@adrianmcroft Silly me - cable_expr

@HISEnberg haha... yeah. I am always ready to learn new things.

@aaronventure said in global send / receive:

@d-healey I vaguely remember reading somewhere about it. It should be the same whether the network gets converted to C++ on project export or on its own first, no? The only difference is some nodes, like SNEX and Faust, require in-HISE compilation first, and if you have a really hungry network, then you can get the CPU usage down by compiling it.

Though it's probably not a bad idea to get @Christoph-Hart to confirm.

@Christoph-Hart What say ye?

@JeanC You mean like a VST? Then no.

@Morphoice

Doesn't even need to be used once. There are approximations or lookup tables that can be used.

@Christoph-Hart said in ScriptNode - switching between myultiple modulation signals...:

@Lindon https://docs.hise.dev/scriptnode/list/container/branch.html

That‘s made precisely for this use case.

Happy to be of service... :-)

@Lindon said in Emulating vintage DACs with bitcrusher & μ-law compression:

@Morphoice yeah I tried 512 but no difference.

Since the mu law is a weighted repartition of a signal, the ratio oscillate between linear at mu=0 toward logarithmic for higher values. So the higher mu the less impact on higher signals. All that to say the difference lies in the tiny signals like reverb trails, etc... So you can keep crushing higher signals like falling fire from hell on a nursery while enjoying delicate small signals with a warm latte

@HISEnberg Yeah its coming back to me now, I got it to work after leaving out tempo sync abilities... I'll do some more testing myself, but it seems to be a bug

@Lindon @Morphoice and the control nodes, in particular the bang node

@Christoph-Hart definitely something like that, see my second post where you can effectively trick it into working -- the node detects the channel count of the network on load, and doesn't react if the network channel count changes.

@Christoph-Hart thanks

@ccbl your welcome.

@southfieldsound you're right about 32 bit and the test I have made confirm there's still not zero bits after -1000dB, and even batman can't hear that...
But superman could! After all, each -3dB just divide by 2 the previous factor, so you never reach 0 until the last bit can computationally handle it...

Now to be strict, the math node does the job of nulling to death, so yeah, if you need pure mathematical silence, use this instead.
Just pair it with a smoothed_parameter node to prevent clicks and you're against any upcoming super hero!

@d-healey Thank you, yes the comparison will work. For some reason, getAttribute() works with the string I pass, but not getAttributeIndex(). In any case, I've got all my answers, and thank you as always.

@aaronventure said in How we can display the gain reduction value from the ScriptNode ( updown_comp ):

well......
yes the lower panel seems to be quieter, and assuming this is the "after" panel we are pretty much back where we started - the inflator is an expander - so it should be higher not lower....

Meanwhile the skew seems to be connected to some non-existant node called unscaller, and removing that doenst seem to allow it to connect to either of the pow nodes...they seem to be from some other network:

38e9869f-fb53-4d21-afc1-6143915a2b87-image.png
so I cant get anything to connect to the pow nodes, and still the text display = 1.0...

As I say I think we should just call it quits here.

Actually I got the pow to work - bu tthe advantage over just displaying the original pre and post peaks escapes me... hey ho, lets all move on.. thanks.

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7AxIRKRLzWRbPBfiXsyetaiBvwaGEkzCu3JfxFurf1ec853c8zZBROazfQ9DRBTtmapi0j+8GkeG0EZcVbc0ekH.N.7dVYFDfRzIVC4vp0vdEbsf+YxJ4LYXf+Ih0IgLd.nF5BRNxRJEQBeHlpgA7nOmlfVXvCH2ZiWyTgTf2rrTzo9qtmMJ.iowQUyNE7ZCBDkmFz2mcnD7Fqg9iSOn7VxhKkCnlocSrbtik0j6Zux8kSQtGp9BiKM3cAy8JqrBLm8CIRYYLz8GIY.MBu293CGZKYQm6LAfKSkeGxjuwIyAbt9.qsm7p.l6MAlLo6jfNNJSYy00ozcZ0r0FNvPeiCLXqlq9.76qVE.q8vyiN0kBVlHoDOkY.zppoczl1QaZAeJgQBA56qgSCh15CFQbmlvIB3o6tdKzio2He+bn.GsSOpizN.AhH7TLdBrOqOCuQ7YwfDASJXfIMHk3qFpfN.CSih7O4P4iRDTdZ4Qyx.NeQE98XrVY3a.kCAisVn2Ognupvi37ZQWeXc0Ev.uOvPzUdiW1fjsbq7PCi+e.GyPZM

There are some helpful notes in the source code:

/* REGARDING THE MAX PEAK: This method assumes that the only important * sample in a look-ahead buffer would be the highest peak. As such, * instead of storing all samples in a look-ahead buffer, it only stores * the max peak, and compares all incoming samples to that one. * The max peak has a hold time equal to what the look-ahead buffer * would have been, which is tracked by a timer (counter). When this * timer expires, the sample would have exited from the buffer. Therefore, * a new sample must be assigned to the max peak. We assume that the next * highest sample in our theoretical buffer is the current input sample. * In reality, we know this is probably NOT the case, and that there has * been another sample, slightly lower than the one before it, that has * passed the input. If we do not account for this possibility, our gain * reduction could be insufficient, resulting in an "over" at the output. * To remedy this, we simply apply a suitably long release stage in the * envelope follower. */ /* REGARDING THE ATTACK: This limiter achieves "look-ahead" detection * by allowing the envelope follower to attack the max peak, which is * held for the duration of the attack phase -- unless a new, higher * peak is detected. The output signal is buffered so that the gain * reduction is applied in advance of the "offending" sample. */ /* NOTE: a DC offset is not necessary for the envelope follower, * as neither the max peak nor envelope should fall below the * threshold (which is assumed to be around 1.0 linear). */ /* REGARDING THE GAIN REDUCTION: Due to the logarithmic nature * of the attack phase, the sidechain will never achieve "full" * attack. (Actually, it is only guaranteed to achieve 99% of * the input value over the given time constant.) As such, the * limiter cannot achieve "brick-wall" limiting. There are 2 * workarounds: * * 1) Set the threshold slightly lower than the desired threshold. * i.e. 0.0dB -> -0.1dB or even -0.5dB * * 2) Clip the output at the threshold, as such: * * if ( in1 > thresh_ ) in1 = thresh_; * else if ( in1 < -thresh_ ) in1 = -thresh_; * * if ( in2 > thresh_ ) in2 = thresh_; * else if ( in2 < -thresh_ ) in2 = -thresh_; * * (... or replace with your favorite branchless clipper ...) */ }

@Christoph-Hart I had a feeling but I did have to ask 😆

@Christoph-Hart

I think there might be a glitch here somewhere. Another one of my projects has suddenly stopped working with my polyphonic filters. I'll do some investigations

@HISEnberg Ah, there was a random parameter control with no name on it, that fixed it, cheers!