How to Make a Noise: Sample-Based Synthesis

How to Make a Noise: Sample-Based Synthesis by Simon Cann

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Coombe Hill Publishing

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coombehillpublishing.com

Copyright © Simon Cann 2005, 2007, 2011

All rights reserved. No part of this publication may be reproduced in any form or by any means without the prior permission of the publisher.

While every effort has been taken to ensure accuracy, the author and the publisher make no representations about the accuracy of the information contained in this book and cannot accept responsibility for any errors or omissions, or for any loss based on the content of this book.

Kontakt 4 and Battery 3 are part of Komplete which was created by Native Instruments GmbH.

This book has not been endorsed by any of the corporations or individuals listed. Equally, neither the publisher nor the author endorse or offer any support for any of the products featured in this book.

Front cover background image © Pablo631 | Dreamstime.com

ISBN 978-0-9554955-6-4

20111002-1.0

About the Author

Simon Cann is a writer based in London. He is the author of a number of music- and business-related books.

His music-related books include: Cakewalk Synthesizers: From Presets to Power User, Becoming a Synthesizer Wizard: From Presets to Power User, and Jumpstart Your Music Career. His business-related books feature the experiences of international entrepreneurs who have built successful companies in the hottest business locations around the world.

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You can find out more about Simon and keep up-to-date in many places:

Read more about Simon at his website, simoncann.com.

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Introduction

Welcome and thank you for buying this book.

How to Make a Noise: Sample-Based Synthesis introduces the tools and techniques so you can create and control sounds using sample-based synthesis.

Sample-based synthesizers come in three main variants:

The most conventional form of sampler is intended to recreate the sounds of natural instruments. As such these are highly powerful instruments with a wide range of tools, and while they can do much more than simply reproduce a real instrument perfectly, their main focus is on ensuring detail and fidelity in sonic reproduction.

Then there are the instruments that use samples as the basis for creating new sounds. The source sound is processed and manipulated—both by twisting the source in the digital domain and by applying a range of analog-like tools and techniques. These samplers can behave like more conventional samplers; however, the main focus of these tools is on creating sounds rather than fidelity in reproduction.

The third group of samplers are the specialist tools—the main specialism being rhythm and drum-specific samplers. As well as creating and controlling the sound of individual drums, many of these units have dedicated tools for replaying loops.

To a certain extent there is a cross-over between the three categories and it is hard to draw a line where one group ends and the next begins, but for the purposes of this book, let's accept that these are the main options.

This book is about sample-based synthesis techniques—it is not about specific samplers and it only deals with sampling in the context of creating content that can be deployed. While the techniques are illustrated in this book with two specific samplers, they can be applied to virtually any sampler.

And by the way, this book is not "the missing manual" for either of the featured samplers (since they both come with very good manuals). Instead, it is intended as a ground-up learning experience, in other words, we'll look at deploying these techniques from first principles. At all stages I would encourage you to try things out and get your hands dirty: there is no need for precision—get out there and create the sounds that sound good to you!

... And Then They Invented Sampling

Until the 1980s, if you wanted a piano sound or a drum sound, or any specific sound for that matter, the only realistic option was to get hold of that instrument, someone who could play it, and find a recording studio. Sure, there were some instruments that could reproduce the tones of other instruments (most notably the Mellotron), but in reality, if you wanted an instrument on your recorded, then you needed to find the instrument.

However, this changed in the early 1980s.

Perhaps the first major step forward was the Fairlight CMI which is regarded by many as the first digital sampling synthesizer. In parallel with the Fairlight came the Synclavier, and together the characteristic sounds created by these new instruments dominated the charts for a while. They also gave rise to a new phenomenon of singing animals—by sampling a cat or a dog, it was possible to then play a melody thereby making cats and dogs sing. In case you're wondering, yes, it was that bad, but that didn't stop advertisers using it.

The Fairlight and the Synclavier were both keyboard instruments run on a computer (and a 1980s computer at that, with nasty monochrome monitors). At the time these were the top flight instruments and were ferociously expensive—you could (literally) buy a house for less than you would pay for one of these two.

In parallel with these instruments, a whole new class of instrument started to develop: the digital drum machine. These were also hugely expensive, but not quite in the same price league as the Fairlight or Synclavier (roughly the price of a good new car rather than a house).

The first widely adopted digital drum machine was the LM-1 Drum Computer, often called the LinnDrum (as it was produced by Linn Electronics). However, strictly speaking, the LinnDrum was the product that followed the LM-1.

The LM-1 was very expensive, but out of the price range of most musicians. The $1,000 price barrier (and remember, that's $1,000 in the early 1980s) was first broken by E-mu with their Drumulator. This was a simple machine offering only 12 sounds. However, when it was released, it was revolutionary.

The Drumulator was succeeded by the SP-12 which in turn was later succeeded by the SP-1200. These machines are basic by today's standards, using 12-bit audio (which I'll explain in the next chapter). Today we think of 12-bit as lo-fi and gritty, but in the early 1980s it was the highest quality that was available. However, the grittiness (which some perceived as giving a vinyl record-like sound) was used for its character, and the E-mu machines were rapidly taken to their hearts by the hip-hop community.

In parallel with the dedicated digital drum machines, dedicated hardware samplers came to the market. At the affordable end of the market (in other words, expensive, but you didn't need to sell a body organ), three companies dominated: E-mu, Akai, and Roland. From the mid-1980s and through the 1990s these companies produced a wealth of ever more powerful samplers.

With each new model, the size of the memory increased. This was a good thing... In order to play a sample on a hardware sampler, it had to be loaded into memory. Therefore the size of the sample sets was limited by the size of the sampler's memory. With drum machines this wasn't a problem, but with more complex instruments—especially instruments with a wide pitch range, a wide range of volume tones, and a long sustain sound, such as pianos or orchestral instruments—this was a problem.

In order to reproduce these complicated instruments, compromises were made. Compromises were made in three areas:

The number of loudness layers was reduced. Sometimes there was only one loudness layer and any tonal changes were effected by a filter.

The range over which each sample was played was increased. This meant that there were fewer samples, which led to noticeable tonal changes when moving from one sample to another.

The third compromise was in the length of the samples. In order to reduce the file size, the audio waves were cut and the missing audio was replaced by a looped section.

These compromises all had a detrimental effect on the sound, and the initial thrill for musicians in being able to reproduce real instruments soon led to disappointment about some of the results. That being said, many programmers were able to achieve excellent results with tiny files.

The next major change came with the introduction of software instruments in the early 2000s. This brought two significant changes:

With the interface now being on a computer screen (rather than being a tiny LCD display as was common on most hardware samplers), it became a much more straightforward task to create and edit sound sets.

The second change was perhaps even more significant: disk streaming. The first sampler to allow this was the NemeSys GigaSampler (which later became the Tascam GigaStudio). The significant innovation this tool introduced was to play samples directly from the hard drive rather than requiring them to first be loaded into the memory. This broke the limitations of installed memory on a computer.

Disk streaming led to a significant change for sample libraries—they became larger and they became more detailed too, including more tonal nuances and playing techniques. In parallel, many sample developers also released their newly enlarged sample sets as instruments rather than simply as wave sets. By releasing an instrument the samples would be deployed exactly as they were intended to be used.

Today, there are few hardware samplers. Even musicians on the road tend to prefer software samplers (although some of those pieces of software may be wrapped in a hardware unit like Receptor).

There are a number of top flight software samplers which are used by a wide range of professional musicians. These include Halion from Steinberg and Kontakt from Native Instruments. In addition many hosts include samplers, for instance Sampler which comes with Ableton Suite, and EXS24 which comes with Apple's Logic.

There are also many instruments which use a sampling engine as the basis for their sound creation. For instance, Dimension Pro and Rapture from Cakewalk both have extensive sample-playback capabilities and the ARIA Engine is a dedicated sample playback tool intended to create sounds of the highest quality.

The market is, of course, always changing with new products regularly being released, so please don't let this brief summary suggest that there are only a few samplers available.

Understanding Sampling and Synthesis

Synthesis and sound creation comes in many forms.

A key to creating sound with any synthesizer is to understand the strengths and the limitations of your chosen tool (and by implication, the strengths and limitations of the synthesis techniques available on your chosen synthesizer). There's no point in using the wrong tool for a job—rather than force a synth to do what it's not designed to do, you will usually find it better to change tools.

When it comes to synthesizers, there are three main choices for creating sounds:

sample-based synthesis

analog synthesis, and

frequency modulation synthesis.

Let's briefly look at these in turn.

Sample-Based Synthesis

Sample-based synthesis offers the facility to recreate the sound of real instruments with a high degree of sonic fidelity and musical control. It then allows the musician to go further and create super-real instruments in that the sampled instrument never goes out of tune, never has a bad microphone located in the wrong position, and can have effects (such as EQ and compression) built-in to the sound.

Beyond that, sample-based synthesis can take raw waveforms, then twist and manipulate them in the digital domain and by applying a range of analog-like tools and techniques to create wholly new sounds. And of course, sample-based synthesis can be used to create and control loops as well as other drum and percussion elements.

There are, however, times when analog or FM synthesis is a better option. Just one example of this comes when you are using analog or FM samples where you will lose the flexibility and nuance of using the original source.

Analog Synthesis

Analog, or subtractive, synthesis is used in many classic synthesizers such as the Moogs (Mini Moog, Moog Modular, Taurus, Little Phatty), the Oberheim synths (OB-8, OB-X), the Rolands (SH-1000, Juno-6, Jupiter-4, and Jupiter-8), Prophet-5 from Sequential Circuits, and the ARPs (ARP 2500, Arp 2600).

You will hear people describe analog synthesizer sounds as warm, rich, simple, and powerful. They are all of these and more. You will have heard these techniques used on many recordings and will immediately recognize the sounds they produce.

Analog synthesis works by taking a sound source and shaping it with a filter. This is a simple but powerful technique which allows you to create the warm, rich, energetic, and commanding sounds, and with some care you will find tones with detail, complexity, and delicacy.

Frequency Modulation Synthesis

Frequency modulation synthesis (or FM synthesis as it is more often called) adopts a completely different approach to creating sounds when compared with either analog or sample-based synthesis.

Instead of starting with a bright sound and making it dull and/or shaping it with a filter, FM synthesis starts with a simple wave—usually a sine wave—and distorts it (usually with another sine wave). The interaction between the two (or more) waves creates and shapes the sound over time.

The sound from FM synthesis is often characterized as being harsh or cold. It can be, but more significantly, FM synthesis is capable of creating a broad, detailed, highly nuanced range of tones which can be controlled with an exceptional level of detail.

In some ways FM seems complex since it offers an infinite range of possibilities and permutations. However it is this range of choices which gives the broad array of tonal nuances.

Analog synthesis and frequency modulation synthesis are covered in separate books which are discussed in greater detail in the Further Resources section.

The Featured Samplers

This book is about creating sounds with sample-based synthesis techniques.

To make the link from the page to the synthesizer, the book shows the practical application of the techniques with two sample-based synthesizers: Battery and Kontakt, both of which were created by Native Instruments. These synthesizers are used for illustration: you don't need to own them, but it might help. Even if you don't own the synthesizers, the principles set out in this book can be applied to virtually any other sample-based synthesizer.

Battery

Battery (see Figure 1) is a dedicated drum sampler.

Figure 1 Battery.

Battery has a 128 cell matrix, and up to 128 samples can be loaded into each cell (in other words, a single instance of Battery can deploy over 16,000 samples). This matrix makes it easy to control groups of samples (either rows or columns can be readily selected). As well as its organization capabilities, Battery also includes a wide range of tools to further control, manipulate, and enhance your sounds.

While Battery is intended as a dedicated drum sample player, you can load any sample into the tool, so you could use it as a conventional sampler (for instance, there's nothing to stop you loading a group of piano samples and then playing Battery as a conventional sampled piano).

Kontakt

Kontakt (see Figure 2) is one of the most powerful samplers available on the market today and is one of the most widely used.

Figure 2 Kontakt.

It would take me the rest of the book (and several volumes after that) to try to describe all of Kontakt's features, so let me just say, that it's very powerful, it probably does everything you could ever need it to (in a sampling context), and there's a reason why so many professional musicians use it, and you can extrapolate from there.

Format of this Book

This book is only available in electronic form and has been designed, and optimized, to be read on electronic reading devices such as the Kindle, nook, and iPad, as well as the desktop version of any reading tools.

The graphics have been have been optimized for the intended reading medium and are in color. These color images render on a grayscale eInk screen without any problems. The layout (including the images) has been designed to work on a wide range of screens including a standard 6" eInk screen, a phone-sized screen, and an iPad-sized screen.

