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;
}
}
}
void AudioProcessing::TruePeak::process( const juce::AudioSampleBuffer & buffer )
{
if (m_inputs.getNumSamples() >= numCoeffs)
{
// we have enough data from a previous process
for ( int ch = 0 ; ch < buffer.getNumChannels() ; ++ch )
{
m_inputs.copyFrom( ch, 0, &m_inputs.getArrayOfReadPointers()[ch][m_inputs.getNumSamples() - numCoeffs], numCoeffs );
}
// resize if necessary
if ( m_inputs.getNumSamples() != numCoeffs + buffer.getNumSamples() )
m_inputs.setSize( buffer.getNumChannels(), numCoeffs + buffer.getNumSamples(), true, false, true );
}
else
{
// setSize clears buffer content too
m_inputs.setSize( buffer.getNumChannels(), numCoeffs + buffer.getNumSamples(), false, true );
}
// copy buffer to inputs with numCoefs offset
for ( int ch = 0 ; ch < buffer.getNumChannels() ; ++ch )
{
m_inputs.copyFrom( ch, numCoeffs, buffer, ch, 0, buffer.getNumSamples() );
}
processPolyphase4AbsMax( m_inputs );
}
void AudioProcessing::TruePeak::processPolyphase4AbsMax( const juce::AudioSampleBuffer & buffer )
{
// reset current tru peak
m_truePeakValue = 0.f;
memset( m_truePeakChannelArray, 0, sizeof( m_truePeakChannelArray ) );
int sampleSize = buffer.getNumSamples();
for ( int ch = 0 ; ch < buffer.getNumChannels() ; ++ch )
{
const float * input = buffer.getArrayOfReadPointers()[ ch ];
for ( int i = 0 ; i < sampleSize ; ++i )
{
for ( int j = 0 ; j < 4 ; ++j ) // number of polyphase filters
{
float absSample = fabs( polyphase4ComputeSum( input, i, buffer.getNumSamples(), filterPhaseArray[j], numCoeffs ) );
if ( absSample > m_truePeakValue )
m_truePeakValue = absSample;
if ( absSample > m_truePeakChannelArray[ch] )
m_truePeakChannelArray[ch] = absSample;
}
}
}
}
/**
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 );
}
}
}