All Waves Plugins Explained
Sanjuana Gautam
I guess Waves is looking for other ways to earn money than to keep customers loyal to Waves, which is all the proof I need to know the sound quality I can expect from Waves and the kind of support I would get if needed.
Of course I am not spending a cent on Waves plugins or subscriptions ever again but I paid others and I am thankful I had this problem with Waves to have the opportunity to realize how much better are other brands plugins.
It was not advertised that I would need to install Rosetta for installing the plugins. It was advertised that Waves v14 were 100% fully native and never explained that I would still need to install Rosetta.
Rosetta is an Apple technology and is installed from Apple servers after MacOS rejects running Waves Central on Apple Silicon. A dialog pops up and asks you if you want to install Rosetta and and if you do, you are required to enter your administrator password. It is not installed stealthy nor is it installed by Waves.
Again. This is explained in the tech specs which you should be looking at before purchasing to make sure that plugin will work on your system and with your DAW. You also have the option to install the demo before you purchase to make sure it meets your expectations.
When you purchase a plugin, right above the confirm button, is All Sales Are Final. Nothing about this is unfair. There were multiple steps in the process where you could have decided not to purchase these plugins.
I did read the tech specs as I do my job. I also tried the demo in another system, but as Rosetta was launched stealthy and silent in that one computer, I never realized Waves v14 still requires Rosetta being installed.
Aliasing and intermodulation distortion can happen in any non-linear analog circuit or algorithm. In the analog domain, there is no hard limit to the frequency spectrum, only the limitations of the components or the intentional design of the circuit, for example a brickwall low pass filter (that can introduce ring modulation) or a limiter circuit.
Any schmuck can use the stock compressors, eqs and limiters on their daw (and frankly, if they know what they are doing, get spectacular results), but the really cool mix engineers who consider themselves artists before engineers know how to apply saturation to give a recording warmth, glue and character. These people may or may not have beards, are trying to quit smoking (again), wear a variety of interesting hats, drink artisanal whiskey and sometimes wear slightly ironic clothing items that only they understand. In other words, these are people I can relate to who are at least trying to contribute something interesting to society.
In the illustration below, the first bar labeled 1 would be the frequency you recorded. The harmonics are the higher order frequencies 2-8 generated by some non-linear formula. Complex saturation (like hard clipping or a synth saw wave) can generate harmonics on low frequency information all the way through the sonic spectrum, and even subtle saturation algorithms with just a few even harmonics (a gentle tube saturator) add harmonics that exceed the Nyquist limit the upper frequencies.
After 10 seconds, the signal contains frequencies higher than 22050 (the Nyquist frequency), and we see the corresponding line in the spectrogram reflect/bounce/fold across the Nyquist value, and we hear the decreasing pitches that result.
Between 10 and 24 seconds, more and more of the waveforms reach the Nyquist frequency. At about 24 seconds, all component sine waves have reached the limit, and so we should not hear anything at all (all frequencies are beyond the upper frequency limit of our hearing) and yet we do hear the aliased signal.
DAWs now operate by default at 24bit resolution or greater, this offers a massive 144dB of dynamic range to safely capture audio without clipping transients (hint hint: gain staging). Exporting 24bit audio to CD or MP3 yields 16-bit resolution at 96 dB of dynamic range. Dither noise is added intentionally during the conversion process, usually between -80 and -60 dBFS, which is effectively beyond hearing perception vs the program material, and certainly quieter than noise added by many analog devices like tape.
Since music is an extremely complex set of every changing frequencies, all those accumulating aliasing tones compound and slather across the frequency spectrum, creating all kinds of low-level non-musical noise that starts to mask the clarity of the sound.
But if your computer, audio converter and software could support it, and you were to remove this aliasing by increasing your sample rate to 192k, I can almost guarantee you would notice a difference in clarity and definition of the entire mix.
