Downsampling PCM Audio

Question

I'm looking everywhere for solutions for audio downsampling. However for one reason or another they are incomplete, sometimes buggy, and sometimes are in a programming language I cannot understand.

So here's my use case:

I have an array containing two channels (left-right, it is a microphone) with Float32 data that gets interleaved and then converted to PCM like this:

function interleave(left, right) {
    var length = left.length + right.length;
    var result = new Float32Array(length);
    var _index = 0;
    for(var index = 0; index < length;) {
        result[index++] = left[_index];
        result[index++] = right[_index];
        _index++;
    }
    return result;
}

function convertToPCM(raw) {
    var output = new Buffer(raw.length*2);
    for (var i = 0, offset = 0; i < raw.length; i++, offset += 2){
        var s = Math.max(-1, Math.min(1, raw[i]));
        output.writeInt16LE(s < 0 ? s*0x8000 : s*0x7FFF, offset);
    }
    return output;
}

This is NodeJS\Javascript, but I think it's pretty clear how it works.

Now here's the issue. The output of these function needs to generate a WAV file downsampled to 22000 or 16000, but the source has a sample rate of 44100.

Could you please highlight what algorithm should I use in order to achieve this?

Thank you very much!

Answer 1

Breaking down the conversion into an upsampling and a downsampling step as explained in mbaz's answer is not necessary - band-limited interpolation achieves the same result in one single process, for any conversion ratio, and its quality is easy to fine-tune (through various approximations of the band-limited pulse).

libsamplerate is an example of open-source implementation that does not rely on upsampling/downsampling. It should be possible to bind it to NodeJS through an add-on, or even to rewrite it in javascript - once stripped of the code handling corner cases (variable rate, compensation of processing delay), the code fits in a 50-line loop.