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Diffstat (limited to 'media/libsoundtouch/src/RateTransposer.cpp')
| -rw-r--r-- | media/libsoundtouch/src/RateTransposer.cpp | 302 |
1 files changed, 302 insertions, 0 deletions
diff --git a/media/libsoundtouch/src/RateTransposer.cpp b/media/libsoundtouch/src/RateTransposer.cpp new file mode 100644 index 000000000..f1e3fd043 --- /dev/null +++ b/media/libsoundtouch/src/RateTransposer.cpp @@ -0,0 +1,302 @@ +//////////////////////////////////////////////////////////////////////////////// +/// +/// 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 15:57:21 +0000 (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 +} |