There are, of course, a few limitations with this format—perhaps the biggest relates to the graphics. Many electronic readers can only deal with graphics of a certain file size. Accordingly the graphics have been squeezed to extremes. This is not a problem for the line drawings (such as the images of the envelopes), but it is more of a challenge for the screenshots (which are reduced in terms of pixel size as well as file size). With this extreme compression, some of the very fine details simply do not render as clearly as I would like on an eInk screen.

You will also see that I have included clickable links to external websites (which will require an internet connection). To differentiate between the internal and external links, external links have a small image to their right, thus.

Navigating this Book

The clickable links in the book are intended to make it easier for you to jump to a specific point. This means that you can go straight to the section you are looking for, and if you don't like the order in which I've put the chapters, then you can read them in the order you want to and hopefully there are links to any earlier material that is necessary to get a full understanding in the later chapters.

And on the subject of the order that the information is presented, the book is structured as follows:

Chapter 1: Getting Started with Sample-Based Synthesis. Chapter one introduces you to the basic principles of sample-based synthesis and the concepts we look at through the book.

Chapter 2: Capturing the Raw Samples. Chapter two looks how the basic elements of sample-based synthesis—the waveforms—are captured.

Chapter 3: Deploying Samples. Chapter three looks at how the raw samples can be deployed in a sampler in order to start constructing sound sets.

Chapter 4: Controlling Samples and Loops. Chapter four looks at how the waves and loops which have been deployed in a sampler can be controlled by the musician.

Chapter 5: From Samples to Sounds. Chapter five looks at the practicalities of creating sounds using sample-based synthesis techniques.

Chapter 6: Filters. Chapter six looks at filters which can shape the sound both to smooth the tone and to radically alter the sound you hear.

Chapter 7: Modulation: Real-Time and Player Controls. Chapter seven looks at modulation, the process by which all of the elements are controlled to shape the sound and looks at the main tools (including envelopes and LFOs) which are available to the musician.

Chapter 8: FX. Chapter eight looks at the effects units that are available to the musician and how they can be used to create and shape sound.

Further Resources. If you're wondering what to do after you've read this book, then the further resources section will give you some ideas, including directing you to some free books and videos.

OK. That's enough introduction: let's get on with the music.

Chapter 1: Getting Started with Sample-Based Synthesis

There are three factors to consider when contemplating sample-based synthesis:

First, there is the sample, or more likely, samples.

Second, there is the sampler—the tool in which the samples are deployed.

Third, there are the decisions you make about how the samples are treated within the sampler.

That's not everything... but it's a good basis from which to start thinking about sample-based synthesis, and in that context, let's start be looking at samples and sampling.

What is Sampling?

Sampling is the digital recording of sound so that:

The sound can be reproduced.

The sound can be manipulated.

And that is it: sampling is about creating a digital waveform. The source from which the waveform is created can be an acoustic sound, an electric sound, or a digital sound—you just need to come up with a waveform that you can deploy and process.

The Benefits of Sampling

Sampling—and more to the point, the deployment of samples—brings huge benefits for the musician. Here are some of the main benefits:

With samples you can reproduce any instrument that you want in any environment. Once you have a set of samples, you do not need to have that instrument or a competent player of that instrument—you just need to load up the samples and go.

Beyond simply having one or two instruments, with samples you can have every instrument that you want and you can have more instruments than would fit in your studio. Not just that, but you can carry a whole orchestra (and more) on your laptop.

To get the sound you want, all you need to do is load up the appropriate samples and you will get a consistent, perfect, sound. You are not constrained by the acoustics of your location and the recording gear that is available.

Samples are ultimately controllable.

Samples never go out of tune.

Samples never get bored, demand a pay rise, go sick, or have a better job offer. If you're a composer writing a piece, you can keep auditioning parts without worrying about your musicians.

Unusual instruments (some with non-standard tunings) can be played across a conventional keyboard.

What is a Sample?

The term "sample" can be applied to mean several different things. In all cases, these definitions relate to the sample and not to how it is deployed.

Single Cycle Waveform

The simplest form of sample is a single cycle waveform (see Figure 1.1).

Figure 1.1 A single cycle sine wave.

A single cycle wave is very short—for instance, a single cycle wave pitched at 440Hz would complete its cycle in just over 0.002 seconds—and therefore this form of wave is usually looped. Many synthesizers include a range of single cycle waves.

Single Shot

A single shot is what most people mean when they talk about a sample. It is a sample of a single note or a single drum strike (see Figure 1.2). The sample starts in silence, the single shot audio plays, and then there is silence again.

Figure 1.2 A single shot stereo snare drum wave opened in a wave editor.

As well as single notes and single drum strike, there are many other forms of single shot samples. For instance, there are phrases (the clichéd James Brown "oh yeah" sample), birds squawking, or machinery making a noise.

Multi-Sample/Sample Set

There is no such thing as a multi-sample, but you will hear them referred to often (and indeed, I will talk about them extensively in this book).

However, what you might find is a collection of (usually single shot) samples which have been organized to work together, usually to create an instrument. So for instance, if you have a collection of bass guitar samples that they could be called a multi-sample or a sample set.

Loop

A loop is a sample of a phrase (see Figure 1.3). That phrase could be a drum pattern lasting for a bar, it could be a bass figure played for several bars. The content is irrelevant—what matters is that there is a pattern that can be deployed, and to make the loop function as a loop, the sample must be capable of being repeated (without any audio artifacts when the sample ends and restarts).

Figure 1.3 A stereo drum loop opened in a wave editor.

If you want an example of a loop, turn on the radio—virtually every song you hear will use loops.

Understanding Digital Audio

To get our sounds into a form that they can be deployed and manipulated in a sampler, we need digital files. In the case of a real instrument, this means the sound has to be converted from analog into digital. In the case of a digital instrument, the sound needs to be captured in a form that can be deployed.

Whether you're starting in the analog or the digital realm, there are several factors that will affect the quality of your sound. These elements will affect the quality of the samples—in other words the raw material that you feed into your sampler.

Analog to Digital Conversion and Digital to Analog Conversion

Audio is analog. Your ears are analog devices. All real instruments (guitars, drums, pianos, and so on) produce analog sound. This analog sound interfaces perfectly with your analog ears.

However, this analog sound is no use in the world of digital, and sampling is digital.

Therefore we need to convert the analog audio into a form that a sampler can consume. The device that does that is an analog-to-digital converter (ADC as it is often called). In order to listen to the edited content which will be in the digital domain, the sound needs to be converted into a form that is compatible with the human ear. The device to do that is called a digital-to-analog converter (DAC).

Most computers have audio inputs and outputs: these are ADCs and DACs. However, the quality of a built-in sound card is usually not sufficiently high for audio work and, in particular, the quality of these cards would be too low for sampling. The best option for sampling is a specialized converter—these come as dedicated sound cards which will fit into your computer or as external models (which will then require a digital interface on your computer).

You should get the most expensive ADC you can afford or beg/borrow a more expensive model for a sampling session.

The quality of the DAC has some relevance when sampling. When you play your samples, what you hear—irrespective of the quality of the sample you have captured—will only be as good as the quality of DAC that converts your sample. Generally any high quality DAC should be suitable; however, if you are trying to capture very delicate timbres, then you might want to think about a higher quality converter.

There are three technical aspects that affect the quality of results achieved by your ADC:

The quality of the hardware and the algorithms the converter uses. As a basic rule of thumb, higher quality converters cost more.

The bit depth.

The sample rate.

We'll look at the bit depth and the sample rate in the moment, but before we get there, be aware that there are other factors that affect the quality of your sampling:

Your talent and your understanding of the gear you are using.

The quality of your source material—an out-of-tune upright piano is never going to sound like a Steinway just because it has been sampled. The best you can hope for is that an out-of-tune upright piano will sound like an upright piano (assuming you can edit out the pitch issues).

The quality of the player. You might think that all you need to do is strike a key or bang a drum to get a sample—and at a very basic level, that is all you need—however, to find the tonal nuances on any instrument, and to reproduce them with consistency and quality requires a talented player.

The environment in which a sample is captured will obviously have a significant effect. Poor acoustics and background noise can have much more affect than is the case when recording a part for inclusion in a track.

When you take an analog waveform and digitize it, you are essentially slicing and dicing it, in other words, you are cutting it vertically and horizontally:

On the vertical axis, you have degrees of loudness.

On the horizontal axis, you have time.

Let's look at the two factors that determine the quality of those cuts.

Bit Depth

Digital sounds have a finite dynamic range. The bit depth determines how many degrees of loudness are available between zero and this maximum loudness.

At its most basic, a sample could be on or off like a light switch. However, that would give a very basic sample.

Earlier samplers used a bit depth of 8 or 12. Today the minimum standard is 16 bit with 24 bit being the norm.

16-bit means there are 65,536 (2^16) possible loudness values for any one sample.

24-bit gives 16,777,216 (2^24) possible values for any one sample.

For creative reasons, people sometimes like to use lower bit depths (even if their samples were taken at a higher bit depth). 8-bit (which has 2^8, in other words 256 degrees of loudness) can give a much rougher sound, reminiscent of the earlier samplers.

You can play a higher bit depth sample at a lower bit depth—you cannot increase the quality of a sample by simply increasing the bit depth: for that you would need to re-sample the source material.

Sample Rate

The other aspect to sampling is the sample rate. The sample rate determines how frequently the level of the incoming wave is scanned.

A sampling rate of 44.1 kHz would mean that the ADC captures 44,100 slices of the input wave each second.

44.1 kHz is the rate used in CDs and is the minimum value for high quality samples.

Combined Effect of Bit Depth and Sample Rate

The combination of bit depth and sample rate determine the quality of a sample. The two work together like a mesh overlaid on top of the wave—the finer the mesh (see Figure 1.4) the higher the fidelity of the sampled wave.

Figure 1.4 The fidelity of a sample is directly related to its bit depth and sample rate.

We'll look in more at how to capture quality in more detail in the next chapter.

The Downside of Digital

Sampling offers huge advantages, but as you would expect, for every gain there is a downside and there are some downsides to digital—or rather, there are some issues with digital that need to be born in mind while manipulating digital audio.

The real downside of sampling is that it is not the same as the source. Taking an instrument, capturing it digitally, and then playing those digital sounds may give you a stunning result, but it will never—can never—be better than listening to the source material.

To get closer to the source material, a lot of samples are needed. As in a lot. Ideally you will have one sample per note (samples played out of their key range sound unnatural very quickly... which can be great for effects) and you will need several layers of samples for different volumes (perhaps 16 as a minimum for a detailed acoustic recreation). On top of that, you will want specific playing techniques to be recorded. Pretty soon, you will have a lot of samples...

Digital audio degrades. Now, that sounds like quite a bold statement, so let me explain. To play a sample at a pitch that is different from the pitch at which it is captured, a sampler will in effect play the sample faster or slower.

If the sample is played faster (and so it sounds at a higher pitch than the source), slices are discarded.

If the sample is played slower (and so it sounds at a lower pitch than the source), then slices are created by interpolation.

The reason that samples are created or discarded goes back to the sample rate (in other words, the number of slices per second). A sample will have one sample rate—it is technically unfeasible to have a different sample rate for each sample that is being played. The different sample rate would arise by having more samples (when the sample is played faster) or fewer samples (when the sample is played slower).

The discarding of slices and creation of new slices by interpolation leads to a drop in the sonic fidelity of the sample, hence a drop in quality, and the greater the pitch is changed, the more the quality is degraded.

Samples can also be degraded when they are edited, especially if they are re-pitched (which is necessary if the instrument wasn't perfectly in tune when the sample was captured).

Another form of digital distortion is aliasing. Aliasing is a flaw in the reproduction of a digital waveform—it is not a problem with the sampling process. It is a problem that has largely been fixed by clever sampler design, however, you may still encounter it with older gear and if your sampler offers it as an option (some people feel that the crunchiness it can give a sound imbues it with a certain vintage vibe).

To understand aliasing, you need to understand the Nyquist frequency. The Nyquist frequency is one-half of the sampling rate, so if the sampling rate is 22,000 Hz (which is a very low rate by today's standards), then the Nyquist frequency will be 11,000 Hz which is well within the audio spectrum. Any sound (or element of a sound) which has its frequency above the Nyquist frequency will lead to digital distortion or aliasing.

There is no aliasing inherent in any sample: it is a sample deployment issue and becomes more noticeable when samples are tuned up in pitch, since any increase in pitch moves closer to the Nyquist frequency.

What is a Sampler?

