Acustica Audio MAGENTA4 Review

I reviewed AA’s newly revamped Magenta version 4 in an RHR video:

Here are the script notes…

Magenta and I go way back. It was the first Acustica Audio plugin that I ever tried and reviewed on this channel.

Magenta4 is unofficially based on Manley Labs hardware. Known as the king of the mid-range.


B – Combines Optical and Vari-Mu compressors.

C1 – Vari-mu, compress mode. Ratio starts at 1.5 then varies dynamically in response to signal level.  Not suitable on sources with fast transients. Great on vocals and mix busses!

C2 – Vari-mu Limit mode. Starts with a higher ratio and goes up to 20:1.

Jerry Tubb’s sweet spot (source ):

Look to the Medium Recovery setting, Slow Attack, Input/Output set for Unity, Threshold set to yield less than a dB of Gain Reduction, then open up the Input a little to bring the level up a few dB… up to a dB or so of GR. Work the Threshold a bit to control the amount of HF transients you let through and… voila’ you’ve got it! (or at least a good start)

D1 – Optical limiter/compressor. Langevin design. Possibly the compressor section of the Manley Core. Great on vocals.

Good on bass and acoustic guitar. Avoid using on drums. Very transparent.

D2 – FET Compressor. SLAM! Emulation or possibly the Limiter section of the Manley Core channel strip. Use this one on drums.

Compressor attack/release times (put on screen)

MODEL B: 12.6mS, 17.6mS, 34mS, 58.5mS, 59mS

MODEL C1: range from 25mS to 70mS

MODEL C2: 0.25mS, 0.85mS, 2.5mS, 2.8mS, 2.9mS

MODEL D1: fixed

MODEL D2: 0.15mS, 21mS, 55mS

  • RECOVERY: sets the compressor’s release time.

MODEL B: 0.07S, 0.347S, 1.281S, 2.4S, 3.2S

MODEL C1: 0.266s, 0.5S, 1.2S, 2.2S, 4.56S

MODEL C2: 0.8S, 1.65S, 3.41S, 6.62S, 13.6S

MODEL D1: fixed – 0.4S

MODEL D2: 0.005S, 0.07S, 0.08S, 0.15S, 0.2S, 0.27S, 0.4S, 0.7S, 1.1S, 2.3S, 4.2S


Wet/Dry knob

SHMOD: Changes the attack envelope’s shape.

Filter = Internal sidechain, high pass.

SC = External sidechain

Equalizers – Two of them.

They work in a series fashion, as opposed to the parallel processing of the hardware.

They got rid of the boost/cut switches. That is a great thing!

Q control – Controls the bandwidth. Widest Q is at the fully counter-clockwise setting. At the narrowest Q setting, the full 20 dB of gain is allowed.

CL – Links the left/right channel controls

A – Enables EQ A

B – EQ B. Not available in mono version. No Q or shelf settings.

Similar to Pultec equalizers. I believe it is based on the EQ section of the VoxBox

Yellow filter numbers are the hardware sampling.


A1 Massive Passive Standard
A2 Massive Passive Mastering version

B – Maybe the Manley Core, Mic input

C – Variable Mu, compressor mode, Line input

D – Mono Pre, Manley SLAM! Mic input

D Mic – Stereo Microphone Input SLAM

D FLT – Stereo, 100 Hz HP filter on.

Waves Abbey Road Chambers Review

I review the new reverb/tape delay/EQ plugin from Waves in this new video:

Script notes are as follows:

Waves Abbey Road Echo Chambers Review

Not just the echo chamber is emulated. In addition to the room it also models a tape machine, tape delay, feedback loop, speakers and microphones.

The chambers use impulse responses and the rest is algorithms.

The STEED process. (show STEED on screen, vertically) Combined tape delay techniques along with the echo chamber and a feedback loop.

Plugin GUI is setup different from the signal flow. The tape delay is before the chamber.

Set your input level so that there is headroom.

Microphone: Neumann KM53 – Slightly bright. The manual says

Schoeps MKH-2s but I could only find information on a MK 2S model. Either way, it is characteristically flat and a more modern microphone.

Position: Click and choose which mic position you like.

Chamber: Classic = Studio 2 chamber, half tiled.

Mirror = Bright/reflective

Stone: Dark and small.

Time X – Controls the reverb tail duration. 0.5 is 50% and 1.5 is 150% of the original duration.

Speaker Type: Altec 605, 1950s/1960s vintage. B&W 802 is the modern one.

Room: More of a direct sound.

Wall: More diffuse sound.

Top Cut is 24/dB per octave

Bass Cut is 12 dB/octave

Delay Mid Filter = 3.5 kHz

Drive = tape saturation with auto gain adjustment.

Mod = Modulation, AM and FM to the feedback signal.

Filters to Chamber = Post feedback section.

The Abbey Road Studio Reverb Trick: On the reverb send, PRIOR TO THE REVERB EFFECT, low pass equalizer filter at 4-10 kHz. High pass filter at 600 Hz.

Audified SpeakUp Review

My review of Audified’s Speak Up plugin is now up at

As per the usual, here are my script notes.

Audified SpeakUp Review Easy Voiceover & Music Blending

Audified asked me if I could make a review and small tutorial for their new plugin, SpeakUp. Either they did not see my previous DW Drum Enhancer review or they did not care. Either way, not every product is the same and since they asked nicely I figured I would give the company another shot to impress me. This is a review of version 1.0.0

Neither Audified nor a third party are paying me to make this video. I’m using the 30 day trial. If this plugin is a dud I’ll have no problem saying it.

