Downsampling using interpolators

Audio Plugins
1

Hi guys,

for a project I’m working on, I need to process a signal at a fixed sample rate. I’m trying to resample the incoming buffer with either Lagrange or Catmull-Rom interpolators, but the result shows an heavy aliasing/discontinuities, even if I play a 100Hz sine.

I have a buffer to store the decimated samples and, for each channel, two interpolators (one to lower the sample rate and another to bring it back to the system’s one).

In prepareToPlay:

const double decimRatio = sampleRate / 18000.0;
const int decimNumSamples = static_cast<int>(samplesPerBlock / decimRatio);

lowResBuffer.setSize(2, decimNumSamples);


//here the interpolators are reset

Then, I have a method to perform the resampling. For testing purposes, I do both the decimation and upsampling in the same place.

void MyProject::ResampleBuffer(AudioBuffer<float>& inBuffer)
{
	//Decimate to target SR
	const float** inputs = inBuffer.getArrayOfReadPointers();
	float** outputs = lowResBuffer.getArrayOfWritePointers();
	const double decimRatio = getSampleRate() / 18000.0;

	for(int chan = 0; chan < inBuffer.getNumChannels(); ++chan)
	{
		decimator[chan]->process(decimRatio, inputs[chan], outputs[chan], lowResBuffer.getNumSamples());
	}

	//Upsample to original SR
	const float** inputs = lowResBuffer.getArrayOfReadPointers();
	float** outputs = inBuffer.getArrayOfWritePointers();
	const double upsampRatio = 18000.0 / getSampleRate();

	for(int chan = 0; chan < inBuffer.getNumChannels(); ++chan)
	{
		upsampler[chan]->process(upsampRatio, inputs[chan], outputs[chan], inBuffer.getNumSamples(), lowResBuffer.getNumSamples(), true);
	}
}

It sounds like the output samples produced are not the same as the expected, but AFAIK the interpolator methods won’t return the number of samples produced.

Any hints?
Thanks

2

In which way does the result differ from the input?