In the days of hardware it was easy to define a sampler—the hardware device was the sampler. If you wanted to sample an instrument, you plugged a microphone into the sampler and hit the capture button.

In the days of software, it is something of a misnomer to call samplers, samplers. The devices don't actually sample. In the context of software, what we call a sampler, is actually a sample playback device. However, everyone (myself included) calls these things samplers and so I shall use that term in this book.

So what is a sampler?

A sampler is a tool that performs several functions:

It arranges samples. It takes raw waves and arranges them so that each sample relates to the correct pitch and velocity (in other words, the loudness) range. Clearly this is highly important if you want to play a melody or if pitch information is at all important.

The sampler receives incoming MIDI data (pitch, velocity, player controls, and so on) and transmits that data to the samples. In other words, it makes the connection between the note played and the sound heard.

Samples can be edited. Depending on the particular sampler, this editing could involve trimming the start and/or end point of a sample, reversing the wave, or more detailed changes.

The sampler also allows for the samples to be processed using synthesis tools (such as envelopes and filters). Most samplers also include an array of effects units to further sweeten, process, and enhance the sounds.

In short, a sampler is what takes the waves, and makes a playable and controllable instrument. Some samplers have more features, some fewer, but in essence they all perform a similar function.

From Theory to Practice

So now we've got an idea about sampling and digital audio. Let's move on and look at how to capture samples.

Chapter 2: Capturing the Raw Samples

In the early days of sampling it was not uncommon to capture a sample and use it immediately. One of the main reasons for this was that, from a practical perspective, you couldn't capture more than a few samples, and so the editing and deployment phase was comparatively straightforward.

However, with disk streaming giving the musician a virtually unlimited size of sample set, the number of samples that can be captured has grown exponentially, and as a result, sample capture has become a whole dedicated operation. Added to which, now that the quality of sample reproduction is so high, any flaws in the raw material will be easily detected and hence the conditions in which samples must be captured are more critical than is the case for conventional audio recording.

This chapter looks at the process for capturing samples and some of the key considerations during the capture process. One key factor—but by no means the only factor—in determining what and how you sample, is how the samples will be deployed. Very often, you need to know what you're going to do with the samples before you can capture the samples. For instance, if you're trying to create an acoustic piano with a high degree of sonic fidelity, then that will have an impact on the number of samples you need to capture, the acoustic environment within which you will capture the samples, and the quality of gear you will need.

In the next chapter (Chapter 3: Deploying Samples) we will look at how samples can be arranged to create instruments. Since the deployment of your samples may impact and inform decisions you make at the sample capture stage, you might want to jump ahead and check out that chapter and then come back and read about the sampling process. Equally, if you have no interest in creating your own samples, then you might want to jump ahead.

Preparation

Once a sample is captured, you can edit it. However, you cannot change the underlying sample.

So for instance, if you sample the wrong timbre, you cannot change that error through editing. If you do not sample the full sustain period of a note, you may be able to work-around the shortcoming, but it will be a work-around, which, by its nature will be of a lower quality than if you got the right sample in the first place. Equally, if you don't capture a specific sample (or set of samples), then you won't be able to deploy them.

I hope it goes without saying that if you want to capture the sound of a specific instrument, in a specific acoustic environment, as heard by specific microphones that you need to bring together that instrument in that environment, and those microphones.

So there are many logistical challenges (getting a player, the instrument, the sampling venue, and the gear together on the same day) and there are other decisions, specifically what you are going to capture. These decision all need to be made before you hit the record button.

Environment

At its most basic level, the process to capture samples is no different to recording any acoustic performance, such as a voice.

However, sampling is different.

When you record a voice (as an example) that voice will almost always be heard with some sort of backing. Therefore if there are any unwanted elements captured with the recording—perhaps some background hiss, some buzzing, or maybe an external noise bleeding through into the microphone—then those elements will usually be masked by the backing track.

With sampling, there is a greater likelihood that an instrument will be heard on its own. Equally, there are very frequently pieces of music where there is no masking, for instance, you might have a piano sample which plays with accompaniment from drums and a bass. In this case there could be limited masking due when the other instruments cover a different frequency range.

And of course, the sample will always be heard in its raw, unadorned state when a musician is auditioning sounds.

But there's more to this. If you record a singer, you might get (say) a low-level buzz, which can be masked. Now think about a piano if you capture a buzz when you sample. When a piano is played the musician will typically play several notes at once, and if the sustain pedal is used, then maybe 10 or 20 notes could be heard at once. If each sample includes a low-level buzz, then that buzz will be multiplied 10 or 20 times, thereby becoming far more prominent.

In a studio environment, you can expect silence (or at least a lack of extraneous noises—there should be no traffic noise, no buzzing lights, and so on). However, if you are trying to record a specific acoustic environment, such as a concert hall, then you may have more of a challenge. The two biggest issues you are likely to have are:

knowing there's a problem, and

if you do find a problem, figuring what the source of the noise is.

To know that there's a problem—in other words, to know that a microphone is picking up unwanted noise—is hard work. Unfortunately, there's usually only one solution—listen to each microphone individually after it has been set.

What to Capture

In the next chapter (Chapter 3: Deploying Samples) we'll look at key ranges and velocity layers in more detail, but let me give you the quick version since it has an impact on how many samples you take.

Key Range to Capture

When a sample is deployed, it is usually playable on its corresponding key. So a middle-C sample should sound when you play middle-C on a keyboard. However, this logic rather presupposes that you have (at least) one sample per key. It is not necessary to have one sample per key and excellent results can be achieved without one sample per key.

However, if you have fewer samples than keys, then you need to fill the gaps (if you are trying to recreate a chromatic instrument) and to fill the gaps, samples can be played over a wider range of keys (see Figure 2.1).

Figure 2.1 Playing a sample over a key range—the dot shows the center key and the shaded area the range over which it is played (in this illustration, each sample is played over plus or minus two semitones).

Playing a sample over a wide key range leads to two main difficulties:

First, the original sample will be tuned out of a its normal range and this retuning may make the sample sound unnatural.

Second—particularly when individual samples are playable over a wide range—the points at which one sample range ends and the next range begins are noticeable.

Depending on the source material and the key range, these two issues may matter more or less. However, as the range over which any sample is played increases, the sound of the whole sound set will be degraded.

The way to ensure highest fidelity is to have (at least) one sample per key. This has consequences—primarily: it takes longer to capture the sample set, it is harder to maintain consistency when capturing samples, the edit process for the samples will take longer, and the file size of the sample set can become large. But as I said, it is the way to ensure the highest fidelity.

Velocity Layers to Capture

The other aspect is velocity. MIDI velocity is the loudness with which a note is struck, and according to the MIDI specification, you can have up to 128 layers of loudness.

When you strike an acoustic instrument with differing levels of force, there are two changes:

the volume increases (or decreases), and

the tone changes—generally a note will be brighter if it is struck with more force.

We can use velocity layers to mimic these changes, and as with key range, the more velocity layers that can be included, the greater the fidelity of reproduction.

Clearly, capturing 128 velocity layers is quite a logistical challenge (not to mention a lot of editing work, especially for a 88 note piano where you may be capturing different pedal options too).

The number of layers you need in practice is very much dependent on the source material. With drums you may get away with three layers (and I do mean get away—it's not always a pretty solution). With a piano, you'll probably want at least 16 layers to get good audio fidelity.

Once you have addressed the issues relating to key range and velocity layers, there are two further aspects to be considered:

how much to capture—in other words, which performance elements should be caught, and

what to capture—in other words, which sonic elements should be captured.

In both cases—as with key ranges and velocity layers—it is always better to capture too much, if you can. You can always use a sub-set of what you capture—you don't have to edit every sample you capture and you don't have to deploy every sample in the sample sets you build. However, if you haven't captured a specific element, then you have a gap.

Performance Elements to Capture

If you're sampling a piano, you have a limited number of decisions to be made about what to sample: each key produces one note and the only variable is the power (velocity) with which the key is struck, which in turn affects the volume and the tone of the note.

If you're a piano player (or if you know anything about the piano) you'll realize that this analysis is simplistic—there are more issues here, for instance whether the sustain pedal, the sostenuto pedal (if there is one), or the soft pedal is depressed. However, for the basic tone, it is possible to capture the sound with some consistency.

The real difficulty when capturing a piano is accessing the broad range of loudness levels (and accessing these levels with some degree of consistency between the samples) and to address this, some samplists use mechanical devices to strike the keys in order to smooth the response.

However, when you start looking at an instrument like the bass guitar you find—despite the limited pitch range when compared to an 88-key piano—a whole range of different tones are available within one instrument. For instance:

The player may use fingers or a pick. Equally the player may slap or pop the strings instead of picking. Each choice gives a different tone and different players (having different physical characteristics and different playing styles) will create different tones. Equally, there are different tones within a single style, for instance, if the player is using a pick, the downstrokes will likely sound different to the upstrokes.

The player can hit the string in different places—closer to the neck gives a warmer, thicker tone, while closer to the bridge gives a thinner tone.

There is often more than one pickup on a bass—each will generate a different tone.

And of course, in addition to sounds created with conventional means, there are a whole range of scrapes, slaps, slides, and so on which are very much dependent on an individual player's style.

When it comes to sampling, it makes a lot of sense to create sound sets with consistent tones. So for instance, you might want to sample a bass guitar at all pitches and at a range of velocities, and if you did so you would want the player to retain a consistent playing position on the strings and a consistent plucking style (for instance, using a plectrum for all samples).

Once the first sample set has been completed, then you could move on and perhaps use a different position on the string or a different plucking style.

Sonic Elements to Capture

With hardware samplers, there was little thought about what was captured. You had a single input, or if you were lucky a stereo input; you would then plug in your instrument, or a microphone, and sample. And whatever you sampled was it.

But in the world of software synthesizers, and in a place where the sampling process and sample playback are divorced, there are many new options, and those new options focus much more on how a sample will be deployed.

Today, when you come to sample, you will probably not be recording a single sample, but instead, you may be recording a whole batch of samples which are intended to be deployed simultaneously. Let me give you a real-world example—say you want to sample a bass guitar.

For the moment, ignore the varying pitches, the playing styles, and the different loudness/tone combinations. Instead, focus on what can be recorded, and what might be captured if you had a player in a studio to record a part for a track. For a conventional recording, there are several aspects that might be captured:

First, the direct sound of the bass—in other words, the unprocessed sound that comes from the instrument—could be captured. If a played part was recorded, then, to subsequently change the tone, this sound could be re-amped (in other words, the recorded track could be played into an amplifier and that sound can then be recorded). However, re-amping takes away the input of the player who will change their performance based on the sound they are actually creating. Accordingly, if you were recording a player, you would also want to record the sound coming out from their speaker cabinet.

The most straightforward way to record the audio signal from a speaker cabinet is to place a microphone pointing directly at the middle of a speaker.

While pointing a microphone directly at a speaker is the most direct way to record the sound, it may not give the best tone—typically, it will give a bright tone. As a result often a microphone will be placed towards the edge of the speaker and at an angle—this will usually give a slightly mellower tone.

A third microphone is sometimes added, perhaps 12 to 36 inches away, but pointed towards the speaker. This can give more of the "sound" without being affected by the nuances of pointing a microphone at different areas of a speaker.

Often speaker cabinets have several speakers. Sometimes all the speakers are the same size, in which case the "best" sounding speaker is usually recorded. However, sometimes there are different sized speakers (to reproduce the full audio spectrum). If there are several speakers, then close microphones may be placed in front of several speakers.

It is also common practice to record a cabinet from the back. This gives a thicker sound and is less prone to being influenced by individual speakers.

In addition to the positioning, the other aspect to take into account is the microphones themselves. Different microphones sound different and capture different aspects of a sound. As a result it is not uncommon to have several microphones recording one source—the sound from each microphone can then be blended to get the right tone.

As you can see, perhaps eight elements could be captured to record one part (and that's before any ambience or reverberation is recorded). This is not an unusual set-up. However, if you are recording (as in this case) a bass player in a studio, then you will generally know the sound you want—it will be the sound that works within the context of the track for which the part is being recorded.

Knowing that context, you would be able to make decisions, so for instance, if the sound captured by the microphone pointing directly at the speaker was too bright, then you might not use that, instead, relying on the edge microphone. Knowing the context you might be able to record with one or two microphones and get the exact sound that you are after.

However, when you are sampling, you cannot know the context in which the sound will be deployed. Indeed, if you are capturing sounds for commercial distribution, you may never meet the person who uses the samples. Accordingly, when you're sampling, you need to capture several sources if you want to give flexibility to the producer who will be using your samples.

