void AudioProcessing::convolution( const juce::AudioSampleBuffer & a, const juce::AudioSampleBuffer & b, juce::AudioSampleBuffer & result )
{
jassert( b.getNumChannels() == 1 );
int sampleSize = a.getNumSamples() + b.getNumSamples();
result.setSize( a.getNumChannels(), sampleSize );
const float * bSignal = b.getArrayOfReadPointers()[ 0 ];
for ( int ch = 0 ; ch < a.getNumChannels() ; ++ch )
{
const float * aSignal = a.getArrayOfReadPointers()[ ch ];
float * res = result.getArrayOfWritePointers()[ ch ];
for ( int i = 0 ; i < sampleSize ; ++i )
{
float sum = 0.f;
for ( int j = 0 ; j < b.getNumSamples() ; ++j )
{
int indexA = i - j;
if ( ( indexA >= 0 ) && ( indexA < a.getNumSamples() ) )
sum += ( aSignal[indexA] * bSignal[j] );
}
res[i] = sum;
}
}
}
/**
Upsample by 4 signal (polyphase coefficients are for a 48 kHz signal).
Processes 4 samples of the output signal in a single loop iteration.
In one loop, each of the 4 processed samples is filtered with its own set of coefficients
*/
void AudioProcessing::polyphase4( const juce::AudioSampleBuffer & source, juce::AudioSampleBuffer & result )
{
int sampleSize = source.getNumSamples();
result.setSize( source.getNumChannels(), 4*source.getNumSamples() + numCoeffs, false, false, true );
for ( int ch = 0 ; ch < source.getNumChannels() ; ++ch )
{
const float * input = source.getArrayOfReadPointers()[ ch ];
float * res = result.getArrayOfWritePointers()[ ch ];
for ( int i = 0 ; i < sampleSize ; ++i )
{
res[4*i] = polyphase4ComputeSum( input, i, source.getNumSamples(), filterPhase0, numCoeffs );
res[4*i+1] = polyphase4ComputeSum( input, i, source.getNumSamples(), filterPhase1, numCoeffs );
res[4*i+2] = polyphase4ComputeSum( input, i, source.getNumSamples(), filterPhase2, numCoeffs );
res[4*i+3] = polyphase4ComputeSum( input, i, source.getNumSamples(), filterPhase3, numCoeffs );
}
}
}