So, there genuinely was quite an advantage for pro studios (or all studios) in the early digital years, before ITB mixing and plugins began to dominate the mix business, where your digital system was only used as a capture and playback device, and all sound manipulation, including the non-linear processing was done using mixing desks and outboard gear. Even if the AD converters were not all that great, or running at 16/44.1, it sounded damn good if the outboard gear was good.
A plugin that generates saturation can and should internally upconvert the sample rate, process the signal, then down convert with a steep filter applied to carve out the ultrasonic information. All downsampling does this using filters and dithering and is what every DAW does when you export a high sample rate project to 44.1 CD or Mp3.
More recent research has led to antiderivative anti-aliasing (abbreviated as ADAA). This was first introduced in a 2016 DAFx paper by engineers from Native Instruments. The basic idea is to apply anti-derivatives of a given function to suppress aliasing harmonics reducing the need for intensive oversampling or eliminating aliasing altogether. This is extremely attractive to manufacturers of DSP devices, as it means more features at lower cost.
There are many bobby buzzkills on various forums that did a bunch of tests like this on a huge range of popular plugins, but I noticed that one of the more annoying threads was based on tests where the gain staging was completely jacked and they were pushing them to extreme settings. Clipping the input of a plugin generates harmonic distortion exceeding any aliasing compensation, and would never happen under normal operational conditions, so I took great care to pay attention to the gain staging and any special levels that plugins are calibrated to.
Wow. To say this is a sleeper hit would be an understatement. Voxengo may not make sexy GUI plugins, but their engineering has always been reputable and first rate. This freeware gem has been around for many years, and I admittedly overlooked it. It is one of the best options available for tube saturation and distortion on high harmonics thanks to generous oversampling options at 2x,4X and even 8x. At gentle default saturation settings, no aliasing occurs in a 48k project up to 22khz.
Despite being free, the IVGI2 is widely regarded as one of the best saturation options at any price point. The plugin has a wonderfully subtle and musical tone and 4x oversampling is automatically applied under the hood. This is an excellent plugin to apply at low settings (DRIVE between 1-5) to many channels for vintage warmth and character.
Even with the oversampling overhead, CPU usage is reasonable. At higher drive settings, negligible aliasing occurs at very low db levels. It would be rare for heavy distortion to be applied across many channels, but buildup could be possible in such cases as smashing every channel in a multitracked drum kit.
While I would love to review every AO plugin, the BritPre73 is one of the more minimal saturation options and I did quick tests of many other AO plugs and the results were similar, but worth noting that some of the channel strips with multiple saturation stages are more prone to compounding aliasing issues and very high CPU use.
My reliable but aging i7 would be limited to about 12 instances of this in oversampling mode before the CPU would start to top out, and stacking other non-linear plugs from AO will only exacerbate the issue.
Bootsy (Vlad) is a musician and software engineer who just happened to build a bunch of incredible plugins, give them away for free and then got a new day job (in software engineering) that kept him too busy to update all his old 32bit classics to 64bit architecture.
The result is very modest CPU usage and aliasing that is effectively inaudible. There is the chance that placing this plug on every channel might cause build up, but I only ever used it for subtle modern tape saturation and compression, usually on drums.
This was my favorite plugin for over a decade, and I searched far and wide for a 64bit replacement, but none of them did quite what this does for sheen and very expensive sounding saturation on a specific frequency range.
I love the forward looking design to this channel strip (that Cubase apparently has followed), and the other modules are really solid, but wow, there was obviously no attempt whatsoever to deal with aliasing beyond a manual hipass and mix slider.
This is a simple and beautiful compressor that use extensively for smooth leveling duties both ahead and behind a faster 1176-style comp on vocals and many other sources. It adds subtle warmth and glue without aliasing and it just sounds great on everything.
Variable-mu design with lots of vibe, capable of smooth leveling but also heavy pumping effects. With its two gain stages and interstage transformer simulation it offers a very deep and lively soundstage.
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