michael@0: ////////////////////////////////////////////////////////////////////////////////
michael@0: /// 
michael@0: /// Sample rate transposer. Changes sample rate by using linear interpolation 
michael@0: /// together with anti-alias filtering (first order interpolation with anti-
michael@0: /// alias filtering should be quite adequate for this application)
michael@0: ///
michael@0: /// Author        : Copyright (c) Olli Parviainen
michael@0: /// Author e-mail : oparviai 'at' iki.fi
michael@0: /// SoundTouch WWW: http://www.surina.net/soundtouch
michael@0: ///
michael@0: ////////////////////////////////////////////////////////////////////////////////
michael@0: //
michael@0: // Last changed  : $Date: 2014-04-06 10:57:21 -0500 (Sun, 06 Apr 2014) $
michael@0: // File revision : $Revision: 4 $
michael@0: //
michael@0: // $Id: RateTransposer.cpp 195 2014-04-06 15:57:21Z oparviai $
michael@0: //
michael@0: ////////////////////////////////////////////////////////////////////////////////
michael@0: //
michael@0: // License :
michael@0: //
michael@0: //  SoundTouch audio processing library
michael@0: //  Copyright (c) Olli Parviainen
michael@0: //
michael@0: //  This library is free software; you can redistribute it and/or
michael@0: //  modify it under the terms of the GNU Lesser General Public
michael@0: //  License as published by the Free Software Foundation; either
michael@0: //  version 2.1 of the License, or (at your option) any later version.
michael@0: //
michael@0: //  This library is distributed in the hope that it will be useful,
michael@0: //  but WITHOUT ANY WARRANTY; without even the implied warranty of
michael@0: //  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
michael@0: //  Lesser General Public License for more details.
michael@0: //
michael@0: //  You should have received a copy of the GNU Lesser General Public
michael@0: //  License along with this library; if not, write to the Free Software
michael@0: //  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
michael@0: //
michael@0: ////////////////////////////////////////////////////////////////////////////////
michael@0: 
michael@0: #include <memory.h>
michael@0: #include <assert.h>
michael@0: #include <stdlib.h>
michael@0: #include <stdio.h>
michael@0: #include "RateTransposer.h"
michael@0: #include "InterpolateLinear.h"
michael@0: #include "InterpolateCubic.h"
michael@0: #include "InterpolateShannon.h"
michael@0: #include "AAFilter.h"
michael@0: 
michael@0: using namespace soundtouch;
michael@0: 
michael@0: // Define default interpolation algorithm here
michael@0: TransposerBase::ALGORITHM TransposerBase::algorithm = TransposerBase::CUBIC;
michael@0: 
michael@0: 
michael@0: // Constructor
michael@0: RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer)
michael@0: {
michael@0:     bUseAAFilter = true;
michael@0: 
michael@0:     // Instantiates the anti-alias filter
michael@0:     pAAFilter = new AAFilter(64);
michael@0:     pTransposer = TransposerBase::newInstance();
michael@0: }
michael@0: 
michael@0: 
michael@0: 
michael@0: RateTransposer::~RateTransposer()
michael@0: {
michael@0:     delete pAAFilter;
michael@0:     delete pTransposer;
michael@0: }
michael@0: 
michael@0: 
michael@0: 
michael@0: /// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
michael@0: void RateTransposer::enableAAFilter(bool newMode)
michael@0: {
michael@0:     bUseAAFilter = newMode;
michael@0: }
michael@0: 
michael@0: 
michael@0: /// Returns nonzero if anti-alias filter is enabled.
michael@0: bool RateTransposer::isAAFilterEnabled() const
michael@0: {
michael@0:     return bUseAAFilter;
michael@0: }
michael@0: 
michael@0: 
michael@0: AAFilter *RateTransposer::getAAFilter()
michael@0: {
michael@0:     return pAAFilter;
michael@0: }
michael@0: 
michael@0: 
michael@0: 
michael@0: // Sets new target iRate. Normal iRate = 1.0, smaller values represent slower 
michael@0: // iRate, larger faster iRates.
