer::transpose(SAMPLETYPE *dest, const SAMPLETYPE *src, ui
|
||||
void RateTransposer::setChannels(int nChannels)
|
||||
{
|
||||
assert(nChannels > 0);
|
||||
if (numChannels == nChannels) return;
|
||||
|
||||
// assert(nChannels == 1 || nChannels == 2);
|
||||
numChannels = nChannels;
|
||||
if (pTransposer->numChannels == nChannels) return;
|
||||
pTransposer->setChannels(nChannels);
|
||||
|
||||
storeBuffer.setChannels(numChannels);
|
||||
tempBuffer.setChannels(numChannels);
|
||||
outputBuffer.setChannels(numChannels);
|
||||
|
||||
// Inits the linear interpolation registers
|
||||
resetRegisters();
|
||||
inputBuffer.setChannels(nChannels);
|
||||
midBuffer.setChannels(nChannels);
|
||||
outputBuffer.setChannels(nChannels);
|
||||
}
|
||||
|
||||
|
||||
@ -351,7 +354,8 @@ void RateTransposer::setChannels(int nChannels)
|
||||
void RateTransposer::clear()
|
||||
{
|
||||
outputBuffer.clear();
|
||||
storeBuffer.clear();
|
||||
midBuffer.clear();
|
||||
inputBuffer.clear();
|
||||
}
|
||||
|
||||
|
||||
@ -362,36 +366,36 @@ int RateTransposer::isEmpty() const
|
||||
|
||||
res = FIFOProcessor::isEmpty();
|
||||
if (res == 0) return 0;
|
||||
return storeBuffer.isEmpty();
|
||||
return inputBuffer.isEmpty();
|
||||
}
|
||||
|
||||
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// RateTransposerInteger - integer arithmetic implementation
|
||||
// LinearTransposerInteger - integer arithmetic implementation
|
||||
//
|
||||
|
||||
/// fixed-point interpolation routine precision
|
||||
#define SCALE 65536
|
||||
|
||||
// Constructor
|
||||
RateTransposerInteger::RateTransposerInteger() : RateTransposer()
|
||||
LinearTransposerInteger::LinearTransposerInteger() : LinearTransposerBase()
|
||||
{
|
||||
// Notice: use local function calling syntax for sake of clarity,
|
||||
// to indicate the fact that C++ constructor can't call virtual functions.
|
||||
sPrevSample=0;
|
||||
RateTransposerInteger::resetRegisters();
|
||||
RateTransposerInteger::setRate(1.0f);
|
||||
resetRegisters();
|
||||
setRate(1.0f);
|
||||
}
|
||||
|
||||
|
||||
RateTransposerInteger::~RateTransposerInteger()
|
||||
LinearTransposerInteger::~LinearTransposerInteger()
|
||||
{
|
||||
if (sPrevSample) delete[] sPrevSample;
|
||||
}
|
||||
|
||||
|
||||
void RateTransposerInteger::resetRegisters()
|
||||
void LinearTransposerInteger::resetRegisters()
|
||||
{
|
||||
iSlopeCount = 0;
|
||||
delete[] sPrevSample;
|
||||
@ -404,7 +408,7 @@ void RateTransposerInteger::resetRegisters()
|
||||
// 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
|
||||
int RateTransposerInteger::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
|
||||
int LinearTransposerInteger::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
|
||||
{
|
||||
int i, remain;
|
||||
LONG_SAMPLETYPE temp, vol1;
|
||||
@ -453,7 +457,7 @@ end:
|
||||
// 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
|
||||
int RateTransposerInteger::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
|
||||
int LinearTransposerInteger::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
|
||||
{
|
||||
int i, remain;
|
||||
LONG_SAMPLETYPE temp, vol1;
|
||||
@ -504,7 +508,7 @@ end:
|
||||
}
|
||||
|
||||
|
||||
int RateTransposerInteger::transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
|
||||
int LinearTransposerInteger::transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
|
||||
{
|
||||
int i, remaining;
|
||||
LONG_SAMPLETYPE temp, vol1;
|
||||
@ -561,37 +565,37 @@ end:
|
||||
|
||||
// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower
|
||||
// iRate, larger faster iRates.
