@clevername27 said in Best Way to Implement This?:

@VirtualVirgin I appreciate the enthusiasm. When I do take the wraps off, I'm following @d-healey's lead, and releasing the rest of my code to the community. Let me know if there's something specific you'd like.

Thank you! When I have enough experience, I would like to contribute as well.

Free mankini with every commercial license???

@d-healey Hi Dave, thanks for answering. Yes, I have. I don't remember changing anything in my code…but I must have changed something?

@clevername27 said in All About Latency:

Is the idea that if you reported your latency correctly, the scheduler would be sample-accurate? If so, given that it's impossible to measure a plugin's latency, how do we know if the scheduler works?

Yes. While playing back audio, a properly functioning DAW will take into account PDC for each track and then play back first the track with the most delay, then play others so that they're all matched up. In Reaper, you can view PDC for each track in the Performance Meter window. You can even see it change based on the HISE's method call to report the latency.

To check that it works, have a plugin that simply delays the audio for a given value. Create a knob, and add a callback that reports the latency based on the current knob value x current sampling rate, then set the delay. You can use the stock delay node in ScriptNode.

Now create a track, add an audio file, duplicate the track and put the plugin on one of the tracks and invert the phase on it. Now hit play and adjust your knob for the delay. No matter what you set it to, you should be getting a silence (except for a few moments while turning the knob during playback where there's a mismatch between the reported delay being adjusted by the DAW and the delay node reacting instantly).

@clevername27 said in All About Latency:

I mean that a if plugin is performing a non-constant-time operation (O1 complexity), then the amount of time the DSP code takes to execute may vary, meaning the audio output may gets written later in the buffer (or the next buffer)?

Aren't all realtime audio operations like that? That's what the buffer is for. Even if you're overflowing the buffer, you'll get audio breaking up but an offline render will still render the audio correctly in sync.

@Orvillain @d-healey Thank you, again (and @d-healey). All set.

Oddly enough, with this envelope, 1ms on the Attack is actually 0ms on the attack. And…you can't set it to zero.

065dd025-a2fb-49ef-8269-183c425f3250-Screenshot 2025-03-22 at 7.55.57 pm.png Screenshot 2025-03-22 at 7.55.57 pm

@clevername27 No problem! 🙏

@Christoph-Hart That's the one. Maybe I'm having a GIT issue? Thank you. Yes, GIT issue. All good.

@d-healey You MADMAN—five Monkeys awarded to thee! 🐒🐒🐒🐒🐒

Like Sir David wrote, you need to resize a panel—any panel—and the whole interface will scale, accordingly. Here's code you can cut and paste:

Link Preview Image [Feature Request] GUI Scaling with window handle?

@Christoph-Hart Do I need to parent the ZoomPanel ? If I make a child of ZoomPanel knob/Slider etc cursor are also being changed to doubleArrow Cursor

favicon

Forum (forum.hise.audio)

Important: Only the compiled version will resize—and if you think about it, that makes sense, because you can already resize within the HISE IDE.

@clevername27 Looks nice!

@d-healey Restored from Time Machine. I suspect the problem is the Faust libraries losing their Gatekeeper clearance. (I tried to upvote your response, but I've already upvoted you six times today. 😂)

@Christoph-Hart Appreciated - the shader isn't behind the panel—that's just a static image. (See the example snippet.) The shader is in the header.

@Christoph-Hart Thank you very much,.

@Christoph-Hart said in Can't Build New Develop Branch with Profiling (macOS):

@clevername27 FIY, that's fixed with today's commit 6bc83f40afce511cc97efe5390ead05538eb3bb6

Thank you.

@Christoph-Hart cheers, mate.

@griffinboy You can tear open the units - I can get you any support you need from the companies. It's just a question of if you can do it. If you think you can, DM me. Or…DM me anyway, it's all good. As you're a student, I'm impressed with the work you're doing - keep it up!

@d-healey @Lindon

Is that always the case? I vaguely remember documenting a situation where threads were executing (twice) before onInit. Something to do with default presets

I haven't used the default preset thing so that might change things but the usual behaviour is as I described.

Wisely said; here's the thread: https://forum.hise.audio/topic/10723/preset-component-callbacks-triggering-twice-on-startup/11?_=1742336098253

Is that always the case about Callbacks?

I will investigate - it will take me a couple weeks, but I will get back to you.

Do you mean manually inserting a conditional check
Yes, in every place you don't want it to execute automatically.

I appreciate your thoughts. For me, at least, that solution isn't tenable. I have far too many callbacks, and it has emerged, on my end, that I am unable to always tell if a callback is being called at startup—or even what means, when you get into the various preset handling systems. But overall (and this is a @Christoph-Hart thing, not you), to me it is an illogical and limiting design choice. I get why it's there, and I'm free to write my own HISE. I just think it's wrong, much like with the broadcasters. 👻

How do you pass data between (for example) UI and real-time threads.
The same way I'd pass information between my UI and an effect - or any other HISE module. Either using processor/parameter ID or setAttribute.

I thought about this, a lot. Because generally, I agree with you (and I teach my students as such) that global variables are to be eschewed (and because your code is so elegant). But there are legitimate usage cases for them. And more to the point—if the language says it supports global variables, then it should do that. At one point, @Christoph-Hart improved the handling of globals by introducing the Global object…but…and please let me know if I have this wrong…didn't fix them so they work properly.

Out-of-order execution during startup.

I have looked high and low for the screen-capture movies I made, but I don't think I was able to upload them because they were over 10MB, so I deleted them. I do remember that another user appeared to be having the same issue. I simply had to spend a couple weeks completely refactoring my code to get around that issue. 😭

Anyway, thank you, as always, for giving thought to my questions.

@d-healey Thank you, Dave! I uploaded the 48kHz file by mistake, but as usual, your questions took me down the right paths to fix the problem.

To anyone arriving here having had the same issue:

When I opened the original sample file to check if it was the correct sample rate, I decided to re-export all the samples for good measure, and recreate the Samplemap from scratch. (Not a big deal; there were five samples in that map.) And now it seems fine. That was a critical issue for me — thanks again @d-healey.

@d-healey I didn't mean to imply it was your code—it was mine, in that I had assigned an LAF to a Component that no longer existed. And thank you, as always.