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* DSP: Implement Pipe 2 Pipe 2 is a DSP pipe that is used to initialize both the DSP hardware (the application signals to the DSP to initialize) and the application (the DSP provides the memory location of structures in the shared memory region). * AudioCore: Implement codecs (DecodeADPCM, DecodePCM8, DecodePCM16) * DSP Pipes: Implement as FIFO * AudioCore: File structure * AudioCore: More structure * AudioCore: Buffer management * DSP/Source: Reorganise Source's AdvanceFrame. * Audio Output * lolidk * huh? * interp * More interp stuff * oops * Zero State * Don't mix Source frame if it's not enabled * DSP: Forgot to zero a buffer, adjusted thread synchronisation, adjusted format spec for buffers * asdf * Get it to compile and tweak stretching a bit. * revert stretch test * deleted accidental partial catch submodule commit * new audio stretching algorithm * update .gitmodule * fix OS X build * remove getopt from rubberband * #include <stddef> to audio_core.h * typo * -framework Accelerate * OptionTransientsSmooth -> OptionTransientsCrisp * tweak stretch tempo smoothing coefficient. also switch back to smooth. * tweak mroe * remove printf * sola * #include <cmath> * VERY QUICK MERGE TO GET IT WORKING DOESN'T ACTIVATE AUDIO FILTERS * Reminder to self * fix comparison * common/thread: Correct code style * Thread: Make Barrier reusable * fix threading synchonisation code * add profiling code * print error to console when audio clips * fix metallic sound * reduce logspam
301 lines
7.8 KiB
C++
301 lines
7.8 KiB
C++
////////////////////////////////////////////////////////////////////////////////
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///
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/// Linear interpolation algorithm.
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///
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/// Author : Copyright (c) Olli Parviainen
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/// Author e-mail : oparviai 'at' iki.fi
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/// SoundTouch WWW: http://www.surina.net/soundtouch
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///
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////////////////////////////////////////////////////////////////////////////////
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//
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// $Id: InterpolateLinear.cpp 225 2015-07-26 14:45:48Z oparviai $
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//
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////////////////////////////////////////////////////////////////////////////////
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//
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// License :
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//
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// SoundTouch audio processing library
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// Copyright (c) Olli Parviainen
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//
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// This library is free software; you can redistribute it and/or
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// modify it under the terms of the GNU Lesser General Public
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// License as published by the Free Software Foundation; either
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// version 2.1 of the License, or (at your option) any later version.
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//
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// This library is distributed in the hope that it will be useful,
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// but WITHOUT ANY WARRANTY; without even the implied warranty of
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// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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// Lesser General Public License for more details.
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//
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// You should have received a copy of the GNU Lesser General Public
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// License along with this library; if not, write to the Free Software
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// Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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//
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////////////////////////////////////////////////////////////////////////////////
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#include <assert.h>
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#include <stdlib.h>
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#include "InterpolateLinear.h"
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using namespace soundtouch;
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//////////////////////////////////////////////////////////////////////////////
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//
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// InterpolateLinearInteger - integer arithmetic implementation
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//
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/// fixed-point interpolation routine precision
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#define SCALE 65536
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// Constructor
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InterpolateLinearInteger::InterpolateLinearInteger() : TransposerBase()
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{
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// Notice: use local function calling syntax for sake of clarity,
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// to indicate the fact that C++ constructor can't call virtual functions.
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resetRegisters();
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setRate(1.0f);
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}
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void InterpolateLinearInteger::resetRegisters()
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{
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iFract = 0;
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}
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// Transposes the sample rate of the given samples using linear interpolation.
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// 'Mono' version of the routine. Returns the number of samples returned in
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// the "dest" buffer
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int InterpolateLinearInteger::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
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{
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int i;
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int srcSampleEnd = srcSamples - 1;
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int srcCount = 0;
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i = 0;
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while (srcCount < srcSampleEnd)
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{
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LONG_SAMPLETYPE temp;
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assert(iFract < SCALE);
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temp = (SCALE - iFract) * src[0] + iFract * src[1];
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dest[i] = (SAMPLETYPE)(temp / SCALE);
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i++;
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iFract += iRate;
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int iWhole = iFract / SCALE;
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iFract -= iWhole * SCALE;
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srcCount += iWhole;
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src += iWhole;
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}
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srcSamples = srcCount;
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return i;
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}
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// Transposes the sample rate of the given samples using linear interpolation.
