TimeStretcher::TimeStretcher(const SoundFile &soundFile) {
this->soundFile = soundFile;
channels = soundFile.getChannels();
inputSamples = soundFile.getSamples();
stretchInBufL.resize(maxProcessSize);
stretchInBufR.resize(maxProcessSize);
stretchInBuf.resize(channels);
stretchInBuf[0] = &(stretchInBufL[0]);
stretchInBuf[1] = &(stretchInBufR[0]);
stretchOutBufL.resize(maxProcessSize);
stretchOutBufR.resize(maxProcessSize);
stretchOutBuf.resize(channels);
stretchOutBuf[0] = &(stretchOutBufL[0]);
stretchOutBuf[1] = &(stretchOutBufR[0]);
rubberband = new RubberBand::RubberBandStretcher(
soundFile.getSampleRate(), channels,
RubberBand::RubberBandStretcher::DefaultOptions |
RubberBand::RubberBandStretcher::OptionProcessRealTime);
rubberband->setMaxProcessSize(maxProcessSize);
playheadPos = 0;
}
int main(int argc, char* argv[])
{
#ifdef TESTS
cout << "===== RUNNING TESTS =====" << endl;
RegressionTest::runAllTests();
TestConvolver::runAllTests();
cout << "==== TESTS COMPLETED ====" << endl;
system("pause");
#endif
/* Ensure that valid dry, IR, and output file names are provided */
if (! checkArgs(argc, argv))
return 1;
/* Read wave files */
SoundFile* dry = SoundFile::create(argv[1]);
SoundFile* ir = SoundFile::create(argv[2]);
/* End program if any input file is corrupted or nonexistant */
if (dry == nullptr || ir == nullptr)
return 1;
int dryDataSize = dry->getDataSize();
int irDataSize = ir->getDataSize();
cout << "DR Size: " << dryDataSize << " IR Size: " << irDataSize << endl;
int P; // Length of result
short* result; // Resulting data of convolution to write to file
/* Normalize the input data to -1 and +1 */
double* dryNormalized = new double[dryDataSize];
double* irNormalized = new double[irDataSize];
Convolver::dataToSignal(dry->getData(), dryDataSize, dry->getAbsMinValue(), dryNormalized);
Convolver::dataToSignal(ir->getData(), irDataSize, ir->getAbsMinValue(), irNormalized);
if (ir->getNumChannels() == 1) {
P = dryDataSize + irDataSize - 1;
/* Perform time domain convolution */
result = new short[P];
#ifndef FFT
Convolver::convolve(dryNormalized, dryDataSize, irNormalized, irDataSize, result, P);
#else
Convolver::fftConvolve(dryNormalized, dryDataSize, irNormalized, irDataSize, result, P);
#endif
/* Save resulting .wav file */
SoundFile::save(argv[3], ir->getNumChannels(), dry->getBitsPerSample(), dry->getSampleRate(), result, P);
}
else if (ir->getNumChannels() == 2) {
/* Calculate half size of impulse response ONCE and store in halfM */
int halfM = irDataSize >> 1;
/* P = N+(M/2)-1, where P is the length of each half of this two-channel convolution */
P = dryDataSize + halfM - 1;
/* Split normalized stereo impulse response into left and right channels */
double* irLeftNormalized = new double[halfM];
double* irRightNormalized = new double[halfM];
for (int i = 0, i2 = 0; i < halfM; i++) {
i2 = i << 1;
irLeftNormalized[i] = irNormalized[i2];
irRightNormalized[i] = irNormalized[i2 + 1];
}
/* Perform time domain convolution */
short* resultLeft = new short[P];
short* resultRight = new short[P];
#ifndef FFT
Convolver::convolve(dryNormalized, dryDataSize, irLeftNormalized, halfM, resultLeft, P);
Convolver::convolve(dryNormalized, dryDataSize, irRightNormalized, halfM, resultRight, P);
#else
Convolver::fftConvolve(dryNormalized, dryDataSize, irLeftNormalized, halfM, resultLeft, P);
Convolver::fftConvolve(dryNormalized, dryDataSize, irRightNormalized, halfM, resultRight, P);
#endif
/* Interleave left and right channel data */
result = new short[P*2];
for (int i = 0, halfI = 0; i < P*2; i+=2) {
halfI = i >> 1;
result[i] = resultLeft[halfI];
result[i+1] = resultRight[halfI];
}
/* Save resulting .wav file */
SoundFile::save(argv[3], ir->getNumChannels(), dry->getBitsPerSample(), dry->getSampleRate(), result, P*2);
delete[] irLeftNormalized;
delete[] irRightNormalized;
delete[] resultLeft;
delete[] resultRight;
}
