sE Electronics DM1 Dynamite Review

Check out my review video on the DM1 in-line microphone pre audio recording accessory:

Script notes:

DM1 Review

In this video I am taking an in-depth look at sE’s new DM1 inline microphone preamplifier. By my request, they sent me one to review. I’ll be testing most of their marketing claims, to the best of my ability.

Clean/transparent gain: (DM1 Voiceover project) Frequency response test with sweeps and pink noise.

The pink noise average was perfect. The frequency sweep was not but it could be due to the slight change in distance when I unplugged/plugged in the XLR cable. Or that I did not exactly match the preamp levels. Either way, I think this claim passes the meter test very well.

Now, let’s test it on some music material…

28 dB of gain, consistently between loads/microphones (Mic Test for Video project)
Using Youlean Loudness Meter Pro and JoeCo Cello interface…
767a – 30.1 True Peak difference

CO4 – 29.1 TP

SM57 – 29.2 TP

D112 – 29.1 TP

ND 767a Pink noise – 29.6 TP and 29.5 LUFS integrated loudness

Certainly passes and hey I’ll take that extra decibel or two over the 28 dB product specs.

Low noise. (set gain to lowest setting. Boost both. Does the noise sound different? With and without a microphone attached. The AKG may be best for this test next to the SM7b)

Direct plugin vs. short cable vs. long cable performance test.

So four tests:
Direct + 30 feet Mogami

Direct + 10 feet generic

Livewire Advantage → DM1 → 10 feet generic

Livewire → DM1 → Mogami 30 feet

Gain bunching advantage. Set precise levels easier. Ideally every segment would be equally divided. Unfortunately, they are not.

Sound Magic Blue Grand 5 Review

Here is the Real Home Recording video review:

And here’s the script/notes:

The Best Computer Music and Acoustic Technology Inc, better known simply as Sound Magic, have released version 5.1 of their Blue Grand virtual instrument. I requested a license to review it. So, let’s take a listen to it.

Sound Magic sampled four different Steinway pianos. A vintage Steinway B. Nearly antique 1927 Steinway D. Another Steinway D that was built in Hamburg Germany. And then one more Steinway D library that is designed for lower end computers with two different microphone positions.

If you’re not familiar with the Steinway brand, let’s just say that is by far the preferred concert piano, worldwide. And they cost as much as brand new luxury automobiles.

This Blue Grand collection gives you a good variety of mellow and semi-bright grand pianos. My favorite is Blue X then VGS (Grandma Dream). Then Blue, LCD and LSD in that order.

Approximately 16 GB library.

Hybrid sample/algorithmic.

(show on screen)

These are Sound Magic’s descriptions.

– Legacy Blue aka Blue. A Steinway B boasts an elegant sound associated with a Steinway grand piano, yet its inherent versatility works well with a wide range of today’s most popular music genres.

– Vintage 1927 aka BlueX. A 1927 Steinway D offers a rich, luxurious sound with a dynamic range suited to accompanying all music genres, including classical, jazz, pop, and more;

– Living Stereo aka LCD and LSD. Steinway D. Blends crisp, rich tones and great sounding resonance to provide users with a live, vibrant feel that is a great fit for real time performance; thanks to its balanced frequency and singing tone. LSD = player microphone position. LCD = close mic position.

– Grandma Dream aka VGS. Steinway D built in Hamburg, Germany. is characterized by mellow bass and expansive resonance, embracing its Germanic heritage while flawlessly fitting a variety of musical genres;

Authorization is through a machine ID challenge/response system. The web site, email and PDF manual walk you through it pretty well. If you run into any issues, Sound Magic are quick to respond.

I ran into trouble with pasting the keycode. I normally control-V to paste but had to Right-click within the plugin. Also, for some reason one of the codes wouldn’t go through but that was resolved within an hour.

Sound Magic already has a walkthrough video of the features and their manual is well written. So to avoid being redundant, I’ll just give you my opinion and some of the standout controls.

Instrument names aren’t the same as the marketing literature. I wish that wasn’t the case.

Because Neo Piano is a hybrid sample library/algorithmic VI, to a point you can customize all of the pianos to your liking.

Switching dynamic settings after the fact (far right side) can produce interesting results.

Mouseover help hint pop ups.

There were noise blast bugs. Once I loaded the libraries completely into RAM I didn’t run into those issues.

F3 key on REAPER to drop all MIDI.

I didn’t care for the built in reverb.

How to judge piano VIs, courtesy of Sweetwater:

Stereo separation
Attack of the strings
Definition between the notes in the midrange
Tuning of the notes
Overall dynamics of the performance

Choice to store samples in RAM or read off a disk. As with all virtual instruments, solid state drives are recommended for best performance.

I wish they would add the name to the icon. Luckily, if you know how to modify image files you can make your own like I did.