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// 'Stereo' version of the routine. Returns the number of samples returned in
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// the "dest" buffer
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int InterpolateLinearInteger::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
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{
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int i;
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int srcSampleEnd = srcSamples - 1;
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int srcCount = 0;
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i = 0;
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while (srcCount < srcSampleEnd)
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{
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LONG_SAMPLETYPE temp0;
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LONG_SAMPLETYPE temp1;
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assert(iFract < SCALE);
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temp0 = (SCALE - iFract) * src[0] + iFract * src[2];
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temp1 = (SCALE - iFract) * src[1] + iFract * src[3];
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dest[0] = (SAMPLETYPE)(temp0 / SCALE);
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dest[1] = (SAMPLETYPE)(temp1 / SCALE);
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dest += 2;
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i++;
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iFract += iRate;
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int iWhole = iFract / SCALE;
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iFract -= iWhole * SCALE;
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srcCount += iWhole;
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src += 2*iWhole;
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}
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srcSamples = srcCount;
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return i;
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}
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int InterpolateLinearInteger::transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
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{
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int i;
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int srcSampleEnd = srcSamples - 1;
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int srcCount = 0;
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i = 0;
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while (srcCount < srcSampleEnd)
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{
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LONG_SAMPLETYPE temp, vol1;
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assert(iFract < SCALE);
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vol1 = (SCALE - iFract);
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for (int c = 0; c < numChannels; c ++)
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{
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temp = vol1 * src[c] + iFract * src[c + numChannels];
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dest[0] = (SAMPLETYPE)(temp / SCALE);
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dest ++;
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}
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i++;
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iFract += iRate;
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int iWhole = iFract / SCALE;
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iFract -= iWhole * SCALE;
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srcCount += iWhole;
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src += iWhole * numChannels;
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}
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srcSamples = srcCount;
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return i;
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}
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// Sets new target iRate. Normal iRate = 1.0, smaller values represent slower
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// iRate, larger faster iRates.
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void InterpolateLinearInteger::setRate(double newRate)
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{
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iRate = (int)(newRate * SCALE + 0.5);
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TransposerBase::setRate(newRate);
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}
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//////////////////////////////////////////////////////////////////////////////
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//
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// InterpolateLinearFloat - floating point arithmetic implementation
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//
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//////////////////////////////////////////////////////////////////////////////
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// Constructor
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InterpolateLinearFloat::InterpolateLinearFloat() : TransposerBase()
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{
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// Notice: use local function calling syntax for sake of clarity,
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// to indicate the fact that C++ constructor can't call virtual functions.
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resetRegisters();
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setRate(1.0);
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}
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void InterpolateLinearFloat::resetRegisters()
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{
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fract = 0;
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}
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// Transposes the sample rate of the given samples using linear interpolation.
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// 'Mono' version of the routine. Returns the number of samples returned in
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// the "dest" buffer
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int InterpolateLinearFloat::transposeMono(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
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{
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int i;
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int srcSampleEnd = srcSamples - 1;
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int srcCount = 0;
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i = 0;
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while (srcCount < srcSampleEnd)
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{
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double out;
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assert(fract < 1.0);
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out = (1.0 - fract) * src[0] + fract * src[1];
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dest[i] = (SAMPLETYPE)out;
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i ++;
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// update position fraction
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fract += rate;
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// update whole positions
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int whole = (int)fract;
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fract -= whole;
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src += whole;
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srcCount += whole;
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}
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srcSamples = srcCount;
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return i;
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}
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// Transposes the sample rate of the given samples using linear interpolation.
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// 'Mono' version of the routine. Returns the number of samples returned in
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// the "dest" buffer
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int InterpolateLinearFloat::transposeStereo(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
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{
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int i;
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int srcSampleEnd = srcSamples - 1;
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int srcCount = 0;
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i = 0;
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while (srcCount < srcSampleEnd)
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{
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double out0, out1;
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assert(fract < 1.0);
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out0 = (1.0 - fract) * src[0] + fract * src[2];
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out1 = (1.0 - fract) * src[1] + fract * src[3];
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dest[2*i] = (SAMPLETYPE)out0;
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dest[2*i+1] = (SAMPLETYPE)out1;
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i ++;
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// update position fraction
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fract += rate;
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// update whole positions
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int whole = (int)fract;
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fract -= whole;
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src += 2*whole;
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srcCount += whole;
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}
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srcSamples = srcCount;
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return i;
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}
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int InterpolateLinearFloat::transposeMulti(SAMPLETYPE *dest, const SAMPLETYPE *src, int &srcSamples)
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{
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int i;
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int srcSampleEnd = srcSamples - 1;
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int srcCount = 0;
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i = 0;
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while (srcCount < srcSampleEnd)
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{
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float temp, vol1, fract_float;
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vol1 = (float)(1.0 - fract);
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fract_float = (float)fract;
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for (int c = 0; c < numChannels; c ++)
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{
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temp = vol1 * src[c] + fract_float * src[c + numChannels];
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*dest = (SAMPLETYPE)temp;
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dest ++;
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}
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i++;
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fract += rate;
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int iWhole = (int)fract;
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fract -= iWhole;
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srcCount += iWhole;
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src += iWhole * numChannels;
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}
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srcSamples = srcCount;
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return i;
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}
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