DIY and no more libraries.

Real-time audio processing.js

function dsp(time){
	var master=Math.sin(Math.PI*2*440*time)*Math.exp(-3*time);
	return master;
}
var audioContext=new AudioContext(),
    bufferSize=2048, //256,512,1024,2048,4096,8192,16384
    node=audioContext.createScriptProcessor(bufferSize,0,2), //bufferSize,inputChannels,outputChannels
    time=0;
node.onaudioprocess=event=>{
	var L=event.outputBuffer.getChannelData(0),
	    R=event.outputBuffer.getChannelData(1),
	    value;
	for(var i=0;i<bufferSize;i++){
		value=dsp(time+(i/audioContext.sampleRate));
		if(Array.isArray(value)){ //Stereo.
			L[i]=value[0];R[i]=value[1];
		}else{ //Mono.
			L[i]=value;R[i]=value;
		}
	}
	time+=bufferSize/audioContext.sampleRate;
};
node.connect(audioContext.destination);

Oh no, I shouldn't have used event.inputBuffer here where it was used for source.connect(scriptNode); in MDN's example, therefore no input channels should've been used in this snippet. source.start(); was used to play the audio file that's loaded asynchronously into source.buffer.
I now have to go through my p5.js projects because of this. But I knew already that I've done something wrong here (its mild verbosity can be seen, and uses input.length as if it were the size of the output buffer) but it just took a while for me to figure out why.

But I've fixed it now. :^)