The Tor Browser: media/libsoundtouch/src/RateTransposer.cpp@97036ab72558 (annotated)

media/libsoundtouch/src/RateTransposer.cpp@97036ab72558 (annotated)

media/libsoundtouch/src/RateTransposer.cpp

Tue, 06 Jan 2015 21:39:09 +0100

author: Michael Schloh von Bennewitz <michael@schloh.com>
date: Tue, 06 Jan 2015 21:39:09 +0100
branch: TOR_BUG_9701
changeset 8: 97036ab72558
permissions: -rw-r--r--

Conditionally force memory storage according to privacy.thirdparty.isolate;
This solves Tor bug #9701, complying with disk avoidance documented in
https://www.torproject.org/projects/torbrowser/design/#disk-avoidance.

 ////////////////////////////////////////////////////////////////////////////////
 ///
 /// 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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media/libsoundtouch/src/RateTransposer.cpp@97036ab72558 (annotated)

media/libsoundtouch/src/RateTransposer.cpp