The range of samples that is needed will vary depending on the instrument and the level of control you want to give.

For instance, if you are sampling a bass guitar, then you might want to simply record the direct sound and a mic'd cabinet giving what you regard as being the best sound. For a bass guitar, that could be a reasonable decision (especially if you are creating a sample set with a wide range of tonal nuances).

By contrast, if you are recording a drum kit, then you are likely to want to give the producer a range of tonal controls. Typically with a drum kit, you might have a close microphone for each drum, an overhead pair, some room microphones, and some pressure zone microphones (PZM) to pick up the ambience.

You may not want to offer all of these options for a drum kit, but simply offering a dry close-mic'd kit is unlikely to give satisfactory results for anyone looking to reproduce the sparkle of a real drummer. Equally, offering a drum kit only captured by the room microphones is also unlikely to give a satisfactory result. These microphones will have picked up the ambience and that ambience cannot be removed—you can reduce its effect, but that would be likely to have a detrimental effect on the sound.

An orchestra could present other challenges. Typically you might want to close-mic each of the key instruments (and/or groups of instruments). This would give a fairly dry sound which would have similar limitations to a dry drum kit. To get some more ambience you might record the whole orchestra (or rather position the microphones to record the whole orchestra, but then record single instruments and/or sections), perhaps using a Decca Tree or similar microphone arrangement. However, to then get something of the concert hall, you would need to record in a concert hall (or at least a reasonably large space with a sympathetic reverberation—an aircraft hangar may not give the smoothness of tone appropriate to the textures of orchestral instruments).

We'll look at how to deploy these multi-position sound sets in the next chapter (in the section Deploying Multi-Position Sound Sets).

Labeling Your Samples

You will have figured by now that you might end up with a lot of samples, very quickly.

If you're going to have a realistic chance of creating coherent instruments, then you need to adopt a naming convention for your individual wave files. From your wave file name, you should be able to immediately identify:

The instrument sampled.

The playing technique/configuration (so neck pickup, fingered).

The microphone placement.

The pitch.

The velocity.

You don't need a full text naming system—you simply need to be able to identify waves so that you can identify the correct waves to create sample sets. However, one aspect you might want to pay attention to is how you label the pitch and the velocity information. Some samplers will automatically map uploaded samples to pitch and velocity ranges based on their names. If you adopt a naming convention that accords with how your sampler works, then you will save yourself a lot of work when you start creating sound sets in your sampler.

Sampling Hardware

There's a very simple principle in sampling—you want to do as little editing as possible. Every edit degrades the signal, and so the less your sample needs to be edited, the higher the fidelity of your resulting sound set will be.

The key to minimizing edits is maximizing the quality of the sample. There is no one thing you should do to maximize quality. Instead, you need to look at your whole audio chain, and while it may sound a cliché, your sampled sound is really only as good as the weakest link.

I've already talked about the environment in which the sample will be captured. Clearly, you need silence—as in a lack of any unwanted noises—so that no extraneous noises will be captured (this is particularly the case if you are using a multi-microphone set-up to capture several different sonic elements). And to accompany your sterile environment, you want a perfect audio source—there's no point in having the most expensive gear you can find if you're using a collapsing piano that isn't even tuned and which produces a strange knocking sound every time a key is pressed or released.

Once you've sorted the sound source and the environment, the first link in the chain—assuming you're capturing an acoustic source—is the microphone.

Microphone

You don't necessarily need the most expensive microphone you can get hold of—you need the microphone, or more likely microphones, that are right for the job. In other words, you need a high quality microphone which will capture the sound you want.

Not only should the microphone be the most appropriate for the task, it should be used in the correct manner. That is to say it should be located in the optimum position and the levels should be correctly set.

Cables

Between the microphone and the pre-amp is a piece of wire... a cable. These cables should be of a quality that is comparable with your microphone and pre-amp. The cable should also not have suffered extreme physical abuse—broken wires, loose connections, and the like will simply degrade the signal transmitted along the cable thereby reducing the quality of your sample.

Seriously. Cables are pretty much the cheapest element in the sampling set-up. Go and buy a good one before you start.

Pre-Amp

The pre-amp increases the level of the signal coming out of the microphone so that it is at an appropriate level for converting from analog to digital.

Pre-amps come in two forms. First, free-standing units, the output from which is then sent to the analog-to-digital converter. Second, built-in to the digital-to-analog converter. Free-standing units allow you to select the highest quality pre-amp that you can find, however, they lack the convenience of integrated units.

Any noise created in the microphone or picked up by the cable will be amplified by the pre-amp, therefore it is preferable to find a pre-amp that doesn't add any more noise and doesn't add any color to the signal.

Analog-to-Digital Converter

The final step in the chain is the analog-to-digital converter and naturally you should use the highest quality unit you can (legally) lay your hands on.

The factors determining the quality of result of analog to digital conversion were discussed in the last chapter in the section Analog to Digital Conversion and Digital to Analog Conversion.

The one element that is not discussed is the input level. You will usually be recording only one note of one instrument at a time. Accordingly it should be possible to set an audio input level. Conventionally, levels are set fairly high, however, since digital distortion will ruin a sample, provided you are using a high quality analog-to-digital converter at a high bit depth, you can afford to pull the level back by a small amount to ensure there is no distortion.

Recording the Digital Audio

Once the digital waveform is streaming into your computer, you need to record it. For this, there is a range of software you can choose.

At the low price end, you could use a wave editor like Audacity. This is free and good quality, but basic.

Moving up in price, a digital audio workstation such as Logic, Pro Tools, SONAR, or Cubase will give excellent results.

These packages can all record multiple streams of audio, however, they may not necessarily be designed for the detailed level of editing that may be necessary and so you may want to consider a dedicated digital audio editor package such as Sound Forge or WaveLab.

Whether you go for a digital audio workstation product or a dedicated audio editor, you should normally find that the quality is good enough (however you may find that the audio editor offers a higher quality in some areas, for instance offering higher sample rates). Subject to minor differences, the main determinant as to which package you use is personal taste—go with the software that you are comfortable using and with which you are familiar.

As with anything digital, the market is always changing, so please check out the latest tools before making any choice.

When it comes to capturing the digital audio, whichever tool you use, there are a few simple rules:

First, capture the wave without any processing. For instance, don't use a software compressor to tame the level. (And by the way, you don't do this for two reasons—first, because any processing degrades the wave, but second, and far more importantly, because it achieves nothing. If the wave needs a compressor to tame its level it would already have been distorted at the analog-to-digital converter, in which case the level is already too high).

Second, capture the audio in an uncompressed format (which means WAV format if you're using a Windows machine and AIFF if you're using Apple). Run away from anything compressed (such as MP3 format).

As long as you've got these basic issues sorted, it's really just a case of hitting the record button.

Editing

The final step in capturing samples is editing. But before we start editing, there are a few things to remember:

You don't have to edit everything. You can edit a sub-set of your samples. We're dealing with digital files—your full sample set will still be there if you want to go back to it.

High quality samples can be degraded. You can degrade the samples in order to reduce file size or to create a different sound.

Low quality samples cannot have their quality increased. You cannot add back data that was not present when the sample was captured. Equally, you cannot edit away audio artifacts and poorly placed microphones.

Unless you are happy with the samples that are available, there's little point in editing your audio files. You will be much better served by going back and recapturing the samples before you move on.

Assuming you are happy with the samples, then let's move on and think about some of the edits you might undertake.

Trim

The most frequent edit that you will make is to trim an audio file.

When you capture a sample, you will start recording before the sample begins and you will stop recording after the sample has decayed to zero. This means that you will have some excess audio at the start and end of the sample which should be trimmed (see Figure 2.2).

Figure 2.2 The noise at the beginning of the sample (ringed in red) needs to be trimmed.

The aim with trimming is to cut off the excess, but to leave the material you were trying to capture. This means that when your sample is triggered it will begin immediately (and not after the excess material has played).

Trimming is a precision job—you need to be able to differentiate between the wave you were trying to capture and any background noise. If you cut too far, then you will truncate the sound you want.

When you do cut, one thing to be cautious about is to ensure that you cut at a zero crossing point—in other words, you cut when the wave crosses the horizontal zero axis in your wave editor (see Figure 2.3). If you don't cut at a zero crossing point, then when the wave plays it will jump from zero to the level at which you cut the wave. You will probably hear this as a click.

Figure 2.3 A zero crossing point—one of many in this wave—is marked by the vertical red line.

It can be a challenge to find the end of a wave since the sound will not fade to silence, but will probably fade to the background noise. Some people use an audio editor to fade the end of their samples to silence. This approach is sure to avoid any unwanted clicks; however, arguably it does degrade the quality of the sample. Depending on the material that you are editing, you will have to decide whether to fade (and then trim) or simply to trim at a zero crossing point.

The other challenge with editing samples arises when you have captured several different sonic elements (for instance, when you have recorded a direct bass, a close mic'd cabinet, a rear mic'd cabinet, as well as some ambience). The challenge here is to ensure that—through your editing—you are not changing the relative time of any of the samples. This can happen if you trim more off the start of one sample than another.

There are two reasons why you want to avoid changing the timing of samples relative to one another. First, you may create phase shifts—in other words, one wave may partially cancel another wave. Second, you may create sonic logic problems—if you hear the reverb before a drum is struck, that just won't sound right.

The way to avoid these problems is to synchronize the waves and then ensure that an equal amount is trimmed from the start of every wave, and to do this, you will find one of the dedicated wave editors is an invaluable tool.

Tune

When you listen back to your samples you may find that there are tuning issues.

The best way to address these issues is to go back, tune the source, and re-capture the offending sample. However, this is not always possible and if it is not possible, then you should retune the sample in your audio editor.

As you will hear me say on many occasions, if you can avoid processing your samples you should, and this holds true for retuning since any retuning will degrade the quality of your samples.

Level

Occasionally you may want to tweak the level of your sample—perhaps one or two samples just don't sit quite right within the context of a sample set. If you can avoid this, avoid it, but if you can't, a small tweak here or there shouldn't cause noticeable results.

Some people normalize their samples—in other words, they set their sample so that the loudest peak is just below the maximum audio level (see Figure 2.4). This means loud samples are loud and quiet samples are loud. The variation in volume is then effected through the sampler.

Figure 2.4 A sample (on the left) before normalization and (on the right) after normalization.

Normalization is quite a radical audio transformation and hence degrades the signal as well as giving results that are rather unexpected. Unless you really need to, avoid normalization.

Once Your Samples Are Shining

Once you have cut, trimmed, and buffed your samples so that they are shining, then you will be ready to deploy them in your sampler.

Chapter 3: Deploying Samples

The basic element for sample-based synthesis—the foundation on which it is built—is the sample.

As long as you have a single sample, you can create music with a sampler. However, you will find life gets much more interesting if you have a range of samples. And when you have a range of samples, then you have choices about how you deploy them.

The source of your samples is irrelevant. It doesn't matter whether you've captured a sample yourself or whether someone else did (but if you want to know more about capturing samples for yourself, then check back to Chapter 2: Capturing the Raw Samples). What matters is that you understand what you have and know how to use them.

If you don't know what you've got, then all you are doing is messing. But once you have a handle on your source material, then you can make choices and creative decisions about how it is deployed.

The first step before you can start tweaking is to get your samples into your sampler.

Uploading Samples

The process to upload samples is dependent on your specific sampler. With some you load individual samples, some allow you to upload groups of waves, and others require you to load complete sound sets with processing instructions. With some there is a menu system (see Figure 3.1), some have a browser (see Figure 3.2), and others allow you to drag-and-drop samples from outside the instrument (see Figure 3.3).

Figure 3.1 The pad menu in Battery from which you can add a sample to a pad.

Figure 3.2 The sample browser in Kontakt.

Figure 3.3 Dragging-and-dropping a sample on a pad in Battery.

Some other samplers offer tools or processes to organize your samples before you load them and in this way, the samples can be opened much like an oscillator is chosen (although instead of choosing a single wave, you will load a complete sample set).

One example of a format where the samples are organized before they are loaded comes with synthesizers that load SFZ format files. The SFZ format was developed by Cakewalk and is used in their flagship synthesizers (including Rapture, Dimension Pro, and Session Drummer) and many other (non-Cakewalk) synthesizers read this format.

There are two elements to the SFZ format—the waves and a mapping file (which is just a plain text file). The mapping file defines how each wave should be deployed (and can define a wide, wide range of parameters).

