[ANN] LSP Plugins 1.2.15 released!

[sadko4u]

• Added experimental VST3 plugin format support.
• Added built-in presets for Compressor plugin series.
• Fixed bug in smoothing mode of Parametric Equalizer and Filter plugins that could be caused when using large buffer sizes. Additional optimizations to reduce CPU resources when processing in smoothing mode.
• Updated behaviour of band frequency splits in the UI for Multiband Compressor, Multiband Expander, Multiband Gate Multiband Dynamic Processor and Crossover plugin series.
• Several bugfixes and improvements in implementations of VST2, LV2 and CLAP formats.
• Added possibility to manually enter channel name in Mixer plugin series.
• Added MIDI velocity control code display in Sampler and Multisampler plugin series.
• Added support of LR2 (12 dB/oct) filters by the Crossover plugins series.
• Added S/M Apply switch to Crossover plugin series that applies effect of Solo/Mute buttons to corresponding frequency band's outputs.
• Improved grid color and visibility for the Spectrum analyzer plugin series in Spectralizer mode.
• Implemented Hold option for Compressor, Dynamics Processor, Expander and Gate plugin series.
• Implemented Hold option for Multiband Compressor, Dynamics Processor, Expander and Gate plugin series.
• Added Dry/Wet balance control for Compressor, Dynamics Processor, Expander, Gate and Trigger plugin series.
• Added Dry/Wet balance control for Multiband Compressor, Dynamics Processor, Expander, Gate and GOTT Compressor plugin series.
• Fixed phasing issue between dry and wet signal for GOTT Compressor in Linear Phase mode.
• Fixed invalid curve markers display in Multiband Gate plugin series.
• Re-implemented allpass filters for Equalizer that could yield to high frequency pitch buildup.
• Updated split editing behaviour for Multiband plugin series and Crossover plugin series.
• Added basic UI design guidelines to the documentation.
• Several bugfixes in graphic toolkit library.

https://github.com/lsp-plugins/lsp-plug ... tag/1.2.15

[sunrat]

Upgraded. Got the PEQ x8 JACK version running as room correction EQ. All good!
Thanks again for your great work. Wish I could afford much more for my Patreon subscription but with everything else getting rapidly more expensive I'm having to ration things. Will keep those few measly bucks coming your way though; hopefully more people can do the same.

[novalix]

Kudos!

I have a specific question about the implementation of the allpass filter in the peq.

In this video where Eric Valentine talks about the design of an hardware eq, i've learned about the particular feature of allpass filters to frequency dependent alter the phase response.
He starts to talk about it at roughly 15:35 and then again at 30:20 where he shows a use case of this characteristic.

Was this a use case you had in mind when you implemented this type of filter?

[sadko4u]

All-pass filters are the primary filters used by crossovers.
Here's the overall discussion about the phase problems with Calf plugins which helped to solve phase problems for LSP crossovers:
https://github.com/calf-studio-gear/calf/issues/217

The simpliest design principle of allpass filter is just to take low-pass filter, hi-pass filter and sum their characteristics.

For example, if we have low-pass filter and hi-pass filter:
HL[p] = 1/(1+ap + bp²)
HH[p] = (p²)/(1+ap + bp²)

The desired all-pass filter:
HA[p] = HL[p] + HH[p] = 1/(1+ap + bp²) + (p²)/(1+ap + bp²) = (1 + p²) / (1+ap + bp²)

But sometimes we need to understand what's the phase difference between HL and HH. If it is multiple of 360 degrees, then we can just sumarize both characteristics. Otherwise we need to subtract them. So for 2-pole filters the above filter actually is a Notch filter. But if we alter the sign of hi-pass filter, then it becomes the all-pass filter:
HA[p] = HL[p] - HH[p] = 1/(1+ap + bp²) - (p²)/(1+ap + bp²) = (1 - p²) / (1+ap + bp²)

Having the transfer characteristics of the analog filter, there is no problem to design the corresponding digital filter. Consider looking the Bilinear Transformation of filter cascades.

[novalix]

Thanks for your explanation.

Maths is hard

So, do i understand that correctly that the implementation in Parametric EQ is a 2-pole filter where the respective phase shifts cancel each other out and transform to an allpass filter effectively creating a crossover with minimum phase and a flat frequency response?
Whereas the Undertone Audio EQ in the respective settings produces a one-pole allpass filter which creates a flat frequency response with a phase shift, which in turn can be used as a side effect to phase align certain multi miced sources.

That would be:
To create a similar effect as in the bass example in the video i would put a crossover with one point at fourhundred-something on one of the sources, route the outputs to different busses and switch phase on one of them.

[sadko4u]

So, do i understand that correctly that the implementation in Parametric EQ is a 2-pole filter where the respective phase shifts cancel each other out and transform to an allpass filter effectively creating a crossover with minimum phase and a flat frequency response?

Not clear what do you mean. Each analog filter can be defined by the transfer function:

h[p] = (T0 + T1p + T2p² + ... + Tnpⁿ)/(B0 + B1p + B2p² + ... + Bnpⁿ)

The roots of the top polynom define so-called 'zeros' - real or complex-conjugated vectors on the complex number plane that define the frequency at which the magnitude starts constant +6dB/octave amplification.
The roots of the bottom polynom define so-called 'poles' - real or complex-conjugated vectors on the complex number that define the frequency at which the magnitude starts constant -6dB/octave attenuation.
The conclusion is that number of poles should be not less than number of zeros. Otherwise we'll get infinite filter amplification which is not possible.
The trick of the allpass filter is, if we define the pole at some frequency, we define the zero at the same frequency. This gives 0 db/octave attenuation because the zero compensates the pole. But because the pole and the zero are located at different halves of the complex plane (this means that pole has negative real part and zero has positive real part in complex value) we introduce the phase shift.

Whereas the Undertone Audio EQ in the respective settings produces a one-pole allpass filter which creates a flat frequency response with a phase shift, which in turn can be used as a side effect to phase align certain multi miced sources.

I discarded to use one-pole allpass because there is no crossover filter that can utilize it (all crosssover filters are at least LR2 or second-order).
Second, the implementation of one-pole allpass filter made it unstable:
https://github.com/lsp-plugins/lsp-plugins/issues/358

But the slope of the allpass filter depends on the slope. And each slope step adds two more poles and zeros to the filter which makes it more agressively shift the phase while keeping the magnitude being constant.

That would be:
To create a similar effect as in the bass example in the video i would put a crossover with one point at fourhundred-something on one of the sources, route the outputs to different busses and switch phase on one of them.

Or just place an allpass filter from the EQ at the same frequency.

[GMaq]

Congrats @sadko4u!

These are on their way to the 'MX Test' Repo for AVL-MXe users, should be up later today..

[sjzstudio]

Remember to consider donating if you feel satisfied with these tools. Even for the price of a cup of coffee, you can bring joy to developers. If everyone offers coffee, it makes many packages a year.

[Jodax]

Glad I ran across this notice.

Next came the realization that the Gentoo stable repo is a bit outdated (v 1.2.3-r1).

Configured my local over-lay and now I am up to date.

Those presets are a very nice addition. You are doing good work.