|
||||
void RateTransposerInteger::setRate(float newRate)
|
||||
void LinearTransposerInteger::setRate(float newRate)
|
||||
{
|
||||
iRate = (int)(newRate * SCALE + 0.5f);
|
||||
RateTransposer::setRate(newRate);
|
||||
TransposerBase::setRate(newRate);
|
||||
}
|
||||
|
||||
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
//
|
||||
// RateTransposerFloat - floating point arithmetic implementation
|
||||
// LinearTransposerFloat - floating point arithmetic implementation
|
||||
//
|
||||
//////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
// Constructor
|
||||
RateTransposerFloat::RateTransposerFloat() : RateTransposer()
|
||||
LinearTransposerFloat::LinearTransposerFloat() : LinearTransposerBase()
|
||||
{
|
||||
// Notice: use local function calling syntax for sake of clarity,
|
||||
// to indicate the fact that C++ constructor can't call virtual functions.
|
||||
sPrevSample = NULL;
|
||||
RateTransposerFloat::resetRegisters();
|
||||
RateTransposerFloat::setRate(1.0f);
|
||||
resetRegisters();
|
||||
setRate(1.0f);
|
||||
}
|
||||
|
||||
|
||||
RateTransposerFloat::~RateTransposerFloat()
|
||||
LinearTransposerFloat::~LinearTransposerFloat()
|
||||
{
|
||||
delete[] sPrevSample;
|
||||
}
|
||||
|
||||
|
||||
void RateTransposerFloat::resetRegisters()
|
||||
void LinearTransposerFloat::resetRegisters()
|
||||
{
|
||||
fSlopeCount = 0;
|
||||
delete[] sPrevSample;
|
||||
@ -604,19 +608,19 @@ void RateTransposerFloat::resetRegisters()
|
||||
// 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
|
||||
int RateTransposerFloat::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
|
||||
int LinearTransposerFloat::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
|
||||
{
|
||||
int i, remain;
|
||||
|
||||
remain = nSamples - 1;
|
||||
i = 0;
|
||||
remain = nSamples - 1;
|
||||
i = 0;
|
||||
|
||||
// Process the last sample saved from the previous call first...
|
||||
while (fSlopeCount <= 1.0f)
|
||||
{
|
||||
dest[i] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSample[0] + fSlopeCount * src[0]);
|
||||
i++;
|
||||
fSlopeCount += fRate;
|
||||
fSlopeCount += rate;
|
||||
}
|
||||
fSlopeCount -= 1.0f;
|
||||
|
||||
@ -633,7 +637,7 @@ int RateTransposerFloat::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src,
|
||||
}
|
||||
dest[i] = (SAMPLETYPE)((1.0f - fSlopeCount) * src[0] + fSlopeCount * src[1]);
|
||||
i++;
|
||||
fSlopeCount += fRate;
|
||||
fSlopeCount += rate;
|
||||
}
|
||||
}
|
||||
end:
|
||||
@ -647,7 +651,7 @@ end:
|
||||
// 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
|
||||
int RateTransposerFloat::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
|
||||
int LinearTransposerFloat::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
|
||||
{
|
||||
int i, remain;
|
||||
|
||||
@ -662,7 +666,7 @@ int RateTransposerFloat::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src
|
||||
dest[2 * i] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSample[0] + fSlopeCount * src[0]);
|
||||
dest[2 * i + 1] = (SAMPLETYPE)((1.0f - fSlopeCount) * sPrevSample[1] + fSlopeCount * src[1]);
|
||||
i++;
|
||||
fSlopeCount += fRate;
|
||||
fSlopeCount += rate;
|
||||
}
|
||||
// now always (iSlopeCount > 1.0f)
|
||||
fSlopeCount -= 1.0f;
|
||||
@ -685,7 +689,7 @@ int RateTransposerFloat::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src
|
||||
+ fSlopeCount * src[3]);
|
||||
|
||||
i++;
|
||||
fSlopeCount += fRate;
|
||||
fSlopeCount += rate;
|
||||
}
|
||||
}
|
||||
end:
|
||||
@ -696,7 +700,7 @@ end:
|
||||
return i;
|
||||
}
|
||||
|
||||
int RateTransposerFloat::transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
|
||||
int LinearTransposerFloat::transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples)
|
||||
{
|
||||
int i, remaining;
|
||||
|
||||
@ -714,7 +718,7 @@ int RateTransposerFloat::transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src,
