@David-Healey I now managed to install reaper and the compiled plugin now works. so it was those listed bugs

Just realised I wrote "Sinppet" 😆
Will just keep that one so much it fits the issue 😆

@Orvillain ill give that a try thanks. I don't have claude code though. I cant afford paid services at the moment

I'm not sure whether MIDI tempo information messages makes it through to the Faust node, but the tempo sync node is definitely the way to go. just build your Faust node with a absolute delay time parameter and then connect it to the tempo sync node - it automatically sends the correct time value matching the tempo and reacts to tempo changes etc.

upcoming docs:

Tempo Sync (control.tempo_sync)

Converts a musical tempo value to a duration in milliseconds and sends it as a modulation signal.
Tempo Sync converts a musical time value (such as 1/4 note or 1/8 triplet) to a duration in milliseconds based on the current DAW tempo. The output updates whenever the host tempo changes or any parameter is adjusted, making it suitable for driving time-based effects that need to lock to the beat.

The output is an unnormalised modulation signal carrying the raw millisecond value. If the target parameter expects a different unit (such as frequency in Hz), place a control.converter between this node and the target. When Enabled is set to Off, the node outputs the manual UnsyncedTime value instead, allowing a smooth fallback for standalone operation or manual control.

CPU: negligible, polyphonic.

Signal Path

Pseudo-code - hover highlighted terms for details // control.tempo_sync - musical time to milliseconds // BPM + parameters -> ms out (unnormalised) onParameterChange() { if Enabled: output = tempoToMs(bpm, Tempo) * Multiplier else: output = UnsyncedTime } onTempoChange(newBpm) { bpm = newBpm if Enabled: output = tempoToMs(bpm, Tempo) * Multiplier }

@Christoph-Hart that's superb news! I'm so looking forward to bringing my AMS RMX16 clone to dolby atmos to have that gated reverb in my next album. It was a stupid idea to start making an album in atmos anyways but hey... why not

@David-Healey no they need to ring out. i will probably limit it around 16 voice cards or something like a real synth would have

@Morphoice https://www.thewindowsclub.com/uninstall-windows-defender

@Morphoice Sick dude. Well done! Loving the sound and glad you kept that interface clean. I'll trade you a code for my synth if you are interested 👈

@dannytaurus no nothing changed except the plugin version in project settings. the hise version was around half a year newer from the one i last compiled it with on the develop branch

@Morphoice yeee! Let go! 🙏

@dannytaurus that makes sense

@Morphoice The basics are they are using ML models to analyze and replicate the behavior of analog equipment. They run various test signals through them and use the results to train a model that can predict how the device will respond to any input signal.
The Kemper Profiler uses a different approach. They capture the full response of an amp or effect at different settings. It creates a snapshot of the device and builds a profile. This is closer to convolution than ML.
The real problem you're going to run into with time-based dynamics (like a compressor with slow release) is the model will start to emphasize the non-linear behavior. The result of these ML compressors ends up sounding like saturators rather than compressors.
I'm on the same path as @griffinboy where I'm focusing on making models of specific components and inserting them between other analog models.
If you want to start tinkering with Neural models in HISE, here's a thread of my starting process:
https://forum.hise.audio/topic/8701/simple-ml-neural-network/108?_=1748359797127
@griffinboy also posted a thread about some details to the training scripts.
I haven't had time to fully dig in for the last few months, but plan to soon.

@d-healey I tried that recently just to reflect the bypass state of the DAW to the bypass button of the VST, but it wasn't reliable in the end... Maybe I've done it wrong that been said...

@d-healey

Or c++ / snex 😁

But two samplers is the easy way I'm sure.

@Morphoice scroll down in the export dialog, there's the juice.

@griffinboy I'm not at that point yet tbh. Most of what I'm working on right now is audio analysis in realtime and sample playback. I do really want to do a distortion/saturation effect of my own at some point though!