If you want to read more about the SFZ format, then you should check out my book Cakewalk Synthesizers: from Presets to Power User which dedicates huge amounts of space to looking at the format (and has, at the time of publication, the most complete documentation of the format). If you want to edit the SFZ mapping files, then you may be interested in SFZ Tools for UltraEdit which as its name suggests are some tools to create and edit the text files in UltraEdit a leading text editor. One of the benefits of the tools is that they highlight the SFZ opcodes so you can clearly see what you are editing.

For samplers where you load your waveform directly, simply opening the samples in your sampler will not organize them. You will need to do some further work to arrange them so that they play in the manner you are expecting.

Mapping Waves: The Principles

The first piece of organization is to lay-out (or map) the samples over the keyboard (or other trigger device), so that when you hit a key (or other trigger) you hear the sound you are expecting. By bringing together a group of waves, this mapping exercise takes the first step from a bunch of random sounds to a single, cohesive instrument playable over the keyboard (see Figure 3.4).

Figure 3.4 The target for your samples.

Let's look at the parameters that need to be set when we map samples.

Key Range

The key range is the range of the keyboard over which a single sample will sound (see Figure 3.5). Another way of looking at the key range is that it is the trigger key (or keys)—the key (or keys) that, when struck, will trigger the wave.

Figure 3.5 Distributing your samples over the keyboard.

If you want a sample to sound across the whole keyboard, then you would set the upper limit at the highest available note and the lower limit to the lowest available note, and then whenever you hit a key you would hear that sample.

If you want a sample to only be triggered by one key (which is quite common with drum and percussion samples as well as high fidelity acoustic instruments), then the lower and upper notes of the key range will be the same note.

Pitch or Key Center

The pitch or key center is the root note for a sample—it is the note at which the sample does not have any retuning when it is triggered (see Figure 3.6).

Figure 3.6 Defining the pitch center. Note: the pitch center does not need to be the center of the key range.

Conventionally the pitch center will be within the key range—most often it will either be in the middle of the range or at the top of the range (in order to reduce the effects of aliasing). However, it doesn't need to be. Most importantly, the pitch center is determined by reference to the sample—it cannot be changed (except by retuning the source sample). What we are doing by setting the pitch center is determining the reference point for how that sample is treated when it is deployed.

If the pitch center is out of the key range, then the sample will always be retuned when it is played. This will, of course, degrade the sample's quality, but it may give a tonal texture that is right for the instrument you are creating.

And just in case there's any confusion—if you set a key range beyond a single note (in other words, if the higher key range setting and the lower key range setting are different), then the key center does not need to be equidistant between the two notes.

If you set the pitch center incorrectly, then when the note is played, it will be out of tune. So, for instance, if the pitch center is F but you set it at G, then every note over the corresponding key range will be one tone out of tune. If you incorrectly set the pitch one (or more) octaves out of tune, then the played notes will be one or more octaves out of tune (so they will be at the correct pitch, but the character of the notes may be strange, which may or may not be a good thing depending on the context in which the sound is heard).

However, while you might play notes at the "wrong" pitch by setting the pitch center an octave (or more) above or below where it should be set, you may find some interesting tones and textures. As long as your goal is not reproducing an instrument with sonic fidelity, then take some time to explore these options.

Key Tracking

Key tracking is the extent, if any, to which a sample responds to incoming MIDI pitch information across its key range.

Regular instrument samples almost always have their key tracking set at 100 cents (one semitone) per key. With drum and percussion samples you might want to have a greater or a lesser key tracking amount. Then again, if your key range is set to one key, you probably won't care about key tracking.

Key Range Boundaries

In talking about the key range, I have implied that adjacent ranges will meet, but not overlap. In many cases—particularly when you are working with high fidelity samples or drum/percussion samples—this will be the optimal choice.

However, it is not the only choice.

The other main choice is to overlap adjacent waves so that two waves will play over a region. With this arrangement, it is very common to cross-fade over the overlapping region so one gets louder as the other gets quieter (see Figure 3.7).

Figure 3.7 Cross-fading between key ranges.

You might want to cross-fade in this manner to reduce the prominence of any difference when one sample ends and another begins. However, with this process you are mixing two sounds—you are not morphing one into the other—and accordingly, you will have two audible elements which you may perceive as two elements, rather than as a smooth transition.

Velocity Layering

The second dimension when creating detailed sample sets is the velocity layers.

MIDI velocity—in other words, the force with which a note is triggered/a key struck—gives us 128 gradations of loudness (zero force plus 127 variations where the note sounds). Therefore we can have up to 128 different waves triggered by a single key.

When taking your samples, as you strike the source instrument with more force, the tone will change (usually becoming brighter) and the volume will increase. By sampling the instrument when it is struck at different force levels, you can recreate these tonal and volume changes.

In reality with an acoustic instrument it is quite difficult to isolate 128 discrete tones for one note, even if you have a very skillful player. Depending on the source material, you will be able to create realistic sound sets with differing numbers of velocity layers. With drums you may need as few as five layers, but with an instrument such as a piano, you may find far more layers (perhaps 16, maybe over 30) are necessary.

The number of layers you need will depend on the source material, the fidelity you are aiming for, and the skill of your player (or if you are using a mechanical device to trigger your samples, the ingenuity of whoever is building the mechanical device).

Unlike with key range mapping, the changes between velocity layers—even if you have few layers—are less noticeable: the benefit of increasing the number of velocity layers is that it adds tonal nuance and makes the instrument far more playable.

There is, as you might have expected, a third option. You can cross-fade between velocity layers. Depending on the source material, this may work to a greater or lesser extent.

Figure 3.8 shows the three main options for building velocity layers.

Figure 3.8 You have three main options when building velocity layers (from left to right): use a moderate number of layers, use fewer layers but cross-fade, or to achieve higher levels of sonic fidelity, use more layers.

Another factor to consider when building your velocity layers is how incoming MIDI velocity is mapped in your sampler. Most samplers allow you to tweak the parameters to set how the minimum and maximum amounts of velocity are applied and to then tweak the curve between those extremes. Some even allow you to map all velocity levels as one level.

Battery has a flexible system where you can even invert your velocity mapping (so soft notes sound loudly and hard hit notes sound quietly). Figure 3.9 shows Battery's velocity mapping tool which is accessed on the setup tab in the lower pane.

Figure 3.9 The velocity mapping tool in Battery.

Mixing Samples across Velocity Layers

You don't have to restrict yourself to only using samples from one set. With a bit of care you can use velocity layers to switch between different sounds.

An example may illustrate how you could use this technique in practice. Say you have three bass guitar sample sets:

One sample set captures the sound of the bass guitar being plucked with the players fingers. This gives a thick and warm tone.

The second set has the same bass, but this time the strings are struck with a plectrum. This gives a harder, more pointed, but slightly thinner tone.

The third set captures the sound of the bass being slapped. This gives a bright percussive sound.

With these three groups of samples you could create one instrument where:

At lower MIDI velocities (say 0 to 64) the fingered samples could be called up.

In the middle velocity range (say 65 to 110) the picked samples could be deployed.

Across the top of the velocity range (111 to 127) the slapped samples could come into play.

With this set up, you have a warm fingered bass for general purposes and when you want to emphasize a note or phrase, by increasing the velocity the picked samples come into play. However, if you really want to make a point, then you can exaggerate the effect by playing with the slapped samples. Since the samples are all within one set, you can switch between them—and back again—just by using MIDI velocity. This kind of arrangement can create a very flexible, playable instrument.

This set-up also illustrates another aspect: you don't need to use the full dynamic range just for volume. Indeed, in many musical contexts, there's little point in having a particularly quiet set of samples since they won't be heard. Instead, you can use that end of the dynamic range to extend the tonal range of your sounds.

Mapping Waves: The Practice

So that's the theory, let's have a look at how you map some samples in the real world.

In this example I'm going to show you (briefly) how I set up some samples in Battery (as it's much easier to see what's going on). These principles can be applied to any other sampler, however, you may need to tweak a few other controls to achieve the same end.

Setting the Key Range

The first step is to select where the samples are going to be placed. Since I'm using Battery, this is easy—the obvious place is a cell, so for this example I selected the top-left cell. While selected, an orange border surrounds the cell.

In the lower pane, I selected the cell tab and made two changes:

In the name box, I renamed the cell as "Kick" (since I am loading kick drums).

In the Key Range box I set the range as C1/C1, in other words, the samples I load into this cell will only play when the C1 key is hit. Other cells can overlap the C1 key if I set them to overlap, but the samples I am about to load, will only play on the C1 key.

Figure 3.10 shows these two settings—both settings can be subsequently changed at any time.

Figure 3.10 Setting the key range and naming a cell in Battery.

Loading the Samples

Having set the cell and established the key range, my next step was to load the samples.

In the lower pane, I selected the Browser tab (see Figure 3.11). I navigated to the samples and loaded them—I have Battery set to load any sample I double-click on into the currently selected cell.

Figure 3.11 The sample browser in Battery.

Setting the Pitch Center

Having loaded the samples, the first step was to set the pitch center and to make this change I flipped to the Mapping tab in the lower pane (see Figure 3.12).

3.12 The Mapping tab in Battery.

As you can see in Figure 3.12, five samples were loaded and they were each set to respond over the full velocity range by default.

For the selected sample (which is the sample highlighted in orange) you can see (across the top of samples but below the tabs) the following parameters which can be adjusted by clicking-and-dragging:

Low Velocity. The lower velocity boundary for that velocity layer.

High Velocity. The higher velocity boundary for that velocity layer.

Tune. This allows you to fine-tune (or quite radically tune) individual samples. If you're after radical manipulation of your samples, then try retuning them by considerable amounts.

Pan. Each sample can have its own individual pan setting.

Volume. The level of each sample can be individually adjusted.

Root Key. The root key sets the pitch center for the sample. For these five samples, I set the pitch center (root key) to match the key range (in other words C1). This means that there is no retuning when any of the samples is triggered.

Selected Zone Name. This parameter shows the name of the wave and is not adjustable.

Velocity Mapping

Having set the pitch center for each sample, the next step is to set map the velocity ranges.

In Battery the process to set up a basic mapping is straightforward: click on the Edit button and on the drop-down menu select stack zones. This will map your sample across the velocity range. Figure 3.13 shows the two steps.

Figure 3.13 Mapping velocity layers in Battery (before on the left, after or the right).

This approach spreads the samples evenly across the velocity range. Now that each sample has its own velocity range, the layers can be tweaked if the response doesn't feel right.

I've shown the process using a basic auto-mapping tool. Most sample-based synthesizers have similar tools. Some are more sophisticated than I have demonstrated and allow key and velocity mapping based on the sample's name (Battery allows velocity mapping based on the sample's name). Alternatively, you can simply set the velocity layers manually.

The downside of illustrating the mapping with Battery is that—since each cell is addressed separately—you do not see the key ranges laid side-by-side. However, the advantage of Battery is that it is much clearer to see what is going on and the velocity ranges do not need to be stacked on top of each other. Figure 3.14 shows an example of the key and velocity ranges laid out together. In this instance, the samples have been mapped in the mapping editor in Kontakt.

Figure 3.14 The mapping editor in Kontakt showing key ranges and velocity ranges.

And Save

The final step after the samples have been loaded and mapped is to save the cell (see Figure 3.15).

Figure 3.15 Saving a cell's contents in Battery.

By saving a cell in this manner, the individual drum can be recalled at any point if you are building a kit.

Deploying Multi-Position Sound Sets

We looked at capturing samples from multiple perspectives is the last chapter (in the section Sonic Elements to Capture).

In case you skipped over that chapter, let me recap briefly with the example of a drum kit.

If you want to sample a drum kit, the first logical step would be to record the dry sound of each drum. To do this you would close mic each kit piece. This would give you the dry sound of each drum, but it wouldn't give a natural sound of a kit.

When a kit is recorded for a track there would be usually more microphones than just a close mic on each kit piece. Typically with a drum kit, after each close mic you might have an overhead pair, some room microphones, and some pressure zone microphones (PZM) to pick up the ambience. The producer will then balance these additional acoustic parts to create the final sound.

If you are trying to recreate a studio drum kit, then you are likely to have sampled the kit from several positions—probably the same positions that a producer would use when recording a kit in a conventional manner (close, overhead, room, and PZM ambience), so instead of having one kit piece with a range of velocity samples, you have one kit piece captured in four (or more) ways, each of which has a range of velocity samples.

Deploying all these samples is a matter of process (which also illustrates why you need to label every sample).

The first step is to ensure that you have all of your samples separately identified and grouped (so for instance, you have you close mic'd kicks in one group, your overhead mic'd kicks in another separate group, and your PZM mic'd kicks in a third group). You don't need to separate the samples out, but anything you can do to make the samples readily identifiable will make your life easier.