Offers both a standard resolution and 4K ready GUI. For now, the 4K GUI is just a quick up-rez from their standard GUI but it’s better than nothing.

By default the RAM allocation is set very low. If you have a 64-bit computer with 8 GB of RAM or more then set it to 4000.

LCD/LSD are quieter than the others.

Keyboard/pedal velocity response curve detection.

$150 price tag is real good for what you get. May possibly be on sale right now until January 7th. At the normal price, I can’t fully recommend it because right now it’s a half baked product and there is some good competition in the $100-200 price range. Once the major bugs are ironed out, installation process is more streamlined and the 4K GUI is made then yes, I can recommend it.

No demo available except their Yamaha C7 Piano One, which they just put a new version out today.

Blue Cat Audio Axiom Review

Blue Cat’s jam-packed-full-of-features plugin has been out for a month or two and I finally got a chance to review it. The video:

And the script notes:

Version 1.11 was reviewed. BlueCat sent a license for review.

Jam packed full of features. Primarily it is a guitar amp/cabinent simulator…but it’s so much more than that.

In addition to the guitar section, there are over 40 built-in effects. You can even put Axiom inside of itself!

Features List:

distortion pedals, Bit Crusher, Chorus, Comb Filters, Compressor,

Ducker, Echo, Harmonizer, Multitap Delay, EQ, Multimode Filter, Flanger, Frequency Shifter, Gain, Gate, Stereo Pan, Phase Shifter,

Phaser, Pitch Shifter, Pitch Bender, Reverb, Stereo Strip, Sweep Filter, Tremolo, Wah and Waveshaper

Axiom also allows you to host third party plugins in the signal path, INCLUDING VIRTUAL INSTRUMENTS! This is a monster of a plugin, all for 200 dollars or Euros.

All major plugin platforms are supported along with MIDI control. It’s also zero latency, as long as you don’t add third party effects.

A standalone application is also available.

It’s actually difficult to review a plugin like this, because there are so many features.

The amount of controls can be overwhelming. Thankfully there are tons of factory presets to help get you started. The 37 pages long PDF manual is well written

Thanks to the Global GUI Zoom control, the GUI is ready for small screens and 4K monitors alike.

The handy Undo button should maybe be larger and a different color. Redo is also available.

Signal chain: flows left to right and up and down.

Volume Input –> Input Effects –> Parallel amp sim channels A/B with pre and post effects –> Master Channel

Load virtual instruments into the Tools Rack.

Lock parameters button, to prevent preset changes.

The tuner must be enabled.

You can save global presets or presets for each section.

Set up Axiom how you would like it to load, with the Default User preset. I like it at 130% zoom, show controls on (three buttons)

Amp simulator editor (lower case E button, top right corner). This is actually the Destructor GUI, which even has its own PDF manual.

Actually, most if not all of the built-in effects have their own PDF manuals. Just click the question mark icon when their editor windows are displayed

The plugin worked flawlessly with built-in effects. It would crash when loading or unloading certain plugins. Be sure to save your projects before doing so.

Drag and drop functionality was recently added and it is super useful. Duplicate by holding control and dragging an effect. You can even drag between instances of the plugin! .dll files from Windows Explorer right onto Axiom? Yep! There are a few other options, but yeah Blue Cat really thought this feature out.

Dynamics and flexibility are best part. Cleans are great but distortion and pedal effects need a good deal of work to become competitive with other guitar amp sims.

A simple mode would be very beneficial. Their Free Amp plugin is a step in the right direction.

Klevgrand Degrader Review

Degrader is a new bit crusher/resampler/saturation plugin from the maker of Brusfri and GoToEQ. I took a listen to it in the following video.

Script notes are as follows:

Controls: Off, Low-Q, high-Q/steep roll off.

Sample rate goes from 250 Hz to 96000 Hz.

Bit depth 3 bits to 24 bits.

Link buttons

Hold Alt for precise control

Double click for default.

The ever handy input/output and Mix knobs

Pop free bypass Power button.

Preset menu

GUI is not 4K ready

No output meter

VST3 not optional.

Oscillot Audio Perspective Review

I made a double plugin review for two products that are similar and were released within 7 days of each other. Here’s the video:

And here are my script notes for the Oscillot Audio Perspective portion:

With version 4.1 of Sonarworks Reference the speaker emulation and averaging feature was removed for some odd reason. Less than two months later, two new plugins that address this need were released. IK did not remove this virtual speaker function from their ARC system.

Oscillot claims to have been developing this plugin for at least two years. 

The goal of both programs is to save you money on buying new speakers and allow you to quickly flip through them while still in the sweet spot. Listening to your audio on a variety of playback systems is a crucial step of audio production. Making sure the mid range is defined and the lows or high frequencies aren’t too much is a common practice. This also helps avoid ear fatigue. Can also be used for sound design purposes.

