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…)

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Mixing Heavy Metal with the ProChannel & Softube Mix Bundle

The Softube Mix Bundle is a strong and creative addition to SONAR’s ProChannel strip.  This bundle adds 5 solid effects, great for any mix, to the Softube Saturation Knob already in SONAR X3 Producer.

For this article I’ve mixed a Heavy Metal track from the group Dark Ride using mostly Softube ProChannel effects. You can download the project here and follow along if you have the Softube Mix Bundle. If not then the screenshots in this article should suffice.

Setting up the Mix

Listen & add Markers

At first listen I put in Markers throughout the entire project to make navigation and looping sections much easier. Using the shortcut M – it’s pretty easy to drop in a Marker wherever your Now Time Marker resides. After that, you can name them accordingly. This paticular song was relatively short and included an introduction, two verses, 3 choruses, bridge, solo section, and breakdown.

Routing, grouping, and track folders

While you’re mixing it’s easy to become slightly overwhelmed by larger projects. What I do in this instance is make a stereo bus for every group of instruments that I have in the project. This allows me to apply mixing effects to the instrument groups as whole before they hit my main mix bus. The tracks route directly to the buses and then the buses route directly to the 2 bus. For each instrument group I also assigned them a color category and a track folder to make things a bit easier to manage within the Track View.

Levels & panning

Metal in general consists of abrasive-wide rhythm guitars, huge-punchy drums, (more…)

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DAW Best Practices: How to get a bigger drum sound with reverb

The Biggest, Baddest Drum Reverb Sound Ever

[Originally posted as a daily tip on the SONAR forums and reposted for viewers here on the blog.]

by Craig Anderton

You want big-sounding drums? Want your metal drum tracks to sound like the Drums of Doom? Keep reading. This technique transposes a copy of the reverb and pans the two reverb tracks oppositely. It works best with unpitched sounds like percussion.

1. Insert a reverb send.

Insert a send in your drum track, then insert your reverb of choice in the Send bus.

 

2. Render the reverb, isolated from the drum track. (more…)

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Basics: Five Questions about Filter Response

By Craig Anderton 

You can think of filters as combining amplification and attenuation—they make some frequencies louder, and some frequencies softer. Filters are the primary elements in equalizers, the most common signal processors used in recording. Equalization can make dull sounds bright, tighten up “muddy” sounds by reducing the bass frequencies, reduce vocal or instrument resonances, and more. 

Too many people adjust equalization with their eyes, not their ears. For example, once after doing a mix I noticed the client writing down all the EQ settings I’d done. When I asked why, he said it was because he liked the EQ and wanted to use the same settings on these instruments in future mixes. 

While certain EQ settings can certainly be a good point of departure, EQ is a part of the mixing process. Just as levels, panning, and reverb are different for each mix, EQ should be custom-tailored for each mix as well. Part of this involves knowing how to find the magic EQ frequencies for particular types of musical material, and that requires knowing the various types of filter responses used in equalizers. 

What’s a lowpass response? A filter with a lowpass response passes all frequencies below a certain frequency (called the cutoff or rolloff frequency), while rejecting frequencies above the cutoff frequency (Fig. 1). In real world filters, this rejection is not total. Instead, past the cutoff frequency, the high frequency response rolls off gently. The rate at which it rolls off is called the slope. The slope’s spec represents how much the response drops per octave; higher slopes mean a steeper drop past the cutoff. Sometimes a lowpass filter is called a high cut filter.

 Fig. 1: This lowpass filter response has a cutoff of 1100 Hz, and a moderate 24/dB per octave slope.

What’s a highpass response? This is the inverse of a lowpass response. It passes frequencies above the cutoff frequency, while rejecting frequencies below the cutoff (Fig. 2). It also (more…)

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Basics: Five Questions about Audio Specs

By Craig Anderton 

Specifications don’t have to be the domain of geeks—they’re not that hard to understand, and can guide you when choosing audio gear. Let’s look at five important specs, and provide a real-world context by referencing them to TASCAM’s new US-2×2 and US-4×4 audio interfaces. 

First, we need to understand the decibel (dB). This is a unit of measurement for audio levels (like an inch or meter is a unit of measurement for length). A 1 dB change is approximately the smallest audio level difference a human can hear. A dB spec can also have a – or + sign. For example, a signal with a level of -20 dB sounds softer than one with a level of -10 dB, but both are softer than one with a level of +2 dB. 

1. What’s frequency response? Ideally, audio gear designed for maximum accuracy should reproduce all audible frequencies equally—bass shouldn’t be louder than treble, or vice-versa. A frequency response graph measures what happens if you feed test frequencies with the same level into a device’s input, then measure the output to see if there are any variations. You want a response that’s flat (even) from 20 Hz to 20 kHz, because that’s the audible range for humans with good hearing. Here’s the frequency response graph for TASCAM’s US-2×2 interface (in all examples, the US-4×4 has the same specs).

