ratetransposer.cpp

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/*****************************************************************************
 * 
 * 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 @ iki.fi
 * File created  : 13-Jan-2002
 * 
 * Last changed  : $Date: 2004/10/26 19:09:37 $
 * File revision : $Revision: 1.3 $
 *
 * $Id: RateTransposer.cpp,v 1.3 2004/10/26 19:09:37 vjohnson Exp $
 *
 * License :
 * 
 *  SoundTouch sound 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 <memory.h>
#include <assert.h>
#include <stdlib.h>
#include <stdio.h>
#include <limits.h>
#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;
    uint uRate;
    soundtouch::SAMPLETYPE sPrevSampleL, sPrevSampleR;

    virtual void resetRegisters();

    virtual uint transposeStereo(soundtouch::SAMPLETYPE *dest, 
                         const soundtouch::SAMPLETYPE *src, 
                         uint numSamples);
    virtual uint transposeMono(soundtouch::SAMPLETYPE *dest, 
                       const soundtouch::SAMPLETYPE *src, 
                       uint 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;
    float fRateStep;
    soundtouch::SAMPLETYPE sPrevSampleL, sPrevSampleR;

    virtual void resetRegisters();

    virtual uint transposeStereo(soundtouch::SAMPLETYPE *dest, 
                         const soundtouch::SAMPLETYPE *src, 
                         uint numSamples);
    virtual uint transposeMono(soundtouch::SAMPLETYPE *dest, 
                       const soundtouch::SAMPLETYPE *src, 
                       uint numSamples);

public:
    RateTransposerFloat();
    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 s)
{
    // Notice! don't use "new TDStretch" directly, use "newInstance" to create a new instance instead!
    assert(FALSE);  
    return NULL;
}


RateTransposer *RateTransposer::newInstance()
{
#ifdef INTEGER_SAMPLES
    return ::new RateTransposerInteger;
#else
    return ::new RateTransposerFloat;
#endif
}


// Constructor
RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer)
{
    uChannels = 2;
    bUseAAFilter = TRUE;

    // Instantiates the anti-alias filter with default tap length
    // of 32
    pAAFilter = 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)
{
    bUseAAFilter = newMode;
}


/// Returns nonzero if anti-alias filter is enabled.
BOOL RateTransposer::isAAFilterEnabled() const
{
    return bUseAAFilter;
}


AAFilter *RateTransposer::getAAFilter() const
{
    return pAAFilter;
}



// Sets new target uRate. Normal uRate = 1.0, smaller values represent slower 
// uRate, larger faster uRates.
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);
}


// 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);
}


// Adds 'numSamples' pcs of samples from the 'samples' memory position into
// the input of the object.
void RateTransposer::putSamples(const SAMPLETYPE *samples, uint numSamples)
{
    processSamples(samples, numSamples);
}



// Transposes up the sample rate, causing the observed playback 'rate' of the
// sound to decrease
void RateTransposer::upsample(const SAMPLETYPE *src, uint numSamples)
{
    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.

    // 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);

    // Transpose the samples, store the result into the end of "storeBuffer"
    count = transpose(storeBuffer.ptrEnd(sizeTemp), src, numSamples);
    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);

    // 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 numSamples)
{
    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);
}


// 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, uint numSamples)
{
    uint count;
    uint sizeReq;

    if (numSamples == 0) return;
    assert(pAAFilter);

    // If anti-alias filter is turned off, simply transpose without applying
    // the filter
    if (bUseAAFilter == FALSE) 
    {
        sizeReq = (int)((float)numSamples / fRate + 1.0f);
        count = transpose(outputBuffer.ptrEnd(sizeReq), src, numSamples);
        outputBuffer.putSamples(count);
        return;
    }

    // Transpose with anti-alias filter
    if (fRate < 1.0f) 
    {
        upsample(src, numSamples);
    } 
    else  
    {
        downsample(src, numSamples);
    }
}


// Transposes the sample rate of the given samples using linear interpolation. 
// Returns the number of samples returned in the "dest" buffer