Posted on

Multi-Track Drum Editing – Identifying & Splitting Drum Hits

You need to start with a great performance

Before you begin to edit drum stems you have to make sure that you are working with tracks that were recorded close to a click. They need to be consistent. Tightening up the performance is something that is very invasive and requires a lot of time. If the drummer can’t put in the time to learn the parts then you should wait until they are ready to record their parts properly. Having this knowledge will make your life easier and should be something you think about during the preproduction stages of any record.

 

A note about the editing process.

The purpose of this type of editing is to identify the strong hits of the drum beat, split them into tiny parts, and then crop and align those small parts. The splits will depend on which part of the drum falls on each down beat.

In this tutorial Kicks happen every 1/4 note, snares every 2nd and 4th beat, and high hats on every 1/8th note. This happens for about 20 measures with various fills here and there and then it switches to a different pattern. We’ll move in measure by measure increments so that we don’t bite off more than we can chew at first.

Engage the metronome so that you can hear the pulse.  This will help you check your work as you edit. Download the project files here (if you didn’t download them from our previous post)  to get started:

Multi-Track Drum Editing Tutorial

Multi-track drum editing requires you to listen intently to the audio you’re editing. I recommend using headphones for this tutorial so that you can hear subtle edits. Erroneous edits are most exposed in the overheads, high hat, and cymbal frequencies so we’ll need to solo those as well as the kick and snare track while we work through this project.

As we work through the session the high hat and ride will need to be solo’d due to the spot mics that were placed on these. Everything else will follow suit with your editing.

You can also adjust your Track Height in the Track View by dragging the borders of the Continue reading “Multi-Track Drum Editing – Identifying & Splitting Drum Hits”

Posted on

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. Continue reading “DAW Best Practices: How to get a bigger drum sound with reverb”

Posted on

Multi-Track Drum Editing – DLC and Basic Tools

The need for perfect drum production is at an all time high.

In today’s world there is a huge need for all types of drum production. Everything from VST instruments to advanced drum replacement software has been growing in popularity. For the most part, records that require the tracking of live drums always have some sort of drum editing applied. This process is meticulous, long, and can be frustrating if you have never done this much in depth editing before.

Downloadable Content:

Let’s start by getting you the files you need to follow along with this tutorial.

Multi-Track Drum Editing Tutorial

Once downloaded, they should open just fine inside of SONAR X3.

Understanding the basics.

Before diving in, let’s take a look at some essential tools that we’ll be using for major drum editing. These tools may be basic to some, but are definitely the right functions we’ll need in SONAR to edit down these drums.

Creating selection groups

The first step in editing multi-track drums is making selection groups. Once created, these clips will be synced to one another for batch editing tasks – like multi-track editing. During the course of this tutorial we’ll be relying heavily on splitting clips – grouping will make this faster and more efficient.

To create these, choose CTRL+A within the Track View and then right-click on your clips. Near the bottom of the menu there will be an option that says Create Selection Group from selected clips. Select this and a number will appear in the header of your clips indicating that your clips are all in a group now.

As we work through the song the different Split edits will cause the group number to increase. This indicates that a new group has been made. You can change whether or not this occurs within the Preferences here:

 

Tab to Transients

Tabbing to transients locates strong transients and moves Continue reading “Multi-Track Drum Editing – DLC and Basic Tools”

Posted on

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 Continue reading “Basics: Five Questions about Filter Response”

Posted on

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 Continue reading “Basics: Five Questions about Audio Specs”

Posted on

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

Sean Murray

Composer

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. Continue reading “Call of Duty Composer: Sean Murray on SONAR X3 and Gobbler”

Posted on

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? Continue reading “Basics: Five Questions about Latency and Computer Recording”

Posted on

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 Continue reading “Optimizing Vocals with DSP”

Posted on

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 Continue reading “The Art of Transient Shaping with the TS-64”

Posted on

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 Continue reading “Choosing the right compressor in SONAR X3 (Producer & Studio)”