michael@0: void RateTransposer::setRate(float newRate)
michael@0: {
michael@0:     double fCutoff;
michael@0: 
michael@0:     pTransposer->setRate(newRate);
michael@0: 
michael@0:     // design a new anti-alias filter
michael@0:     if (newRate > 1.0f) 
michael@0:     {
michael@0:         fCutoff = 0.5f / newRate;
michael@0:     } 
michael@0:     else 
michael@0:     {
michael@0:         fCutoff = 0.5f * newRate;
michael@0:     }
michael@0:     pAAFilter->setCutoffFreq(fCutoff);
michael@0: }
michael@0: 
michael@0: 
michael@0: // Adds 'nSamples' pcs of samples from the 'samples' memory position into
michael@0: // the input of the object.
michael@0: void RateTransposer::putSamples(const SAMPLETYPE *samples, uint nSamples)
michael@0: {
michael@0:     processSamples(samples, nSamples);
michael@0: }
michael@0: 
michael@0: 
michael@0: // Transposes sample rate by applying anti-alias filter to prevent folding. 
michael@0: // Returns amount of samples returned in the "dest" buffer.
michael@0: // The maximum amount of samples that can be returned at a time is set by
michael@0: // the 'set_returnBuffer_size' function.
michael@0: void RateTransposer::processSamples(const SAMPLETYPE *src, uint nSamples)
michael@0: {
michael@0:     uint count;
michael@0: 
michael@0:     if (nSamples == 0) return;
michael@0: 
michael@0:     // Store samples to input buffer
michael@0:     inputBuffer.putSamples(src, nSamples);
michael@0: 
michael@0:     // If anti-alias filter is turned off, simply transpose without applying
michael@0:     // the filter
michael@0:     if (bUseAAFilter == false) 
michael@0:     {
michael@0:         count = pTransposer->transpose(outputBuffer, inputBuffer);
michael@0:         return;
michael@0:     }
michael@0: 
michael@0:     assert(pAAFilter);
michael@0: 
michael@0:     // Transpose with anti-alias filter
michael@0:     if (pTransposer->rate < 1.0f) 
michael@0:     {
michael@0:         // If the parameter 'Rate' value is smaller than 1, first transpose
michael@0:         // the samples and then apply the anti-alias filter to remove aliasing.
michael@0: 
michael@0:         // Transpose the samples, store the result to end of "midBuffer"
michael@0:         pTransposer->transpose(midBuffer, inputBuffer);
michael@0: 
michael@0:         // Apply the anti-alias filter for transposed samples in midBuffer
michael@0:         pAAFilter->evaluate(outputBuffer, midBuffer);
michael@0:     } 
michael@0:     else  
michael@0:     {
michael@0:         // If the parameter 'Rate' value is larger than 1, first apply the
michael@0:         // anti-alias filter to remove high frequencies (prevent them from folding
michael@0:         // over the lover frequencies), then transpose.
michael@0: 
michael@0:         // Apply the anti-alias filter for samples in inputBuffer
michael@0:         pAAFilter->evaluate(midBuffer, inputBuffer);
michael@0: 
michael@0:         // Transpose the AA-filtered samples in "midBuffer"
michael@0:         pTransposer->transpose(outputBuffer, midBuffer);
michael@0:     }
michael@0: }
michael@0: 
michael@0: 
michael@0: // Sets the number of channels, 1 = mono, 2 = stereo
michael@0: void RateTransposer::setChannels(int nChannels)
michael@0: {
michael@0:     assert(nChannels > 0);
michael@0: 
michael@0:     if (pTransposer->numChannels == nChannels) return;
michael@0:     pTransposer->setChannels(nChannels);
michael@0: 
michael@0:     inputBuffer.setChannels(nChannels);
michael@0:     midBuffer.setChannels(nChannels);
michael@0:     outputBuffer.setChannels(nChannels);
michael@0: }
michael@0: 
michael@0: 
michael@0: // Clears all the samples in the object
michael@0: void RateTransposer::clear()
michael@0: {
michael@0:     outputBuffer.clear();
michael@0:     midBuffer.clear();
michael@0:     inputBuffer.clear();
michael@0: }
michael@0: 
michael@0: 
michael@0: // Returns nonzero if there aren't any samples available for outputting.
