The Tor Browser: comparison media/libsoundtouch/src/RateTransposer.cpp

--1:000000000000
+:26f749190435
+////////////////////////////////////////////////////////////////////////////////
+///
+/// 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: 2014-04-06 10:57:21 -0500 (Sun, 06 Apr 2014) $
+// File revision : $Revision: 4 $
+//
+// $Id: RateTransposer.cpp 195 2014-04-06 15:57:21Z 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 <memory.h>
+#include <assert.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include "RateTransposer.h"
+#include "InterpolateLinear.h"
+#include "InterpolateCubic.h"
+#include "InterpolateShannon.h"
+#include "AAFilter.h"
+using namespace soundtouch;
+// Define default interpolation algorithm here
+TransposerBase::ALGORITHM TransposerBase::algorithm = TransposerBase::CUBIC;
+// Constructor
+RateTransposer::RateTransposer() : FIFOProcessor(&outputBuffer)
+{
+bUseAAFilter = true;
+// Instantiates the anti-alias filter
+pAAFilter = new AAFilter(64);
+pTransposer = TransposerBase::newInstance();
+}
+RateTransposer::~RateTransposer()
+{
+delete pAAFilter;
+delete pTransposer;
+}
+/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
+void RateTransposer::enableAAFilter(bool newMode)
+{
+bUseAAFilter = newMode;
+}
+/// Returns nonzero if anti-alias filter is enabled.
+bool RateTransposer::isAAFilterEnabled() const
+{
+return bUseAAFilter;
+}
+AAFilter *RateTransposer::getAAFilter()
+{
+return pAAFilter;
+}
+// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower
+// iRate, larger faster iRates.
+void RateTransposer::setRate(float newRate)
+{
+double fCutoff;
+pTransposer->setRate(newRate);
+// design a new anti-alias filter
+if (newRate > 1.0f)
+{
+fCutoff = 0.5f / newRate;
+}
+else
+{
+fCutoff = 0.5f * newRate;
+}
+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, uint nSamples)
+{
+processSamples(samples, nSamples);
+}
+// 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 nSamples)
+{
+uint count;
+if (nSamples == 0) return;
+// 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)
+{
+count = pTransposer->transpose(outputBuffer, inputBuffer);
+return;
+}
+assert(pAAFilter);
+// Transpose with anti-alias filter
+if (pTransposer->rate < 1.0f)
+{
+// 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
+{
+// 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);
+}
+}
+// Sets the number of channels, 1 = mono, 2 = stereo
+void RateTransposer::setChannels(int nChannels)
+{
+assert(nChannels > 0);
+if (pTransposer->numChannels == nChannels) return;
+pTransposer->setChannels(nChannels);
+inputBuffer.setChannels(nChannels);
+midBuffer.setChannels(nChannels);
+outputBuffer.setChannels(nChannels);
+}
+// Clears all the samples in the object
+void RateTransposer::clear()
+{
+outputBuffer.clear();
+midBuffer.clear();
+inputBuffer.clear();
+}
+// Returns nonzero if there aren't any samples available for outputting.
+int RateTransposer::isEmpty() const
+{
+int res;
+res = FIFOProcessor::isEmpty();
+if (res == 0) return 0;
+return inputBuffer.isEmpty();
+}
+//////////////////////////////////////////////////////////////////////////////
+//
+// TransposerBase - Base class for interpolation
+//
+// static function to set interpolation algorithm
+void TransposerBase::setAlgorithm(TransposerBase::ALGORITHM a)
+{
+TransposerBase::algorithm = a;
+}
+// Transposes the sample rate of the given samples using linear interpolation.
+// Returns the number of samples returned in the "dest" buffer
+int TransposerBase::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)
+{
+numOutput = transposeMono(pdest, psrc, numSrcSamples);
+}
+else if (numChannels == 2)
+{
+numOutput = transposeStereo(pdest, psrc, numSrcSamples);
+}
+else
+#endif // USE_MULTICH_ALWAYS
+{
+assert(numChannels > 0);
+numOutput = transposeMulti(pdest, psrc, numSrcSamples);
+}
+dest.putSamples(numOutput);
+src.receiveSamples(numSrcSamples);
+return numOutput;
+}
+TransposerBase::TransposerBase()
+{
+numChannels = 0;
+rate = 1.0f;
+}
+TransposerBase::~TransposerBase()
+{
+}
+void TransposerBase::setChannels(int channels)
+{
+numChannels = channels;
+resetRegisters();
+}
+void TransposerBase::setRate(float newRate)
+{
+rate = newRate;
+}
+// static factory function
+TransposerBase *TransposerBase::newInstance()
+{
+#ifdef SOUNDTOUCH_INTEGER_SAMPLES
+// Notice: For integer arithmetics support only linear algorithm (due to simplest calculus)
+return ::new InterpolateLinearInteger;
+#else
+switch (algorithm)
+{
+case LINEAR:
+return new InterpolateLinearFloat;
+case CUBIC:
+return new InterpolateCubic;
+case SHANNON:
+return new InterpolateShannon;
+default:
+assert(false);
+return NULL;
+}
+#endif
+}

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comparison: media/libsoundtouch/src/RateTransposer.cpp

media/libsoundtouch/src/RateTransposer.cpp