Step two is to load up your samples using the same process that we used earlier. You could load every sample into one cell, however, I recommend against this for two reasons. First, you will have a whole morass of samples to wade through and second, you will lose the ability to manipulate each individual group (so for instance, you wouldn't be able to solo the PZM samples at the click of a button).

When it comes to loading samples, my preference is to use the Battery matrix so kit pieces (snare drum, kick drum, and so on) are arranged in columns and the microphone positions (dry, overhead, room, PZM) are arranged in rows (see Figure 3.16). By using this structure I can then solo or mute a row or column allowing me to hear each element separately.

Figure 3.16 Multi-position sound sets laid out in Battery.

With the samples uploaded to their individual pads, they then need to be mapped, and here the process is exactly the same as the process we used earlier. In order to make the multi-position set-up work, there are a few areas where you need to pay attention:

First, the key range for each kit piece sample (irrespective of microphone position) must be the same. Taking the example we used earlier, this would mean that all kick drum samples would have their upper and lower key range mapped to the C1 note, and hence, when the C1 key is triggered, all four kick drum sample groups would be triggered. The other kit pieces would, of course, be mapped to their own key ranges.

The pitch center for each kit piece should be consistent.

The velocity layers should be broadly consistent.

One aspect you do need to think about is the outputs. When you are working with a dry kit (in other words, close mic'd samples) it is common practice to send each kit piece to a separate output. In Battery, you can set the output on a per-cell basis by selecting the cell tab in the lower pane—the output drop-down is then on the right-hand side (see Figure 3.17). Another common practice is to group similar pieces: for instance, to send all the toms to the same output, all the cymbals to the same output, the snare to its own output, and so on.

Figure 3.17 Selecting an appropriate output for each category of samples.

With ambient samples (in our example, the overhead, room, and PZM samples) you can send each individual instrument to a different output. However, there are two challenges with this approach. First, you're likely to run out of outputs and second, there are few reasons why you might need a separate feed and the complication of then mixing the separate feeds outweighs any benefits of separating the feeds.

A much better approach—and the one that is most commonly adopted—is to send all of the (non-close mic'd) microphone-type samples to the same output. So for example, all of the overhead samples (irrespective of the kit piece) could go to one output and the PZM samples could go to another.

With this sort of approach, each of the different sonic elements captured by the different microphone techniques can be controlled by a single fader in the mix. If for any reason there is an imbalance in the elements within the mixed feed, then you can always tweak the output level in respect of a specific kit piece at the cell level.

If you take this approach, using our example with overhead, room, and PZM samples, only three output pairs would be taken—this leaves a large number of spare outputs so the dry samples (or groups of dry samples) can still have their own individual outputs allowing further flexibility at the mixing stage.

Combining Multiple Sound Sets

With the wide-spread adoption of software samplers and wave streaming technologies making memory limitations a comparatively insignificant factor, it is now possible to create comparatively large instruments.

File size on its own is irrelevant. The point about creating a large instrument is that you can include a wide range of tones and nuances meaning that—in the hands of a skillful player—the sampler can behave in a manner that is much closer to a real instrument. In order to create these nuances, you need to capture the samples and then you need to deploy them appropriately.

Capturing the samples means moving beyond simply capturing the basic tones—instead you have to capture the nuances. For instance:

If you are sampling a bass guitar played with a pick, then you should sample the down strokes and the up strokes. By switching between the two, you can then recreate a more natural down-up-down-up sound.

Drummers create different tones based on how hard they hit the drum, but also where they hit the drum. A drum hit in the center gives a different sound to a drum hit towards the edge. Equally, a hi-hat will sound different based on where it is struck and how tightly closed the hat is when it is hit. And if you really want the detail, sample the drummer playing with their left hand and then with their right hand (at least for the snare drum where this second sample set will help you to create far more natural drum rolls).

I could go on, but you get the idea. You also understand that these techniques require a lot of work to capture and edit the samples and then the basic instruments (or rather sample collections using these alternate waves) need to be created.

Once you have the collection of waves, then you can look at how to deploy them. We've already looked at how to layer different samples across the velocity range—let's look at some more options.

Round Robin (Sequences)

If we go back to that example of a bass guitar where we want to implement a down-pick, up-pick, down-pick, up-pick arrangement to give a more natural playing style, there are two ways to achieve this goal:

First, set up two separate sample sets—one with down strokes and the other with up strokes. To play this, each sequential note has to trigger a note from the alternate group. As you can imagine, this would be nearly impossible to play by hand and the time to create a sequence would be longer than anyone would be able to tolerate.

The second option is to use a sequence in a sampler, or as it is often called, round robin.

Using a sequence we can cycle between up strokes and down strokes. If we have enough samples, we can have longer sequences, so for instance, if you have a series of hi-hat samples where the hat is struck in different places (or with different hands), then you could cycle between those different sets of samples.

The precise mechanics to set up a round robin vary by sampler, but there are usually three steps:

Select one of the samples to be included in the round robin.

Assign it to the round robin group. (If it's not in a group, then it won't know how or when to play in sequence).

Assign its position within the sequence.

Then repeat for the remaining samples within the group. And while I'm talking about samples here, what I really mean is groups of samples, so that you can switch between different notes and different velocity layers.

Let me give you an example of this in practice using Battery.

With Battery, you set a round robin to cycle between cells—this allows you to switch between groups of samples (which have had their key range, velocity layers, pitch center, and so on, set).

To set the round robin, you turn to the Cell Activation section under the setup tab in the lower pane (see Figure 3.18). This needs to be set for each cell in the sequence.

Figure 3.18 The Cell Activation section which can be used to set up round robin arrangements.

In the Cell Activation section, by default, the Trigger should be set to Note On (in other words, the sample will play when a note within the corresponding key range is triggered). Under the trigger line are two conditions, selectable from a drop-down menu. In one of the conditions, select the Round Robin option. This will give you three boxes:

Reset CC (which we'll ignore).

Cycle Nr. The cycle nr setting assigns the cell to a group. The group number (in other words the cycle nr value) is irrelevant—all that matters is each cell within the round robin must be assigned to the same group.

Position. The position setting determines the cell's position within the sequence.

Set the cycle nr and the position for each cell and you'll have your round robin set up with each cell playing in sequence.

There is one thing to watch. If your cells are assigned to different key ranges you may get unexpected results if you are not cautious. The round robin functionality in Battery works so that on any key press of any note within all of the key ranges there is a step forward in the cycle. Therefore, if you press a note which should trigger a cell that is not due to sound in sequence, then you will hear silence. In other words, if you are triggering a note in a sequence, for there to be sound you must trigger a note in its right sequence and within its key range. If you get this wrong, then you might keep triggering silence.

This is a very long way of suggesting that for round robins you should set the cells (or the notes if you're using another sampler) to have the same key ranges. In this way you can keep triggering the same note and the cycle will play as expected.

Key Switching

Another way to switch between two sets of samples is through key switching. In short, key switching is turning on (or off) a set of samples with a MIDI note. This could allow you to access a whole new set of samples at the touch of a key (or the bang of a drum). For instance you might want to switch from a set of fingered bass samples to a set of picked bass samples, and then later switch back to the fingered samples.

For key switching to have any real benefit, it should be used when two sets of samples covering the same key range. There is little purpose in using key switching to turn on/turn off ranges where there is no overlap. Assuming you have two overlapping ranges, the process to set up a key switching arrangement usually include the following steps:

Assign every sample set to a group (so that the processing is applied to all of the samples).

For each sample set, set the unique switch key (in other words, the key to turn on that group of samples—they will usually be switched off automatically when another set of samples is switched on).

To set this arrangement in Battery, first load your samples into your cells. For each cell where key switching is to be used, turn to the setup tab in the lower pane (see Figure 3.19).

Figure 3.19 Setting up key switching in Battery. When the red line is showing (as here) this cell has been switched off and another cell will play.

In the setup pane:

First, a voice group is selected (from the voice group drop-down in the voice groups section). Battery offers 128 groups—it doesn't matter which one is selected, it just needs to be consistent for all cells in the key switching arrangement.

In the cell activation section, by default, the trigger should be set to note on. Under the trigger line are two conditions, selectable from a drop-down menu. In one of the conditions, select on key.

Having selected on key, there are two settings low key and high key (which are set by clicking-and-dragging). These set the range of the switch keys (in other words, whenever a key within the specified range is hit, then this specific cell will be active—in Figure 3.19, the key is G2).

Repeat for each cell in the arrangement and then play.

MIDI CC Data

MIDI CC data gives another way to call up samples. However, rather than switching or sequencing between samples, MIDI CC data is more used as a condition. In other words, if MIDI CC number X is in the range between Y and Z, then the sample set will be active.

There are many uses for this. One of the most obvious is for positioning data with electronic drum pads. To mimic the behavior when a drum head is struck in different positions, many electronic drums (as in the MIDI hardware, not the sound you hear) will output MIDI CC data (most commonly CC4, CC16, CC17, and CC18) to reflect where the pad is hit. This data can be mapped to call up different sets of samples.

For Battery as you might expect, we're back to the cell activation section on the setup tab where from one of the condition drop-downs, on controller is selected (see Figure 3.20).

Figure 3.20 Setting a cell to only play when specified MIDI CC data is received.

Having selected on controller, there are then three parameters to set:

Start CC, in other words, the MIDI CC number.

Range Min, the minimum value for the MIDI CC data.

Range Max, the maximum value for the MIDI CC data.

When the appropriate MIDI CC data channel generates data and that data is above the range min value but also below the range max value, then the note will play if it is triggered.

Choke Groups and Polyphony

With real instruments there are certain sounds that cannot coexist. For instance:

An open hi-hat cannot continue to ring after a closed hi-hat has been struck (because the ringing open hi-hat must be closed for the closed hat to be able to sound. Equally, an open hi-hat cannot continue to ring after the player has stamped the hi-hats together.

An acoustic guitar cannot continue to sustain after the player has slapped their hand over the strings, thereby muting them.

A cymbal cannot continue to sound after the drummer has grabbed it.

Choke groups allow you to mimic these behaviors. When several samples (or groups of samples) are present in one choke group, then only one of those samples may play at any time. Therefore—while a note is still ringing—if you trigger a second note in a choke group, the first note will be cut and the second will sound.

To set a choke group in Battery, for all of the cells intended to be included in the group, select the appropriate group in the choke group drop-down in the voice groups section on the setup tab (see Figure 3.21).

Figure 3.21 The choke group drop-down selector is to the bottom-right of the Voice Groups section in Battery.

Battery has a variation on the choke group theme, allowing you to set voice groups. This option is available in many other samplers. The main use for voice groups is to set a limit on the number of voices that may play simultaneously. When the number of voices is limited to one, then this function performs much like a choke group. However, at higher settings, this function can cut out any mush when a group of samples sustain (by limiting the number of sounds).

In Battery (as Figure 3.21 shows) you have the following voice group options:

Voice Group (drop-down) selects the group.

Max Voices sets the maximum number of voices that may be played simultaneously.

Voice Overlap sets the time that the voice that is being muted will continue to ring (in milliseconds) after its muting begins.

Steal Mode sets how the note to be killed is determined. The options are: kill the oldest, kill the newest, kill the highest, and kill the lowest.

Release Samples

Staying with the idea about sounds that cannot coexist, think about what can happen when a sound created by a real instrument is stopped:

Taking the example I used earlier, with an acoustic guitar, the strings can be made to stop ringing by the player slapping their hands across the strings. This could give a thump and a sound of the stings being slapped against the fretboard.

When a piano key is released (assuming the sustain pedal isn't pressed) the hammer will return to rest on the string, muting it.

In both of these examples, after the note stops, there is another sound, albeit for a short time. There are many other examples of these sounds (for instance, a cymbal being muted by grabbing it while it rings). These sounds are called release sounds—in other words, they are what you hear when the key is released.

You can create a set of release samples on many instruments. However, the sounds can be more fiddly to create than conventional samples since there will be no silence preceding the sound. Instead, you will have to cut your wave as close as possible to immediately before the release section.

Once you have your regular sustaining sample and your release sample, then you can deploy them together.

Choke groups are useful for either/or samples, but they are rather unsophisticated in that you need a second sound to be triggered to choke the first. So for instance, if you take a piano, it would be strange to have to trigger a separate key in order to deploy the release sample to choke the sustaining note.

As you might expect there is a better way.

Conventionally, samples are triggered by a note on event—in other words a key is pressed (or other trigger executed) and the sample will sound. However, you can also set the note off event (when the key is released) to act as the trigger.