michael@0: int RateTransposer::isEmpty() const
michael@0: {
michael@0:     int res;
michael@0: 
michael@0:     res = FIFOProcessor::isEmpty();
michael@0:     if (res == 0) return 0;
michael@0:     return inputBuffer.isEmpty();
michael@0: }
michael@0: 
michael@0: 
michael@0: //////////////////////////////////////////////////////////////////////////////
michael@0: //
michael@0: // TransposerBase - Base class for interpolation
michael@0: // 
michael@0: 
michael@0: // static function to set interpolation algorithm
michael@0: void TransposerBase::setAlgorithm(TransposerBase::ALGORITHM a)
michael@0: {
michael@0:     TransposerBase::algorithm = a;
michael@0: }
michael@0: 
michael@0: 
michael@0: // Transposes the sample rate of the given samples using linear interpolation. 
michael@0: // Returns the number of samples returned in the "dest" buffer
michael@0: int TransposerBase::transpose(FIFOSampleBuffer &dest, FIFOSampleBuffer &src)
michael@0: {
michael@0:     int numSrcSamples = src.numSamples();
michael@0:     int sizeDemand = (int)((float)numSrcSamples / rate) + 8;
michael@0:     int numOutput;
michael@0:     SAMPLETYPE *psrc = src.ptrBegin();
michael@0:     SAMPLETYPE *pdest = dest.ptrEnd(sizeDemand);
michael@0: 
michael@0: #ifndef USE_MULTICH_ALWAYS
michael@0:     if (numChannels == 1)
michael@0:     {
michael@0:         numOutput = transposeMono(pdest, psrc, numSrcSamples);
michael@0:     }
michael@0:     else if (numChannels == 2) 
michael@0:     {
michael@0:         numOutput = transposeStereo(pdest, psrc, numSrcSamples);
michael@0:     } 
michael@0:     else 
michael@0: #endif // USE_MULTICH_ALWAYS
michael@0:     {
michael@0:         assert(numChannels > 0);
michael@0:         numOutput = transposeMulti(pdest, psrc, numSrcSamples);
michael@0:     }
michael@0:     dest.putSamples(numOutput);
michael@0:     src.receiveSamples(numSrcSamples);
michael@0:     return numOutput;
michael@0: }
michael@0: 
michael@0: 
michael@0: TransposerBase::TransposerBase()
michael@0: {
michael@0:     numChannels = 0;
michael@0:     rate = 1.0f;
michael@0: }
michael@0: 
michael@0: 
michael@0: TransposerBase::~TransposerBase()
michael@0: {
michael@0: }
michael@0: 
michael@0: 
michael@0: void TransposerBase::setChannels(int channels)
michael@0: {
michael@0:     numChannels = channels;
michael@0:     resetRegisters();
michael@0: }
michael@0: 
michael@0: 
michael@0: void TransposerBase::setRate(float newRate)
michael@0: {
michael@0:     rate = newRate;
michael@0: }
michael@0: 
michael@0: 
michael@0: // static factory function
michael@0: TransposerBase *TransposerBase::newInstance()
michael@0: {
michael@0: #ifdef SOUNDTOUCH_INTEGER_SAMPLES
michael@0:     // Notice: For integer arithmetics support only linear algorithm (due to simplest calculus)
michael@0:     return ::new InterpolateLinearInteger;
michael@0: #else
michael@0:     switch (algorithm)
michael@0:     {
michael@0:         case LINEAR:
michael@0:             return new InterpolateLinearFloat;
michael@0: 
michael@0:         case CUBIC:
michael@0:             return new InterpolateCubic;
michael@0: 
michael@0:         case SHANNON:
michael@0:             return new InterpolateShannon;
michael@0: 
michael@0:         default:
michael@0:             assert(false);
michael@0:             return NULL;
michael@0:     }
michael@0: #endif
michael@0: }