问题
I'm building an emulation of the Roland Juno-106 synthesizer using WebAudio. The live WIP version is here.
I'm hung up on how to deal with updating the filter if the cutoff frequency or envelope modulation amount are changed during the attack or release while the filter is simultaneously being modulated by the envelope. That code is located around here. The current implementation doesn't respond the way an analog synth would, but I can't quite figure out how to calculate it.
On a real synth the filter changes immediately as determined by the frequency cutoff, envelope modulation amount, and current stage in the envelope, but the ramp up or down also continues smoothly.
How would I model this behavior?
回答1:
Brilliant project!
You don't need to sum these yourself - Web Audio AudioParams sum their inputs, so if you have a potentially audio-rate modulation source like an LFO (an OscillatorNode connected to a GainNode), you simply connect() it to the AudioParam.
This is the key here - that AudioParams are able to be connect()ed to - and multiple input connections to a node or AudioParam are summed. So you generally want a model of
filter cutoff = (cutoff from envelope) + (cutoff from mod/LFO) + (cutoff from cutoff knob)
Since cutoff is a frequency, and thus on a log scale not a linear one, you want to do this addition logarithmically (otherwise, an envelope that boosts the cutoff up an octave at 440Hz will only boost it half an octave at 880Hz, etc.) - which, luckily, is easy to do via the "detune" parameter on a BiquadFilter.
Detune is in cents (1200/octave), so you have to use gain nodes to adjust values (e.g. if you want your modulation to have a +1/-1 octave range, make sure the oscillator output is going between -1200 and +1200). You can see how I do this bit in my Web Audio synthesizer (https://github.com/cwilso/midi-synth): in particular, check out synth.js starting around line 500: https://github.com/cwilso/midi-synth/blob/master/js/synth.js#L497-L519. Note the modFilterGain.connect(this.filter1.detune); in particular.
You don't want to be setting ANY values directly for modulation, since the actual value will change at a potentially fast rate - you want to use the parameter scheduler and input summing from an LFO. You can set the knob value as needed in terms of time, but it turns out that setting .value will interact poorly with setting scheduled values on the same AudioParam - so you'll need to have a separate (summed) input into the AudioParam. This is the tricky bit, and to be honest, my synth does NOT do this well today (I should change it to the approach described below).
The right way to handle the knob setting is to create an audio channel that varies based on your knob setting - that is, it's an AudioNode that you can connect() to the filter.detune, although the sample values produced by that AudioNode are only positive, and only change values when the knob is changed. To do this, you need a DC offset source - that is, an AudioNode that produces a stream of constant sample values. The simplest way I can think of to do this is to use an AudioBufferSourceNode with a generated buffer of 1:
function createDCOffset() {
var buffer=audioContext.createBuffer(1,1,audioContext.sampleRate);
var data = buffer.getChannelData(0);
data[0]=1;
var bufferSource=audioContext.createBufferSource();
bufferSource.buffer=buffer;
bufferSource.loop=true;
bufferSource.start(0);
return bufferSource;
}
Then, just connect that DCOffset into a gain node, and connect your "knob" to that gain's .value to use the gain node to scale the values (remember, there are 1200 cents in an octave, so if you want your knob to represent a six-octave cutoff range, the .value should go between zero and 7200). Then connect() the DCOffsetGain node into the filter's .detune (it sums with, rather than replacing, the connection from the LFO, and also sums with the scheduled values on the AudioParam (remember you'll need to scale the scheduled values in cents, too)). This approach, BTW, makes it easy to flip the envelope polarity too (that VCF ENV switch on the Juno 106) - just invert the values you set in the scheduler.
Hope this helps. I'm a bit jetlagged at the moment, so hopefully this was lucid. :)
来源：https://stackoverflow.com/questions/30019666/web-audio-synthesis-how-to-handle-changing-the-filter-cutoff-during-the-attack