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Diffstat (limited to 'src/thirdparty/SoundTouch/source/RateTransposer.cpp')
| -rw-r--r-- | src/thirdparty/SoundTouch/source/RateTransposer.cpp | 302 |
1 files changed, 0 insertions, 302 deletions
diff --git a/src/thirdparty/SoundTouch/source/RateTransposer.cpp b/src/thirdparty/SoundTouch/source/RateTransposer.cpp deleted file mode 100644 index 6b0dd0a5c..000000000 --- a/src/thirdparty/SoundTouch/source/RateTransposer.cpp +++ /dev/null @@ -1,302 +0,0 @@ -////////////////////////////////////////////////////////////////////////////////
-///
-/// 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: 2014-01-06 19:19:38 +0000 (Mon, 06 Jan 2014) $
-// File revision : $Revision: 4 $
-//
-// $Id: RateTransposer.cpp 181 2014-01-06 19:19:38Z 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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