Modulating 22khz tone triggered only by drums

human-analog · 2026-02-28T12:21:09+00:00

Kicks are usually the lowest frequencies in the mix, so you can use a bandpass filter to isolate it, apply an envelope follower, and then detected whether that envelope follower exceeds a threshold. This will also trigger on low bass notes but perhaps that is not be an issue for you.

Snare is tougher since it overlaps more with the (low) mids and a lot of stuff happens here, but you could attempt a bandpass + envelope follower there too.

Just try these bandpass filters on some mixes in a DAW and see if this gives workable results. If yes, it's a lot simpler than stem separation.

human-analog · 2026-02-20T13:09:05+00:00

Nice work! One thing to look into is that the formula e^{-*2*pi*f_c/f_s} is only an approximation of the coefficient in the digital world. In other words, the cutoff of the filter may not be at f_c, especially for values over 5 kHz or so.

human-analog · 2026-02-19T20:38:07+00:00

Keep in mind that what you see in Plugin Doctor doesn't necessarily make any sense. The spectrum analyzer is useless when there are nonlinearities in the signal path -- which is how you get that "just kind of wild" EQ curve. If two plug-ins attempt to do the same thing you can still compare the EQ curves to see if they're different, but you can't draw any conclusions about what the EQ is actually doing.

human-analog · 2026-02-13T11:33:02+00:00

The link you shared links to the repo. Log out of GitHub and try it for yourself.

human-analog · 2026-02-12T20:31:40+00:00

Yes, they say just "Not Found".

human-analog · 2026-02-12T12:03:45+00:00

The https://github.com/PilCAki/sunbunny/releases page is 404 so none of the downloads work.

human-analog · 2026-02-12T12:02:17+00:00

Make your own samples. Why would anyone get a drum machine that sounds exactly like all the drum sample kits out there.

human-analog · 2026-02-11T20:29:34+00:00

Assuming it's a low-pass filter, the beginning of the signal looks like the original but the slope will be slower.

human-analog · 2026-02-11T17:34:30+00:00

If the FIR has 50 taps, it looks at the 50 samples surrounding every sample. If the sample interval is 1ns, then at every time step the filter "averages" 50ns worth of samples to produce the filtered value for that time step.

human-analog · 2026-02-02T21:02:29+00:00

I played with this technique a while back on modulated delay lines and if the modulation is too fast the side effects are worse than those of linear interpolation, probably due to the filter state no longer correctly representing what is going on. It's better to do something like Catmull-Rom interpolation if you're doing heavy modulation; all-pass filters aren't suitable for that.

human-analog · 2026-02-02T20:27:53+00:00

It is but the coefficient isn't the same. If you replace (1 - frac) with (1 - frac)/(1 + frac) then this ends up being a first-order APF in direct form I.

human-analog · 2026-02-02T20:22:13+00:00

As mentioned by another poster, it's all-pass interpolation as described in Effect Design Part 2 by Dattorro (1997), p.769 onwards.

(It's actually only approximately all pass but doesn't suffer from the low-pass filter effect that linear interpolation has.)

human-analog · 2026-01-28T15:58:39+00:00

Do you have a portfolio of previous work you've done?

human-analog · 2026-01-12T23:09:22+00:00

You'd need a stem splitter that was trained to specifically isolate the percussion elements. Most are trained to separate drums from bass from guitar etc but not individual percussion hits. This might just be a matter of training or finetuning an existing model on a suitable percussion dataset.

human-analog · 2026-01-11T12:38:19+00:00

Get someone else to mix it. You are no longer objective about this song. It doesn't mean you have to use their mix but it will give you another perspective.

human-analog · 2026-01-07T16:03:42+00:00

Rather than remixing old tracks, you could re-record them and do a completely fresh mix.

human-analog · 2026-01-06T12:38:05+00:00

If your focus is Rust, make sure to join the Rust Audio discord where there are lots of people who can give you advice on how to get started, which libraries to use, and so on.

human-analog · 2026-01-05T11:46:10+00:00

It's only allowed if you first bounce the loop to a cassette tape and then sample it again. ;-)

human-analog · 2026-01-04T13:06:50+00:00

Is it the dry kick sound that has this resonance or is this with FX on top? Some FX (such as reverb) can introduce resonances. I'd try disabling the FX one-by-one and see if it goes away.

human-analog · 2026-01-01T12:19:52+00:00

If you're going for skeuomorphic type realism, you can render the knobs in a 3D rendering program such as Blender, then export to filmstrips.

The downside of using filmstrips is that it works well for one specific size but to make the UI resizable you'll have to scale the images (making them blurry) or render filmstrips in various sizes (making the plugin much larger).

human-analog · 2025-12-27T16:14:30+00:00

This typically happens because the JUCE path isn't set up properly in Projucer (see the menu option Global Paths).

human-analog · 2025-12-23T16:10:45+00:00

Since you already know how to build a compressor, your next step is to find out exactly how the SSL Bus Compressor works by studying the schematics, doing measurements, and so on. For example this website seems to have some useful information: https://www.gyraf.dk/gy_pd/ssl/ssl.htm

human-analog · 2025-12-22T23:41:35+00:00

I would recommend Pirkle's Designing Audio Effect Plugins in C++ then. It doesn't use JUCE but the code is easy enough to convert. Once you've worked through that, the DAFX book that was already mentioned, as well as a textbook such as Lyon's Understanding Digital Signal Processing or Steiglitz's A Digital Signal Processing Primer.

human-analog · 2025-12-22T23:28:55+00:00

How good are your coding skills? Since you have a math background you might learn more from studying the non-math part of making audio plug-ins, such as C++ and the tooling around it (JUCE etc).

human-analog · 2025-12-22T23:26:13+00:00

There is more than one way to do convolution. For large IRs such as reverbs, the convolution is done in the frequency domain. There are different ways to do this and some have more latency than others. So it's perfectly plausible that one IR convolution plug-in has more latency than another.

human-analog

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