You should be listening through a nice set of speakers, because these plugins cannot turn crappy speakers into nice sounding ones. They can turn nice ones into simulated crappy ones though. For my test, I used Klipsch ProMedia 2.1 speakers with the subwoofer off.

These plugins are also a good lesson in that no matter how good your mix is, it will never sound perfect on every playback system.

Tools that save time and money are a great thing. Now, the question is which of this is better?

I really liked the introduction in Perspective’s manual. It gave a lot of background information on how the plugin came into existence/who is behind the making of it.

It has a talkback feature.

Allows for speaker to speaker calibration. The original MixChecker had a broad feature like this but they got rid of it for the Pro version due to variables in speakers. That’s why calibrating speakers with Reference or ARC and then using Perspective in No Speakers mode may be the best bet.

Illustrated speakers without logos and generic but guessable names are shown, so the guessing game is a little easier.

Intro Price until August 15: $149

Very difficult to read the settings menu.

Noise only works on the automobile speakers.

Minimize button (bottom right).

Perspective has more studio speaker types.

I like Perspectives GUI better but I think MCP sounds more realistic. It has a more 3D sound to it whereas Perspective sounds more like an EQ. Like the speakers are being replaced as opposed to hearing the true devices. If I owned a pair of speakers that Perspective supports then I may feel differently.

Perspective has radio simulation…although it is kind of gimmicky.

Perspective uses less 63 MB less RAM compared to MixChecker Pro.

Perspective takes source speakers into account.

Does not require an iLok dongle.

Perspective accounts for your subwoofer.

Perspective has a good 20 minutes long video tutorial.

Has polarity flip

Perspective is cheaper right now, until August 15, 2018.

Perspective officially supports MacOS 10.7 Lion while MCP only officially supports Mavericks or newer.

Both are Perspective and MixChecker Pro are competent programs that will get the job done. It is a toss up on which is better.  I would recommend both as quick ways to check mixes on virtual speaker setups. Right now, Perspective is the better value and does not require iLok software to run. Audified sounds better to my ears but you may feel differently. The nice thing is, if you are in the market for software like this both can be demoed.

If you use speaker calibration plugins such as Sonarworks Reference or IK Multimedia ARC, you should put those AFTER Perspective.

Audified MixChecker Pro Review

I reviewed Audified’s new Mix Checker Professional version virtual speaker simulator plugin.

Script notes (for the MCP portion):

With version 4.1 of Sonarworks Reference the speaker emulation and averaging feature was removed for some odd reason. Less than two months later, two new plugins that address this need were released. IK did not remove this virtual speaker function from their ARC system.

Audified asked me to review their new MixChecker Pro plugin and sent me a NFR license. Just as I was about to start testing MCP in depth I saw that a new company called Oscillot Audio released a very similar plugin called Perspective.

The standard version of MixChecker was released back in 2016. 

The goal of both programs is to save you money on buying new speakers and allow you to quickly flip through them while still in the sweet spot. Listening to your audio on a variety of playback systems is a crucial step of audio production. Making sure the mid range is defined and the lows or high frequencies aren’t too much is a common practice. This also helps avoid ear fatigue. Can also be used for sound design purposes.

You should be listening through a nice set of speakers, because these plugins cannot turn crappy speakers into nice sounding ones. They can turn nice ones into simulated crappy ones though. For my test, I used Klipsch ProMedia 2.1 speakers with the subwoofer off.

These plugins are also a good lesson in that no matter how good your mix is, it will never sound perfect on every playback system.

MixChecker Pro was released first, so I will talk about it first. I spent over six hours evaluating both plugins. If you guys would have sat with me for all of this, you would have been bored to tears. So, I will try to keep this video short for your and my sanity.

Wish I could flip through presets with the mouse scrollwheel or keyboard arrows.

Can re-arrange buttons and use custom names.

Set Calibration (in the wrench menu) is important for the distortion feature.

Noise is a cool feature. The default volume should be lower though.

It’s a shame they did not include a vinyl option that a Gearslutz user suggested back in 2016.

Supports VST, VST3, Audio Units and AAX. 32-bit and 64-bit for most platforms.

GUI resizing, zero latency, less CPU, even with distortion on.

Custom labels but no pictures and more generic names.

MCP has an app feature. This allows you to keep the plugin closed and also walk around the control room while auditioning different speaker types
Perspective has a Dim function.

MCP has an auto switcher

Requires iLok software to be installed. It doesn’t require the USB hardware dongle though.

Does not account for subwoofer.

Models distortion

Constant loudness/volume match is better than Perspective.

Audified sounds better to my ears but you may feel differently. The nice thing is, if you are in the market for software like this both can be demoed.

To answer the question…calibration software before or after these plugins? I say before. If you have speakers that Perspective supports then consider disabling Sonarworks Reference or IK Multimedia ARC or whatever you have. If you already own ARC then you may not need either of these plugins.

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.