The news are:
- About SlickEQ GE and WYSIWYG curve and frequency analyzer
- About SlickEQ and new “Funky” output saturation mode
- About Limiter6 and the skin
- About oversampled EQs
- About Nova67P
Now let’s go into the details.
Read more of this post
This video features ReaEQ, SlickEQ, Pro-Q2 and IGVI. Video by Dan Worrall.
Okay, this is my first try to make some kind of classification of digital equalizers. It’s mostly based on some defects or features they have in their responses. These defects or features give digital equalizers their unique sound. Sometimes they sound “digital” in bad meaning of this word (i.e. “harsh”) but the other side of digital sound is clean, pristine and maybe too cold sometimes. I don’t have much time to make this classification really glossy so it’s some kind of a draft.
In my opinion there’re 7 main properties of digital equalizers, which affect their sound:
- Frequency response behavior near Nyquist frequency
- Phase response behavior near Nyquist frequency
- Frequency response ringing
- Types of Curves
- Time domain response
Now I’m going to try to illustrate possible cases for each property by some images mostly created by VST Plugin Analyzer. All pictures were created at 44.1 kHz sample rate. Also I do not mention real plugin names used.
1. Frequency Response Behavior Near Nyquist Frequency
Check this picture. There’s a typical 1 kHz peaking curve. Also I shifted this curve using copy and paste in an image editor to show how it’d look for 10 kHz.
Now if someone has implemented cool EQ using equations from “Cookbook formulae for audio EQ biquad filter coefficients” by Robert Bristow-Johnson (RBJ) the curve at 10 kHz could have this kind of a look:
So you can see the right side of the bell shape is distorted. Well it sounds like high Q harsh boost. There’re ways to mathematically match Q at 10 kHz with Q at 1 kHz but the shape distortion remains and such distortion has its own “digital” sound.
Read more of this post
BPB’s Top 40 Freeware VST Plugins Of 2014!
A couple of words about top 3 effects.
I had pretty interesting concept of a mixture of parallel equalization and splitband compression in one plugin but the approach was just of pure interest without any realistic vision of final plugin. The plugin was created for KVR Audio Developer Challenge 2014 in a great hurry and under very stressful conditions. Some parts were created in less than a hour and the most complex part of this project was to bring all components together and quickly create some kind of acceptable GUI. So we have what we have. The sound of the plugin is good, the plugin is really helpful in many complex mixing/mastering situations but GUI usability is far from ideal. That’s why we’re going to re-release this plugin with enhanced workflow and with some new features under Tokyo Dawn Labs. Also I’m very happy with my 2nd place on KVR DC14 too! And also thanks to Lesha for the current skin.
#2. TDR Kotelnikov
I can’t say, I did a lot for this plugin. In my opinion the most remarkable contribution from me was complete removal of feedback path (lol) and maybe a little help to RMS part (which led to high CPU usage we finally have). But in GE version I had a great pleasure to implement FDR, variable slope HPF and equal loudness stuff. Also our conversation with Fabien about the name of this plugin was very fun!
It was really joyful experience to collaborate with Herbert on this project and to program DSP code for it and also to add such nice things as EQ saturation, Soviet mode and auto gain stuff. For GE version I’m still not 100% sure (about 75% actually) for the decision to add tilt filter instead of the 4th band but finally we have unique linear shape tilt, great workflow I’m happy with and also with such small additions as Japanese mode, a couple more saturation types and new HPF/LPF stuff! I like standard freeware version due its extreme simplicity and I use it a lot (although I have GE version too). The CPU usage is higher than desired but we tried to reduce it in recent 1.1.0 update.
See you next year!
Such thing intended to reduce IMD distortions when working on high sample rates. Should I insert it in the beginning of audio processing chain or just before output limiter or after it or between each plugin? I don’t know. I don’t work with high-res. But I like to listen high-res music through it. The depth is greatly increased and mid-range becomes pristine clean. I use this plugin with George Yohng VST Wrapper for Foobar player.
This is the topic about the theory behind this plugin:
And these are download links (32/64-bit, VST, AU, AAX):
[Win (installer)] [Win (no installer)] [Mac bundle]
(they’re also specified in post #173: https://www.gearslutz.com/board/10641174-post173.html)
The most impressive commit is from Niklas (post #203):
I experimented with the Ultrasonic filter on a just-for-fun master at 96kHz. I had about 10 plugins in a chain (all producing harmonic distortion of various degrees, mainly Acustica Audio Aqua and Nebula instances) and smashed into Voxengo Elephant ridiculously hot (limiting up to 8dB at best). I inserted Ultrasonic filter first in the chain and then after (and thus before) each and every plugin, including after elephant (so actually after the limiting.. just before final dither). The difference is ASTONISHING!! It is so ridiculously obvious and much better with the filters than without. A truly mind-blowing “WTF???!!” experience for me. It basically removed a lot of problems I’ve identified in mixes and previous masters that had heavy processing with a lot of harmonics.. mainly that nasty build up of “ringing” around 2.3 to 4kHz area. It also cleared out the congested mids and lower mids.
This is a game changer for me. I will not do any more high resolution mastering without this filtering. Period. Heck, I will never mix at high resolution again without this plugin inserted all over the place.
- Added compressor bypass parameter
- Default preset now has stereo mode setting
- Fixed a couple of possible crashes due to race condition bugs
- New skin by Lesha
Get updated version from <Nova-67P page>
BUG REPORT BUG REPORT. In v1.0.3 auto-gain indication (horizontal black line) doesn’t work. I’ll fix it.
Check also tips & tricks by Herbert <http://varietyofsound.wordpress.com/2014/07/07/tipstricks-with-slickeq/>
Part 1. SlickEQ free version
1.1. Listen to your favorite music through SlickEQ
SlickEQ is good enough to be used as hi-fi equalizer. On Windows you can use <George Yohng’s VST Wrapper for Foobar2000 player> to listen to your favorite music through it!
- Set output level to -3 dB to prevent clipping
- Set “calibrate” to 0 dB to leave harmonics added at the edge of hearing threshold
- Don’t over-EQ the music. Keep gain values below 3 dB.
1.2. Save your CPU cycles when mixing!
SlickEQ is a bit CPU heavy. We performed optimizations for Sandy Bridge architecture but for modern energy saver simplified CPUs in thin notebooks or for old CPUs the CPU load may be too high.
- Turn EQ Sat off and set Out Stage to Linear. If non-linear processing is not used, SlickEQ may consume 3 times less CPU cycles! If you already have some saturation on a track you may choose to turn off the saturation in SlickEQ.
- Turn stereo mode to “Mono” if you’re processing a mono audio signal. In this case SlickEQ will consume 2 times less CPU cycles! NOTE: some DAWs report that they have a mono track and in this case stereo mode button is grayed and SlickEQ operates in mono mode automatically.
Read more of this post