This shows the response is essentially “flat” from 50 Hz to 20 kHz, and down 1 dB at 20 Hz. Response typically goes down even further below 20 Hz; this is deliberate, because there’s no need to reproduce signals we can’t really hear. The bottom line is this graph shows that the interface reproduces everything from the lowest note on a bass guitar to a cymbal’s high frequencies equally well. 

2. What’s Signal-to-Noise Ratio? All electronic circuits generate (more…)

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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…)

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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 “MENU” DSP 

“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…)

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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. 

TS-64 TRANSIENT SHAPER BASICS

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…)

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Choosing the right compressor in SONAR X3 (Producer & Studio)

What is Compression?

Compression is a massively useful tool for pro audio applications. As a simultaneous corrective and creative utility suitable for both tonal shaping and controlling levels,  a compressor is one of the most important pieces of gear in your sonic toolbox.

Instead of explaining the history and value of knowing all the different types of compressors that exist, we’re just going to dive in and show you how to get results. Once you understand this you’ll be able to grasp the larger picture of compression and the many different circuits and types. SONAR X3 Studio & Producer come packed with quite a few different types of compressors, so let’s open them up and take a look.

PC76 U-Type

Modeled after one of the most classic leveling amplifiers in history, the PC76 U-Type is a go-to compressor for (more…)

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How SONAR user Bobbi Tammaro won a SESAC award over many Major Label Artists

Respect and credibility

A few weeks ago SESAC announced their 2013 Jazz Award recipients, and SONAR X3 user Bobbi “Funkeeboy” Tommaro was one of the well-deserved artists on the list.  I have been fortunate to get to know Bobbi in the last few years on and off the SONAR playing-field, and the first word that comes to mind is “respect.”  These days considering the state of the music industry, the word respect has more meaning than ever and Bobbi has earned it from his music peers along with just about everyone else that hears his story.  Besides respect, he has earned much credibility in the Smooth Jazz world from being a repeat-offender on the Billboard charts as an independent artist.

Bobbi who is also PKA “Funkee Boy” has amassed an impressive amount of credits throughout his music career. At the young age of 15 he was already making a name for himself winning the Connecticut State Organ Championship and also opening for national acts such as Spyro Gyra.  As he progressed he scored numerous chart hits across multiple music genres, as well as, several top Billboard chart hits/Top 5 Smooth Jazz hits, and licensing deals on television networks such as ABC, FOX, NBC, CBS, SHOWTIME, VH1, etc.

Before making his own records his music has [and continues to in terms of residuals] appear on some of the most popular and well respected shows ever such as Beverly Hills 90210, General Hospital, All My Children, Sunset Beach, Ripley’s Believe It Or Not, Love Boat, One Life To Live, Young & The Restless, They Call Me Sirr, Soul Food.  Stepping out from behind the scenes as a songwriter/producer & into the forefront as a Smooth Jazz Artist, Bobbi’s track record progressed and continued to impress.

Most recently, starting off 2014 by releasing his 3rd CD “Soul Purpose”, the smooth jazz keyboardist/producer combined his talents with assembling a stellar line up of A-List recording artists. The newest award-winning record “Soul Purpose” features guest appearances from Warren Hill, Najee, Bob Baldwin, Cindy Bradley, Nick Colionne, LEILA, Surface, Lamone, Timmy Maia, Tevin Michael and more!!!

Chalk one up for the hard working jugglers. 

A few key elements set Bobbi apart from the pack that keeps him successful.  The obvious one is the raw talent of songwriting and performing his instrument, but if you go a few levels deeper, you will find a multi-instrumentalist, producer, mixing and mastering engineer.  Peeling back a few more layers exposes an organized machine who literally “does not need a label” to hit the Billboard Top 5 Smooth Jazz Chart.  In fact Bobbi has had many labels approach him and has respectfully declined any offers to do business.  Why?… because he has cracked the code and found a formula that works for himself as an independent artist competing in the major leagues.

It’s not easy

Hitting the Top 5 on any Billboard Chart is not an easy thing to do.  Besides the songs and production alone (which he does ALL in SONAR INCLUDING Mastering), Bobbi also has to oversee the efforts for Radio Promotion, Publicity, Social Media, Sales and Marketing.  If you ask me, this is a very rare skill-set to have as an artist considering just the time it takes alone to write and record a [great] full length record.  Bobbi does have help from his wife Leila who is also a very credible artist, and the two of them seem to have a great formula for getting the music out to the masses as if they were a Major Label.

Cakewalk:          It’s pretty amazing that you do so much to get your music out, can you describe the short-form version of a typical record release?

Bobbi Tammaro:            Sure, it’s hard to keep it short form because so much goes into it… So here goes (more…)

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