LagrangeInterpolator with fractional ratio

[pauloesteban]

@Daniel Found this use of LagrangeInterpolator for resampling.

Resampling Audio using Lagrange Interpolator

void fileConvert(const File& file, const double& dOutSampleRate) { AudioFormatManager formatManager; formatManager.registerBasicFormats(); AudioFormatReader* reader = formatManager.createReaderFor(file); if (reader) { if (reader->lengthInSamples > 0) { if (reader->sampleRate != dOutSampleRate) { //find the out file number of samples juce::int64 iSamples = ceil(dOutSampleRate * reader->lengthInSamples / reader->sampleRate); AudioSampleBuffer tempBuffer(reader->numChannels, iS…

Apparently it works if it is applied to the whole file not buffering. In the case of realtime are you suggesting using an AbstractFIFO?

Many thanks in advance.