The marketing verbiage is as follows:

Creating voiceovers has never been easier.

SpeakUp was created to simply the process of creating voiceovers and YouTube shows.

SpeakUp “So They Can Hear What You Say”

Adjusting the volume of a spoken word in a video can be time-consuming. In Audified, we thought there could be much easier way to edit voice-overs. Just add music and SpeakUp.

The point of this plugin is that when you are mixing a voice over narration track with music, the music is loud until the narration starts. At that point, the music volume is lowered underneath of the voice over track. This is very common and a time consuming task. SpeakUp’s purpose to save audio engineers time and make it so easy that even the talent can use it. We shall see.

Let me show you how it works.

First thing, an iLok 2 or 3 is required along with iLok License Manager version 3.1.6 or newer. Audified does give instructions on how to get everything working on their web site. So, if you don’t already own an iLok then expect to pay around $200 total for both SpeakUp and the dongle.

Step 1, which Audified does not state in their manual, is that you need to balance the overall volume level between voiceover track and music. The tracks need to be about the same before either of these plugins go on, otherwise this whole thing will not work as intended.

Next, add Sensor to your VO track at the end of the plugin effects chain and Performer to your music track, also at the end of the chain.

I wish they would have called the plugins SpeakUp Voice and SpeakUp Music instead of Sensor and Performer.

Starting at the top left, a GUI size option. Then in this box you can type in a name for the plugin instance.

The wrench button gives is the drop down menu area. Basically, the extra stuff button.

Directly underneath the plugin name is the ON AIR indicator. This works in conjunction with the Sensitivity control.

Look ahead allows the plugin to delay the speech track so music fade outs start before speech. Leave this off at the 0ms setting if you are mixing audio for a video, because it may result in lip synch issues. In future versions of this plugin, I hope that can be changed with plugin delay compensation.

Input level meter is kind of useless and should have filled the width of the plugin window.

Sensitivity is your threshold control. Lower it so that when speech is happening the ON AIR indicator lights up. When voice is muted then the ON AIR light should be off.

Force talk is the parameter you want to use if you don’t want the music to fade in and out while talking is happening. If you need to automate, this is the control to use.

Bypass turns the plugin off, so that it is no longer communicating with the Performer plugin.

Control Bus allows you to use more than one Sensor plugin at the same time. Normally you would use A or B.

Onto the Performer plugin controls…

At the top we have GUI size, then a preset area that allows you to pick or save. The Wrench options menu again.

The Sensor/Automation switch gives us two different ways to use the plugin. Automation mode gives more control. Sensor opens up the Ducking Amount control. I don’t find that control very useful, so I would recommend staying in the default Automation mode.

Target attention is the overall volume reduction level once the ducking amount hits 100%.

Speech Attenuation is the unique feature of this plugin. It allows reduction of frequencies that match the voiceover. This allows for a more organic blend of music and voice over. I actually like setting this control first, with force talk on, until the music starts sounding unnatural. Then dial in the Target Attenuation control to taste.

Bypass is also available along with control bus selection. The question is, when AB is selected does the signal get attenuated more with two tracks at the same time? I will find out later.

Moving onto the Fade Out Time control, this sets how long it takes for attenuation to reach 100% after the voice over track is detected. If you set this to 0, the music will pop in right away, which does not sound very professional.

If you are familiar with noise gates, the Hold Time control should be familiar to you. Basically, it allows a little leeway for the voiceover signal to stop (ON AIR indicator off) before the music track starts fading back in. Ducking is kept at 100% even if the voice signal goes under the sensitivity level. Best seen when fade out and fade in times are at 0.

Finally, fade in is how long it takes for the music track to rise back up to its original volume level after the hold time. A 0 ms setting will be immediate, which does not sound professional.

I want to try this on an unmixed voiceover first. That will be a big challenge for a so-called easy to use plugin like this. I have set all of the controls so that I have to change them, otherwise it will not sound very good. By default, the controls are in a decent spot.

So, I will set my overall volume levels on both tracks.

Next, Sensor Sensitivity.

Speech attenuation is the key feature of this plugin. Normally ducking is done at the overall volume level. That leads to low music volume. Voiceover/music blends are much nicer thanks to this. Don’t attenuate too much, otherwise your music track will sound bad.

Another scenario, talk radio show with a main host and guests. The host should have volume priority, so you set the music track to only trigger off the main host. That track would be set to Control Bus AB. The guest tracks would receive Control Bus A with different attenuation settings compared to the music track.

If all hosts are equal then just add the Sensor plugin at the end of each plugin chain.

Find out how the Control Bus AB choice works.

Test the plugin in real time.

With the Performer plugin, enable write/latch automation on the Ducking Amount parameter

Opinion of the Plugin

GUI resize is nice. Reminds me of old radio station audio equipment.

VST3 only during installation? Requires sidechain so that actually does make sense. VST2 is listed on promotional materials though.

I don’t believe they made this as easy to use as they thought. First, the plugin names as I already mentioned. Second, the Sensitivity knob is the opposite of what newbies would expect. 150% should have been the default size, in an age where 1080p monitors are common.

Analog Hardware Mixing in the Cloud

Do you want to process your audio tracks through analog gear but don’t have the budget?

A company called Distopik is currently beta testing a new service called mix:analog 2.0, where you can remotely control analog audio hardware and hear the changes in near real-time.

Here’s a test I did a couple weeks ago: