ProChannel Convolution Reverb with REmatrix Solo

by Dan Gonzalez

Convolution Reverb, now in the ProChannel

New to SONAR Professional and SONAR Platinum is the increasingly popular and imaginative REmatrix Solo. REmatrix Solo uses convolution to mimic real life halls, rooms, plates, and other reverberant spaces. In order to do this, engineers use something called a sine sweep or starter pistols to excite a real space like a church or bathroom. Typically you need to use a space that has a particularly natural short or long decay and does not have artifacts like flutter verbs or cancelling frequencies. You can even capture the sound of other reverb plugins and import that sound into REmatrix Solo.

This version is based off of a the full REmatrix plugin by Overloud. Currently, REmatrix Solo allows users to play a single IR but in the full version, you can play up to 5 IRs at once – allowing you to cross-pollinate your favorite reverbs into one lush space.

What’s the difference between Breverb and REmatrix Solo

Breverb and REmatrix Solo actually use different technology to create reverb. Breverb is based on a famous digital reverb found in almost every major recording studio. Breverb uses a dedicated or similar algorithm to recreate things like Early Reflections, Late Reflections, Pre-Delay, Decay, and other elements of a reverb. Breverb recreates a digital reverb whereas REmatrix uses the aforementioned convolution methods to convolve passing sounds with data from real life spaces. Breverb lets you tweak the elements of reverb and REmatrix creates a space around your sound.

Here’s a in depth look at the REmatrix Solo plugin brand new to SONAR Professional and SONAR Platinum.



SONAR User Norman Matthew brings in CJ Pierce of DROWNING POOL to his “Sound Foundation” Music School

SONAR Platinum user Norman Matthew of Murder FM has been making some serious musical strides on many fronts.  Since touring internationally and becoming a new dad wasn’t enough, he decided to open a “School of Rock” so to speak in what tiny bit of spare time he could squeeze.  ”The Sound Foundation” which houses SONAR Platinum at the core, is based on teaching kids the side of music that one generally will not learn in a class room.  ”This is about teaching kids how to find their heart and spirit through music,” Norman told me at #Namm2015 when we caught up for a bit.  ”The changes I see in the confidence of these kids through the process is amazing.  Each student is treated uniquely to identify exactly what is needed to reach their musical goals.  What happens throughout that process is simply magical to witness, and really what makes this place so special.  SONAR is a big part of this place.  These kids love it and recording some great music on their own thanks to the simplicity of the program.”

Quite frankly, The Sound Foundation is picking up where the labels are falling short on the Artist Development side of the coin.  Since the fall of the Major Label system, the words “Artist Development” are rarely heard.  A lot of people don’t realize that a good amount of money that was coming into the labels back-in-the-day was going right back into developing new artists.  Labels were actually gambling on 150k deals here and there knowing that 1 out of 5 newly signed artists may break long term.  Those days are long gone, and this is where people like Norman Matthew are picking up the slack.

This model is working, and to drive home the point Norman has recently brought on CJ Pierce of Platinum selling Artist DROWNING POOL to join The Sound Foundation.  ”This is a big deal for me.  I have always had so much respect for CJ and Drowning Pool – and to have a guy of this caliber involved with our program is priceless.”

CBS Dallas has bought into Norman’s educational process and ran a piece on The Sound Foundation today:

The Ford Company has also taken notice of Norman’s work ethic and has featured The Sound Foundation on Episode #1 of their Good Works Series:

For more information on The Sound Foundation visit

For more information on the new SONAR family visit


Video from Winter NAMM 2015


Highlights from Winter NAMM 2015
NAMM was our first opportunity to show off the new SONAR line to the public, and the reception was nothing short of spectacular. To handle the crowds, products were demoed at three locations—the Gibson, TASCAM, and Hal Leonard booths. We met artists, press, and of course many, many customers—and we were equally happy to thrill long-time Cakewalk supporters as well as bring new users into the fold.

Cakewalk NAMM 2015

We wish everyone could experience the excitement of NAMM, but to give you a taste just click the links below to see demos and interviews from the show floor. And—there’s also a sneak peek of the new David Bendeth Signature Series Compressor.

Dan Gonzalez demos the new features in SONAR

Audiofanzine gets a demo of the new SONAR

Keyboard magazine interviews Craig Anderton on Membership

Engineer/Producer, John Paterno shows off Overloud REmatrix

Sneak peek at new David Bendeth Signature Series Compressor



Mix Recall Remembers Your Instrument Settings

by Dan Gonzalez

Mix Recall takes your mixing to another level by offering SONAR Artist, Professional, and Platinum users the ability to save different mix scenes of the same mix within a single project. Mix Recall saves track parameters, bus parameters, and even instrument presets. A great way to use this feature is to audition different drumkits using the included Addictive Drums 2.


Instruments these days are full of all kinds of choices, especially ones that are as expansive as Addictive Drums 2. When working on a track I like to take the same pattern and switch between the custom kits that I’ve made. Addictive Drums 2 and Addictive Drums 1 both let the user take pieces of all the different kits that it comes with to make your own. Mix Recall let’s you take this workflow a step further.

Original drum passage

Here we have a simple Indie Kit from Addictive Drums:


Saving the first mix scene

Go to the Mix Recall module in SONAR and click on (more…)


Basics: Five Questions About Panning Laws

By Craig Anderton

It’s not just a good idea, it’s the law…panning law, that is. Let’s dispel the confusion surrounding this sometimes confusing topic.

What does a panning law govern? When a mono input feeds a stereo bus, the panning law determines the apparent and actual sound level as you sweep from one side of the stereo field to the other.

But why is a “law” needed? Doesn’t the level just stay the same as you pan? Not necessarily. Panning laws date back to analog consoles. If a pan control had a linear taper (in other words, a constant rate of resistance change as you turned it), then the sound was louder when panned to center. To compensate, hardware mixers used non-linear resistance tapers to drop the level, typically by -3 dB RMS, at the center. This gave an apparent level that was constant as you panned across the stereo soundstage. If that doesn’t make sense…just take my word for it, and keep reading.

Okay, then there’s a law. Isn’t that the end of it? Well, it wasn’t really a “law,” or a standard. Come to think of it, it wasn’t a specification or even a “recommendation.” Some engineers dropped the center level a little more to let the sides “pop” more, or to have mixes seem less “monoized” and therefore create more space for vocalists who were panned to center. Some didn’t drop the center level at all, and some did custom tweaks.

Why does this matter to a DAW like SONAR, which doesn’t have a hardware mixer? Different DAWs default to different panning laws. This is why duplicating a mix on different DAWs can yield different results, and lead to foolish online discussions about how one DAW sounds “punchier” or “wimpier” than another if someone brings in straight audio files and sets the panning and faders identically.

A mono signal of the same level feeds each fader pair, and each pair is subject to different SONAR panning laws. Note the difference in levels with the panpot panned to one side or centered. The tracks are in the same order as the descriptions in SONAR’s panning laws documentation and the listing in preferences. Although the sin/cos and square root versions may seem to produce the same results, the taper differs across the soundstage between the hard pans and center.

This sounds complicated, and is making my head explode—can you just tell me what I need to do so I can go back to making music? SONAR provides six different panning law options under Preferences, so not only can you choose the law you want, the odds of being able to match a different DAW’s law are excellent. The online help describes how the panning laws affect the sound. So there are really only two crucial concepts:

  • The pan law you choose can affect a mix’s overall sound if you have a lot of mono sound sources (panpots with stereo channels are balance controls, which is a whole other topic). So try mixes with different laws, choose a law you like, and stick with it. I prefer -3 dB center, sin/cos taper, and constant power; the signal level stays at 0dB when panned right or left, but drops by -3 dB in each channel when centered. This is how I built hardware mixers, so it’s familiar territory. It’s also available in many DAWs. But use what you like…after all, I’m not choosing what’s “right,” I’m simply choosing what I like.
  • If you import an OMF file from another DAW or need to duplicate a mix from another DAW, ask what panning law was used in creating the file. One of SONAR’s many cool features is that it will likely be able to match it.

There, that wasn’t so bad. Ignorance of the law is no excuse, and now you have answers to five questions about panning laws.



Basics: Five Questions About Using Stompboxes with SONAR

by Craig Anderton

Plug-in signal processors are a great feature of computer-based recording programs like SONAR, but you may have some favorite stompboxes with no plug-in equivalents—like that cool fuzz pedal you love, or the ancient analog delay you scored on eBay. Fortunately, with just a little bit of effort you can make SONAR think external hardware effects are actually plug-ins.

1. What do I need to interface stompboxes with SONAR? You’ll need a low-latency audio interface with an unusd analog output and unused analog input (or two of each for stereo effects), and cords to patch these audio interface connections to the stompbox. We’ll use the TASCAM US-4×4 interface because it has extra I/O and low latency, but the same principles apply to other audio interfaces.

2. How do I hook up the effect and the interface? SONAR’s External Insert plug-in inserts in an FX bin, and diverts the signal to the assigned audio interface output. You patch the audio interface output to a hardware effect’s input, then patch the hardware effect’s output to the assigned audio interface input. This input returns to the External Effect plug-in, and continues on its way through the mixer. For this example, we’ll assume a stompbox with a mono input and stereo output.

3. What are correct settings for the External Insert plug-in parameters? When you insert the External Insert into the FX bin, a window appears that provides all the controls needed to set up the external hardware.

  • Send. This section’s drop-down menu assigns the send output to the audio interface. In this example, the send feeds the US-4×4’s output 3. Patch this audio interface output to your effect’s input. (Note that if an output is already assigned, it won’t appear in the drop-down menu.)
  • Output level control. The level coming out of the computer will be much higher than what most stompboxes want, so in this example the output level control is cutting the signal down by about -12 dB to avoid overloading the effect.
  • Return. Assign this section’s drop-down menu to the audio interface input through which the stompbox signal returns (in this example, the US-4×4’s stereo inputs 3 and 4). Patch the hardware effect output(s) to this input or inputs.
  • Return level control. Because the stompbox will usually have a low-level output, this slider brings the gain back up for compatibility with the rest of the system. In this example, the slider shows about +10 dB of gain. (Note: You can invert the signal phase in the Return section if needed.)

4. Is it necessary to compensate for the delay caused (more…)


Call of Duty Composer: Sean Murray on SONAR X3 and Gobbler

Sean Murray


Being projected as one of the biggest selling pieces of media in history is a pretty big deal. On November 9th 2010, Call of Duty®: Black Ops was released and is being labeled as just that; and SONAR was the engine behind the scenes for the score. Even more exciting for one man, Sean Murray (Composer/Producer), is the fact that Activision has simultaneously released his soundtrack on a worldwide basis through an arm of Universal Music Distribution to coincide with the release of the game. (more…)


Basics: Five Questions about Latency and Computer Recording

Get the lowdown on low latency, and what it means to you

By Craig Anderton 

Recording with computers has brought incredible power to musicians at amazing prices. However, there are some compromises—such as latency. Let’s find out what causes it, how it affects you, and how to minimize it.  

1. What is latency? When recording, a computer is often busy doing other tasks and may ignore the incoming audio for short amounts of time. This can result in audio dropouts, clicks, excessive distortion, and sometimes program crashes. To compensate, recording software like SONAR dedicates some memory (called a sample buffer) to store incoming audio temporarily—sort of like an “audio savings account.” If needed, your recording program can make a “withdrawal” from the buffer to keep the audio stream flowing. 

Latency is “geek speak” for the delay that occurs between when you play or sing a note, and what you hear when you monitor your playing through your computer’s output. Latency has three main causes: 

  • The sample buffer. For example, storing 5 milliseconds (abbreviated ms, which equals 1/1000th of a second) of audio adds 5 ms of latency (Fig. 1). Most buffers sizes are specified in samples, although some specify this in ms. 

 Fig. 1: The control panel for TASCAM’s US-2×2 and US-4×4 audio interfaces is showing that the sample buffer is set to 64 samples. 

  • Other hardware. Converting analog signals into digital and back again takes some time. Also, the USB port that connects to your interface has additional buffers. These involve the audio interface that connects to your computer and converts audio signals into digital signals your computer can understand (and vice-versa—it also converts computer data back into audio).
  • Delays within the recording software itself. A full explanation would require another article, but in short, this usually involves inserting certain types of processors within your recording software. 

2. Why does latency matter? (more…)


Optimizing Vocals with DSP

Optimizing tracks with DSP, then adding some judicious use of the DSP-laden VX-64 Vocal Strip, offers very flexible vocal processing. 

By Craig Anderton 

This is kind of a “twofer” article about DSP—first we’ll look at some DSP menu items, then apply some signal processing courtesy of the VX64—all with the intention of creating some great vocal sounds. 


“Prepping” a vocal with DSP before processing can make the processing more effective. For example, if you want to compress your vocal and there are significant level variations, you may end up adding lots of compression to accommodate quiet parts. But then when loud parts kick in, the compression starts pumping. 

Here’s another example. A lot of people use low-cut filters to banish rogue plosives (e.g., a popping “b” or “p” sound). However, it’s often better to add a fade-in to get rid of the plosive; this retains some of the plosive sound, and avoids affecting frequency response. 

Adding a fade-in to a plosive can get rid of the objectionable section while leaving the vocal timbre untouched. 

Also check if any levels need to be evened out, because there will usually be some places where the peaks are considerably higher than the rest of the vocal, and you don’t want these pumping the compressor either. The easiest fix is to select a track, drag in the timeline above the area you want to edit, then go Process > Apply Effect > Gain and drop the level by a dB or two. 

This peak is considerably louder than the rest of the vocal, but reducing it a few dB will bring it into line. 

Also note that if you have Melodyne Editor, you can use the Percussive algorithm with the volume tool to level out words visually. This is really fast and effective. 

While you’re playing around with DSP, this is also a good time to cut out silences, then add fadeouts into silence, and fadeins up from silence. Do this with the vocal soloed, so you can hear any little issues that might come back to haunt you later. Also, sometimes it’s a good idea to normalize individual vocal clips up to –3dB or so (leave some headroom) so that the compressor sees a more consistent signal. 

The clip on the left has been normalized and faded out. The silence between clips has been cut away. The clip on the right fades in, but has not been normalized. 

With DSP processing, it’s good practice to work on a copy of the vocal, and make the changes permanent as you do them. The simplest way to apply (more…)


The Art of Transient Shaping with the TS-64

Understand this often-misunderstood processor, and your tracks will benefit greatly 

By Craig Anderton 

Transient Shapers are interesting plug-ins. I don’t see them mentioned a lot, but that might be because they’re not necessarily intuitive to use. Nor are they bundled with a lot of DAWs, although SONAR is a welcome exception. 

I’ve used transient shaping on everything from a tom-based drum part to make each hit “pop” a little more, to bass to bring out the attacks and also add “weight” to the decay, to acoustic guitar to tame overly-aggressive attacks. The TS-64 has some pretty sophisticated DSP, so let’s find out how to take advantage of its talents.

But first, a warning: transient shaping requires a “look-ahead” function, as it has to know when transients are coming, analyze them, filter them, and then calculate when and how to apply particular amounts of gain so it can act on the transients as soon as they occur. As a result, simply inserting the TS-64 will increase latency. If this is a problem, either leave it bypassed until it’s time to mix, or render the audio track once you get the sound you want. Keep an original of the audio track in case you end up deciding to change the shaping later on. 


A Transient Shaper is a dynamics processor that modifies only a signal’s attack characteristics. If there’s no defined transient the TS-64 won’t do much, or worse yet, add unpleasant effects. 

Transient shapers are not just for drums—guitars, electric pianos, bass, and even some program material are all suitable for TS-64 processing if they have sharp, defined transients. And it’s not just about making transient more percussive; you can also use the TS-64 to “soften” transients, which gives a less percussive effect so a sound can sit further back in a track. 

There are two main elements to transient shaping. The first is (more…)