If you take a set of sustaining samples and a set of release samples and map them both over the same key region:

The sustain samples can be triggered by the note on event.

On the note off event, the sustain samples can be set to stop, immediately. This will give a slightly unnatural effect, since acoustic instruments always have a certain ring.

The release samples can be triggered by the same note off event. These sounds will counteract the unnaturally fast termination of the sustain samples.

As you might expect, in Battery, in the cell activation section on the setup tab, there is an option to set the trigger (for the cell) to note on or note off (see Figure 3.22).

Figure 3.22 The Note Off trigger in Battery.

Saving Files

Once you have loaded your samples, you should save the set, and as you map and tweak your samples, you should save regularly.

Depending on your sampler, you will have different options for what you can save. For instance, you may be able to save (and therefore load to deploy):

Groups of waves. For instance in Battery, you can save individual cells. Other samplers allow you to save regions, layers, or zones.

An instrument—in other words, a complete set of samples (which may contain differently mic'd samples, key switching options, and so on).

Several instruments creating a section.

When you save your samples, you will often have two options for how your samples are saved: as monolithic files or as individual samples with a mapping file.

Monolithic Files

A monolithic file is one single file with all your samples stored inside it together with mapping details.

As you can imagine, these files can be quite large, however, they do have the advantage that all of your files are together in one place—you will never lose a file and if you're sharing files you can be sure that everything will be transferred. The downside is that if the file becomes corrupted, then all of your samples are lost.

Some monolithic files can be opened (since they are effectively zip files). However, many cannot be.

Individual Files with Mapping

Some samplers store their files as individual wave files together with a mapping file.

The advantages of this approach are:

You can tweak, edit, or even replace individual wave files.

A corrupted file can be replaced (and/or mapped around).

You can have several mapping files referencing the same group of samples.

Memory Management: Disk Streaming, RAM, and Purging

With the advent of disk streaming, it's easy to forget that memory is still a consideration.

Disk streaming (or DFD—direct from disk—as it is called in Kontakt and Battery) still uses a significant amount of memory. Instead of loading the whole of every sample into memory, when disk streaming is engaged the sampler will load the start of each sample. Then if the sample is triggered, the start plays from memory (since that can be accessed more swiftly) and the remainder is streamed from the hard drive.

Although disk streaming allows the use of sample sets that are larger than the system memory, clearly there is still only a finite amount of memory for the pre-loaded elements. Also, by having part of the sample in memory, part on the disk, and calling the sample from the disk, there will be additional overhead placed on the processor (to calculate what should be played, when, and to call up the appropriate sample on the hard drive) and there is a limit to the amount of data you can get off a hard drive within a limited time. Accordingly, even if you have the fastest processor you can afford, you may still find a bottleneck with your hard drive and moving data around on your computer.

The effect of disk streaming is that you can usually play fewer voices simultaneously than you can if you have loaded your samples into memory. (However, remember this is just a simultaneous voice limit—if you use disk streaming you can still deploy larger sample sets than if you load your samples into memory).

If your computer has a large memory, you may choose to avoid disk streaming in order to avoid system bottlenecks. If you load the samples into memory, then you should consider purging.

Many high-end samplers offer some form of purge where the samples that are not needed can be unloaded from memory. This will allow you to load more samples into that free memory (or use the memory for other tasks).

The process to unload samples usually involves playing the track—the sampler will then note each sample that is used. You can then hit the purge button and your sampler will unload any unneeded sample.

Further Editing

Now that we've got our samples into our sampler we can think about some further tweaks. In the next chapter I'm going to look at built-in editing tools and how loops can be controlled.

Chapter 4: Controlling Samples and Loops

The best strategy when making music with samples is to get your source material right before you deploy it. In other words, sort your samples before you load them and if you've got a problem with a sample, then go back and re-capture the sample from the original source. And if you can't re-capture, then fix the sample as best you can in an audio editor before you load it.

Anyway, that's the theory... but as you might expect, sometimes life doesn't turn out like you hope. In practice there are many reasons why you might want to edit a sample (or a bunch of samples)—these reasons include:

To deal with a flaw in the original source material—perhaps a pop or a click which only becomes evident after the sample has been loaded.

To tweak the sample so that it gives a different sonic effect, for instance, you could cut the attack portion of a sample to give a smoother tone. You could, of course, go further and edit for creative effect and find a result that has no resemblance to the source material.

To make the sample behave in a different manner in different circumstances.

To create a rhythmic effect—for instance, you could remove elements of a sample and create a loop with the material that remains. Equally, you could create a loop and edit elements within that loop in order to change the feel of the rhythm.

The Wave Editor

If you want to edit a sample, then you can open it in an external wave editor. However, many samplers—for instance Battery, see Figure 4.1—have a built in sampler where you can edit your samples.

Figure 4.1 The wave editor integrated into the lower pane in Battery.

Most built-in wave editors do not include the full range of features seen on dedicated wave editors, but this streamlining makes them much faster, and more intuitive, to use.

There are many reasons why you might want to use these built-in editors rather than an external editor:

First, of course, you might want to edit a wave. However, there are other reasons.

With a built-in editor, you can hear the sample on its own and without any other form of processing/retuning/layering/FX, and so on. You can play the whole sample and preview any loop elements within the file.

You can edit the sample and immediately hear the effect of the changes within the context that the sample will be deployed. Sample editors will allow you to zoom in so you can see what you're editing and will often help you find zero crossing points (or other appropriate editing points).

If you have received your samples as a monolithic file, you may not be able to access the raw waveforms in order to edit them.

There will still be times when an external wave editor is the right tool. For instance you may want to undertake a transformation that isn't available or you may want to perform a batch process. However, for many edits, the built-in editor will be the right tool.

Basic Edits

The basic—and most frequent—edit that you are likely to undertake is to trim your samples. When you trim the sample, you cut an element from the start or the end of the sample, so that the sample begins at the cut point and ends at the second cut point.

Figure 4.2 shows the wave editor in Kontakt. The process to trim a wave in Kontakt is straightforward and non-destructive. If you look at Figure 4.1 you will see a green line on the left with an S label—all you have to do is drag this label to set the start point for the wave. On the right, there is an orange line with an E label. To trim the end of the sample you drag the E label.

Figure 4.2 The wave editor in Kontakt. As you can see, the start of the wave has been trimmed (and the end has been prematurely truncated for visual effect).

Another simple edit is to add an envelope. We'll look at envelopes in more detail in chapter 7, but for the moment, let me give you the simple version. Envelopes are levels that change over time.

Usually envelopes are applied to a whole instrument (or at least a collection of waves). Kontakt (along with many other samplers) allows you to apply an envelope (non-destructively) to individual waves.

Figure 4.3 shows a volume envelope drawn on a wave.

Figure 4.3 A volume envelope controlling a wave. The red line is the volume envelope.

Envelopes can be used for creative or surgical purposes. For instance, you could apply an envelope so that a wave fades in. Equally, if you have a click in a sample, instead of trimming, you could use an envelope to drop the level for the duration of the click, thereby minimizing the effect of the click.

As I said, I will look at envelopes in other contexts in chapter 7, but before we get there, a word caution when using envelopes. Envelopes can only modify the source material—they cannot substitute their own intention. So for instance, if you have a sample that gently fades in, you cannot set a sharp envelope to create a pluck sound. If this is the aim, then you might be better trimming the start of the fade-in in order to get sharp start to the sample.

Destructive Edits

The edits we have looked at so far have been non-destructive, in other words, you can reverse your changes (without needing to hit the undo button). In practice, these edits are an overlay which controls how the sample deploys.

Many wave editors also allow you to make destructive—or permanent—edits. Depending on how your sampler stores your waves, very often these permanent edits can then be accessed by other instruments created in your sampler.

The Kontakt sample editor (see Figure 4.4) is a good example of an editor that allows you to make destructive edits.

Figure 4.4 The sample editor in Kontakt.

The edits that are available in the Kontakt sample editor are split under two headings (edit and transform). To make a change an element of the sample must be selected (or for pasting, an insert point selected). As you can see from figure 4.4, the selected region is highlighted and the display in the bottom center of the editor shows the start and end of any selection, together with the length of the selection. To the right of this display, is the undo and redo buttons so you can escape from any unwanted change.

Edit

There are six edits which work very much as you would expect:

Cut, cuts the selected region—the wave before and after the cut will then be joined so there is no gap left.

Crop deletes everything apart from the selected region. This is a destructive way of trimming the start and end of a sample.

Paste inserts a copied region. The paste is made at the current selection point and the waves before and after the insertion are joined to the new section.

Copy copies the selected region.

Delete removes the selected region. The wave before and after the deletion will then be joined to remove any gap left by the deletion.

Duplicate copies the current selection and pastes it at the end of the selection (joining both ends) so that in effect, the selection is played twice.

As you can see, these edits behave in a similar way to how an edit in a word processor performs, however, instead of messing with text, we are changing a raw sonic waveform.

Transform

Rather than cut, copy, or paste, the transforms work to change the waveform. As with all changes to a wave, transformations degrade the source wave. However, the quality difference is likely to be so minor that you are unlikely to be able to detect any change.

Once a transform has been applied, then the wave can be transformed again—in other words, you can edit an edit (or transform a transform). Cleary this has the potential to really mess with waves—but it also has the potential to create some very interesting sounds. And while you have a wide range of editing options—and I would encourage you to use them to their full extent—please don't use these options to fix the unfixable (you're never going to make a piano sound like a trumpet...).

The transforms that Kontakt makes available are fairly typical of what you can expect to find if your sampler has a wave editor.

Fade in sets the level of the wave at the start of the selection to zero and progressively increases the level across the selection. The end level of the selection is then set at the level it was set at before the transformation (in other words, while the volume gets louder over time, the overall volume is not increased—for that, you would need to normalize).

Fade out leaves the level at the start of the selection unaffected, but decreases the level of the wave over the time of the selection, ending at zero at the end of the selection.

Silence sets the level of the complete selection to zero.

Reverse turns the selection around—re-joining each end—so that the selection then plays backwards.

Normalize increases the level of the selection so that the loudest peak is set to the maximum level. The volume of the rest of the selection is scaled in line with the increase in the peak level.

DC Rem removes and DC bias in the wave (which may lead to the wave appearing to be louder than it is).

Sample Loops

There are loops and there are loops. You need to understand the difference to avoid confusion.

There are loops where the whole sample is repeated in a loop. Typically, these samples are beat-related and as the sample repeats, the beat plays.

The other form of loop occurs within individual samples. This is the sort of loop we are dealing with first.

Short Samples

In the early days of sampling, memory was highly limited and therefore it was necessary to use very small samples (also samples tended to be carried around on floppy disks making loading a real chore). There were several ways to reduce the size of samples—use fewer samples and have wider key ranges, use mono samples, being two examples—but the most common (and in many ways most effective) approach was to cut down the size of the sample.

Instead of having a sample that may last, say, 20 seconds it would be common to cut the sample—perhaps so that it was four or five seconds long. The cut could be made in a wave editor or a shorter sample could be captured. Then, to recreate the element that was removed, in the sustain section of the sample, a loop would be added. When the sample reached this point, the loop section would continue to play while the key was held. When deployed, this would give an effective infinite sample.

The loop section is effectively nothing more than some metadata embedded into the sample denoting the loop start and loop end time. When you load a sample with this metadata, your wave editor will usually display it (however, some simple wave editors ignore this data). Figure 4.5 shows a sample with a loop in Kontakt.

Figure 4.5 A sample with a loop in Kontakt. The loop is the orange section in the middle of the wave.

You may find yourself dealing with loops in two situations:

First, if you have a sample which (theoretically) could sustain indefinitely. For instance, a string sound could theoretically sustain indefinitely. In capturing the sample, at some point you're going to have to end the recording, however, on deployment that sample may be required to play for longer than that length captured, in which case a loop will be necessary.

Second, you have a set of samples (probably an old set of samples)which (by design or accident) are prematurely truncated such that you don't have a suitable sustain position.

Creating and Editing Sample Loops

If you want to create loops because your samples do not have an appropriate sustain section, then you are in for some tedious work. However, the wave editors in most samplers will help you to a certain extent. To set a loop, you probably want to follow a process that includes the following stages:

Once the wave is in your wave editor, ensure that you have set your snap options—ideally set your loops to zero crossings. With this setting you can expect that not only will zero crossings be found, but that similar zero crossings (in other words, the zero crossing at the start and end of the loop will both have the wave moving in the same direction—see Figure 4.6. This will make the loop points smoother).

Figure 4.6 Loop start and end points in Kontakt's wave editor. Note how the wave is ascending at both zero crossing points—this makes the loop smoother.

