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diff --git a/media/libsoundtouch/src/RateTransposer.cpp b/media/libsoundtouch/src/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 'at' iki.fi
+/// SoundTouch WWW: http://www.surina.net/soundtouch
+///
+////////////////////////////////////////////////////////////////////////////////
+//
+// 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 = 
+#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
+        true;
+#else
+        // Disable Anti-alias filter if desirable to avoid click at rate change zero value crossover
+        false;
+#endif
+
+    // Instantiates the anti-alias filter
+    pAAFilter = new AAFilter(64);
+    pTransposer = TransposerBase::newInstance();
+    clear();
+}
+
+
+RateTransposer::~RateTransposer()
+{
+    delete pAAFilter;
+    delete pTransposer;
+}
+
+
+/// Enables/disables the anti-alias filter. Zero to disable, nonzero to enable
+void RateTransposer::enableAAFilter(bool newMode)
+{
+#ifndef SOUNDTOUCH_PREVENT_CLICK_AT_RATE_CROSSOVER
+    // Disable Anti-alias filter if desirable to avoid click at rate change zero value crossover
+    bUseAAFilter = newMode;
+    clear();
+#endif
+}
+
+
+/// 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(double newRate)
+{
+    double fCutoff;
+
+    pTransposer->setRate(newRate);
+
+    // design a new anti-alias filter
+    if (newRate > 1.0) 
+    {
+        fCutoff = 0.5 / newRate;
+    } 
+    else 
+    {
+        fCutoff = 0.5 * 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)
+{
+    if (!verifyNumberOfChannels(nChannels) ||
+        (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();
+    pTransposer->resetRegisters();
+
+    // prefill buffer to avoid losing first samples at beginning of stream
+    int prefill = getLatency();
+    inputBuffer.addSilent(prefill);
+}
+
+
+// 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();
+}
+
+
+/// Return approximate initial input-output latency
+int RateTransposer::getLatency() const
+{
+    return pTransposer->getLatency() +
+        ((bUseAAFilter) ? (pAAFilter->getLength() / 2) : 0);
+}
+
+
+//////////////////////////////////////////////////////////////////////////////
+//
+// 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)((double)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(double newRate)
+{
+    rate = newRate;
+}
+
+
+// static factory function
+TransposerBase *TransposerBase::newInstance()
+{
+#ifdef SOUNDTOUCH_INTEGER_SAMPLES
+    // Notice: For integer arithmetic 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
+}