HELP! Resampling Effect (Halftime) using Interpolators

1

Hi guys! I’m trying to code a plugin that changes sound in a similar way as Half-Time does. It is the same effect as when you resample a sound using a lower samplerate.
I tried to use JUCE Interpolators to resample every buffer in the processBlock. Here is the code I´m trying to make it work:

void RESAMPLINGAudioProcessor::processBlock (juce::AudioBuffer& buffer, juce::MidiBuffer& midiMessages)
{
juce::ScopedNoDenormals noDenormals;
auto totalNumInputChannels = getTotalNumInputChannels();
auto totalNumOutputChannels = getTotalNumOutputChannels();

juce::LagrangeInterpolator Resampler;
juce::AudioBuffer<float> bufferResampleado(totalNumInputChannels,buffer.getNumSamples()/2); //buffer of 128 samples and two channels


for (auto i = totalNumInputChannels; i < totalNumOutputChannels; ++i)
    buffer.clear (i, 0, buffer.getNumSamples());
    bufferResampleado.clear();


for (int channel = 0; channel < totalNumInputChannels; ++channel)
{
    auto* channelData = bufferResampleado.getWritePointer (channel);
    auto* channelDataRead = buffer.getReadPointer(channel);

    
    Resampler.process(1.378125, channelDataRead, channelData, bufferResampleado.getNumSamples());

    buffer = bufferResampleado;
    Resampler.reset();
}

}

I’m doing something wrong as the only thing I hear when I load my plugin are my sounds being crushed with distortion.

Anyone has some idea? Do you think I’m goin the right way?

2

One thing that jumps out at me from the docs:

And like with any other stateful filter, if you’re resampling multiple channels, make sure each one uses its own LagrangeInterpolator object.

Try creating a left and a right resampler and see how that works?

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3

the resampler and its buffer likely need to be member variables, because they are stateful

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