|
||||
dest ++;
|
||||
}
|
||||
i++;
|
||||
fSlopeCount += fRate;
|
||||
fSlopeCount += rate;
|
||||
}
|
||||
// now always (iSlopeCount > 1.0f)
|
||||
fSlopeCount -= 1.0f;
|
||||
@ -737,7 +741,7 @@ int RateTransposerFloat::transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src,
|
||||
}
|
||||
|
||||
i++;
|
||||
fSlopeCount += fRate;
|
||||
fSlopeCount += rate;
|
||||
}
|
||||
|
||||
end:
|
||||
|
||||
@ -55,51 +55,47 @@
|
||||
namespace soundtouch
|
||||
{
|
||||
|
||||
/// Abstract base class for transposer implementations (linear, advanced vs integer, float etc)
|
||||
class TransposerBase
|
||||
{
|
||||
protected:
|
||||
virtual void resetRegisters() = 0;
|
||||
|
||||
public:
|
||||
float rate;
|
||||
int numChannels;
|
||||
|
||||
TransposerBase();
|
||||
virtual ~TransposerBase();
|
||||
|
||||
virtual int transpose(FIFOSampleBuffer &dest, FIFOSampleBuffer &src) = 0;
|
||||
virtual void setRate(float newRate);
|
||||
virtual void setChannels(int channels);
|
||||
static TransposerBase *newInstance();
|
||||
};
|
||||
|
||||
|
||||
/// 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;
|
||||
|
||||
int numChannels;
|
||||
TransposerBase *pTransposer;
|
||||
|
||||
/// Buffer for collecting samples to feed the anti-alias filter between
|
||||
/// two batches
|
||||
FIFOSampleBuffer storeBuffer;
|
||||
FIFOSampleBuffer inputBuffer;
|
||||
|
||||
/// Buffer for keeping samples between transposing & anti-alias filter
|
||||
FIFOSampleBuffer tempBuffer;
|
||||
FIFOSampleBuffer midBuffer;
|
||||
|
||||
/// Output sample buffer
|
||||
FIFOSampleBuffer outputBuffer;
|
||||
|
||||
BOOL bUseAAFilter;
|
||||
|
||||
virtual void resetRegisters() = 0;
|
||||
|
||||
virtual int transposeStereo(SAMPLETYPE *dest,
|
||||
const SAMPLETYPE *src,
|
||||
uint numSamples) = 0;
|
||||
virtual int transposeMono(SAMPLETYPE *dest,
|
||||
const SAMPLETYPE *src,
|
||||
uint numSamples) = 0;
|
||||
virtual int transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, uint nSamples) = 0;
|
||||
inline int transpose(SAMPLETYPE *dest,
|
||||
const SAMPLETYPE *src,
|
||||
uint numSamples);
|
||||
|
||||
void downsample(const SAMPLETYPE *src,
|
||||
uint numSamples);
|
||||
void upsample(const SAMPLETYPE *src,
|
||||
uint numSamples);
|
||||
|
||||
/// Transposes sample rate by applying anti-alias filter to prevent folding.
|
||||
/// Returns amount of samples returned in the "dest" buffer.
|
||||
@ -108,25 +104,24 @@ protected:
|
||||
void processSamples(const SAMPLETYPE *src,
|
||||
uint 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);
|
||||
// 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();
|
||||
// static RateTransposer *newInstance();
|
||||
|
||||
/// Returns the output buffer object
|
||||
FIFOSamplePipe *getOutput() { return &outputBuffer; };
|
||||
|
||||
/// Returns the store buffer object
|
||||
FIFOSamplePipe *getStore() { return &storeBuffer; };
|
||||
// FIFOSamplePipe *getStore() { return &storeBuffer; };
|
||||
|
||||
/// Return anti-alias filter object
|
||||
AAFilter *getAAFilter();
|
||||
|
||||
@ -97,7 +97,7 @@ SoundTouch::SoundTouch()
|
||||
{
|
||||
// Initialize rate transposer and tempo changer instances
|
||||
|
||||
pRateTransposer = RateTransposer::newInstance();
|
||||
pRateTransposer = new RateTransposer();
|
||||
pTDStretch = TDStretch::newInstance();
|
||||
|
||||
setOutPipe(pTDStretch);
|
||||
@ -255,7 +255,7 @@ void SoundTouch::calcEffectiveRateAndTempo()
|
||||
tempoOut = pTDStretch->getOutput();
|
||||
tempoOut->moveSamples(*output);
|
||||
// move samples in pitch transposer's store buffer to tempo changer's input
|
||||
pTDStretch->moveSamples(*pRateTransposer->getStore());
|
||||
// deprecated : pTDStretch->moveSamples(*pRateTransposer->getStore());
|
||||
|
||||
output = pTDStretch;
|
||||
}
|
||||
|
||||
Loading…
Reference in New Issue
Block a user