Select your loop area. To do this, you should look for a part of your sample where the level stays fairly consistent.

Select your sample area—this is usually a matter of dragging your mouse over the area. With a longer area you can expect a smoother response—short loops tend to sound very static and unnatural.

Audition the sample to see whether it plays smoothly. If it doesn't, then adjust the start and end points. If your sampler has an automatic find functionality, then you might want to try that. You can get excellent results with automatic finding, however, you can get awful results. If you get a bad result, then keep tweaking. You can also cross-fade the loop ends in order to smooth the loop.

Once you've put the loop into your first sample, then move on.

By the way, if you're wondering whether you've got your sample loop "right", there's a simple test: listen, and ideally listen to just the loop (most sample editors allow you to preview a loop). If you can hear the loop—in other words, if you can hear when the loop ends and starts replaying or you can hear an audio artifact (a click or a buzz) when it loops, then you need to do some more work.

Loop Options

As well as setting the loop start and loop end point, together with any cross-fade to smooth any rough edges, most wave editors have a number of other options. Staying with Kontakt as an example, it offers:

Tuning. A loop can be tuned to a different pitch.

Count. The loop can be set to repeat a specific number of times before the wave continues to play.

Direction. Conventionally, a loop play from its start to its finish and then restarts. However, this doesn't always have to be the case. For instance, the loop could play to the end and then play in reverse and when it reaches the start it can begin to play forwards again and the process continues. Sometimes—but not always—you can get a smoother loop by playing a wave backwards and forwards (since you are effectively joining it with itself).

Loops

Beat loops are the sort of loops you're probably more familiar with. Usually these are rhythmic phrases that are played in a continuous pattern to create a beat. Often they are drum patterns but they can be any instrument. They can be purely rhythmic or they can include a melody.

There are many, many, many tools out there for handling loops. However, your sampler may also be able to help.

There are many reasons to use your sampler with loops—perhaps the most compelling reason is the amount of control that will be available under your finger. Most samplers won't just give you a single option to handle loops, but instead will offer many options, which can usually be used in combination. In addition, since you can edit your raw waveform, you can directly cut up and reorder loops within your sampler giving yet more sonic options.

Loops: The Challenge

Very often you will have a track which needs some rhythm—rhythm that can be supplied by a loop—and you will have a loop that will be perfect, but which is at a different tempo to your track. Looked at another way, you will have a hole, but the plug to fix the hole will be a different size.

To fix the hole you will need to stretch or shrink your plug—in other words, you will need to make your loop longer or shorter. But not only will it need to be longer or shorter, it will need to be tempo-synchronized to the track.

In the early days of samplers, there were two main options for fitting a beat to a track.

The first option was to change the tempo of the track to fit the groove. This had the added advantage that the groove would sound right (because it was at its intended tempo).

The second option was to speed up or slow down the loop thereby making it shorter or longer. The way that this speeding up or slowing down was effected was by retuning the loop—as a loop played faster it became shorter and as it played more slowly, it became longer.

Both options are still valid approaches, but as you will have figured, they lack a certain flexibility and there are consequences—either your track changes or the nature of your loop changes (significantly). Perhaps the biggest challenge with these approaches is that you can't then change the tempo of your track

There are now a wide range of tools available to process loops so that they can be stretched, shrunk, and messed about with to match the tempo, but without changing their pitch. Most modern samplers allow loops to be processed in one of two ways: stretching and slicing. Let's look at those in turn.

Stretching

Stretching takes a number of different names including stretching, compressing, warping. Essentially these are interchangeable terms which describe the process by which a sample is stretched or compressed to fit the allotted time. Complex algorithms increase or decrease the length of the loop while maintaining pitch.

So there's no confusion, when I talk about stretching, I mean stretching, compressing, warping, and doing anything else that isn't slicing.

One key difference between stretching and slicing is that with stretching the loop is treated as a single entity and so will give different results when processed (when compared with slicing). As a general rule, stretching will achieve better results from non-beat or less-beat sources.

However, this is not a hard-and-fast rule and you will need to experiment based on your source material.

Kontakt has a stretching algorithm called time machine. To use this functionality:

Load a sample and open up the wave editor.

As this is intended to be a loop, then—assuming it's a raw wave with no embedded metadata—go to the sample loop tab in the wave editor, select the whole wave and switch on the loop.

Hopefully, the loop has been edited so that it loops properly, but if there are any glitches, then these can be cut out with the wave editor. Alternatively, you can tweak the loop start and loop end parameters on the sample loop tab.

Once the loop is playing smoothly you can switch to the sync/slice tab in the wave editor (see Figure 4.7). Here you should first set the zone length—the bar/beat length of the loop. It is important that you set this parameter correctly as it is the basis on which the sample will be deployed.

Having set the zone length, all that is needed in to click on the use time machine button.

Figure 4.7 The Sync/Slice tab in Kontakt's wave editor.

The loop will now play. Its tempo will match the tempo of the host (or if Kontakt is being used outside of a host, the loop will match the internal tempo set by Kontakt).

The loop will be playable across the entire key range that the loop sample has been assigned to and (provided the tracking button has been engaged—see the bottom section of Figure 4.7) the sample will change pitch according to which note (within the key range) triggers it. Despite the pitch changes, the tempo will remain consistent across the whole key range.

If we hadn't engaged time machine, this loop would be treated like any regular sample and its pitch would have been controllable across the keyboard, but more significantly, it would have speeded or slowed with the change in pitch (thereby meaning it wouldn't have matched the beat of the underlying track).

Kontakt offers other algorithms. There are variants on the time machine algorithm and there is also the beat machine algorithm which acts in a similar manner, however, it is optimized to preserve the peaks during the stretching process and so behaves in a closer manner to beat slicing.

Beat Slicing

The other main way of handling loops is beat slicing. As you might expect, this process takes a loop and slices each individual beat. When the loop is replayed, each slice is played in turn.

Since there are individual slices, changing the tempo simply requires that the slices are deployed more quickly or more slowly. So instead of stretching/compressing/warping the whole loop, all that is changed is rate of deployment.

There are many other advantages to this approach:

Each slice can be assigned to its own trigger key. This means that each slice can be played individually. Most samplers will output (usually by drag-and-drop) a MIDI file to trigger the individual slices in the correct order.

Since there are individual slices, specific elements can be changed, for instance, you could flip out all the beats with a kick drum and replace the kicks with a new element.

Each slice can be individually edited.

Since there is no warping—and all you are doing is deploying each slice more rapidly or more slowly—beat slicing tends to be better suited to rhythmic material. If slices are deployed more quickly then each will overlap the previous. If the slices are deployed more slowly then there will be gaps between the beats. These gaps may or may not be noticeable, depending on the source material and the amount by which the tempo is dropped.

Beat Slicing in Practice

All beat slicers work in a similar manner and offer similar tools, however, specific beat slicers will have specific quirks. I'm going to illustrate the basic beat slicing principles with Kontakt.

The first step is to load a sample and open up the wave editor.

As with when we used the time machine to stretch the wave, we can go to the sample loop tab in the wave editor, select the whole wave, and switch on the loop. Hopefully, the loop has been edited so that it loops properly, but if there are any glitches, then these can be cut out with the wave editor before we go any further—a badly looping wave won't be fixed by beat slicing.

So at this point, you should have a smoothly looping sample which will probably be mapped across the keyboard, so it will speed up and slow down—and its pitch will rise and fall—as you play it above or below the pitch center (see Figure 4.8). In other words, we have a conventional sample which plays across the keyboard.

Figure 4.8 A loop mapped across the keyboard and ready for slicing. In this case, the sample loop tab is displayed.

A crucial difference between a sample and a loop is the beat information. With a loop we need to set the bar/beat length of the sample. This ensures that the sample plays at an appropriate tempo when it is deployed and we use the same controls under the sync/slice tab that we used with the time machine setting. The zone length should be set automatically, but if Kontakt guesses incorrectly, then you can tweak this figure by hand.

With that done, we can move on to the slicing and there are three options for slicing:

fixed slicing

automatic slicing, and

manual slicing.

In Kontakt these options are selected in the grid section (in the bottom-right of the wave editor). This section will guess at the BPM (beats-per-minute) of the source loop and the time signature. As long Kontakt guesses correctly, you don't need to make any tweaks here.

Fixed Slicing

With fixed slicing, a loop is divided into an equal number of slices. This form of slicing can work well when you have a loop with a regular beat, for instance, if you have a drum machine loop sample with a noise on each beat. However, if you have a sample of a real drummer—in particular, a real drummer who can swing—then this option is less useful. Equally, if you don't want to have slices of silence, then this is not the right option.

Figure 4.9 shows fixed slicing deployed in Kontakt.

Figure 4.9 Fixed slicing in Kontakt. In this instance, each bar was cut into 16 slices—the gap between each slice is marked by a vertical blue line.

Automatic Slicing

With automatic slicing, the slice tool looks for peaks and puts a slice marker immediately in front of each beat.

Most slice tools have a sensitivity control so that you can adjust which peaks are picked up (or are not picked up). A sensitivity control can also help where it is not clear where one beat ends and the next starts—a situation that can occur when two beats are close together with the second starting before the first has decayed.

As you can see in Figure 4.10 and 4.11, by adjusting the sensitivity control you can change the number of slices that are created.

Figure 4.10 Automatic beat slicing with a low sensitivity setting creates a limited number of slices.

Figure 4.11 Automatic beat slicing with a high sensitivity setting creates a much larger number of slices, including many that are probably unnecessary.

Manual Slicing and Editing Slices

Fixed slicing and automatic slicing will both give you good results. However, they may not give perfect results.

Before we carry on, perhaps it's worth remembering why we're slicing and what we're trying to achieve:

The intention—as far as possible—is to separate each individual beat within a loop.

Having separated each element within the loop, we can then play each element independently of the other elements.

We can also treat each element separately.

When deploying the loop, the time shifting will work much more smoothly if each beat is separated.

Or put another way: we will get much better results by getting our slices right. And the way to get our slices right is to manually edit the results of fixed or automatic slicing. With manual edits we can:

Add new slices where a beat has not been segregated.

Remove a slice marker where silence has been segregated (since there's no point in having a silent beat).

Move a slice marker where the marker has been placed incorrectly (thereby missing or cutting a beat).

The process to add, remove, and move slices is straightforward in Kontakt. On the right-hand side of the grid section is a section with a flag, a plus button, and a minus button (see Figure 4.12 which also shows a marker being dragged).

Figure 4.12 Manually adding a beat slice.

To add a slice, click on the plus button and then click on the waveform where a slice should be added. Wherever you click a slice will be placed—if you click-and-drag you will add several slices.

To remove a slice, click on the minus button and then click on the triangle at the top of the slice you want to remove.

To move a slice, ensure that the add and remove slices are switched off (neither the plus button nor the minus button should show in blue—both should be gray) and then click-and-drag a marker you want to move.

Repeat until you have each element separately sliced.

By the way, all we are doing with these slices is adding markers, so if you get it wrong, it doesn't matter—we're not making permanent changes to the loop.

Deploying Slices

So now you've got the slices, what next?

The first step is to lock the slices so that you don't accidentally move a marker. In Kontakt you can do this by clicking on the padlock in the bottom-right of the Grid section. Having secured your slices, there are probably two things you will want to do now:

First play your slices—individually or in sequence.

Second, edit individual slices.

One of the advantages of editing samples within a sampler is that you can make changes in real-time (or something close to real-time) with immediate feedback as your loop is played in context.

To deploy individual slices, you need to map them, giving each slice its own key range (usually a single key). In Kontakt, you can drag each slice individually to the mapping editor. Clearly this could be time consuming and there is a better approach.

Under the sync/slice tab, there is a button labeled drag MIDI to host. You can use this to drag a MIDI file containing the trigger pattern to your host. When you play this file, it will then trigger the individual mapped slices in turn to recreate the loop.

The act of dragging the MIDI file auto-maps the slices across the keyboard as you can see in Figure 4.13.

Figure 4.13 Slices auto-mapped across the keyboard in Kontakt.

There are some settings in the Auto-map slices section that you should consider before you drag the MIDI file:

The Auto-Fade time (in Figure 4.13 this is set to zero) can fade the start and end of each slice to smooth out any rough edges.

The Auto Find Empty Keys button will, if engaged (when it turns orange) map the samples at the next convenient position.

If you don't engage the auto find empty keys option, then you can set the base key for the mapping using the Map Base Key function.

As a side note, you can create your own slices with a wave and a wave editor. Each slice can then be individually deployed in your sampler. The advantage of using a specialist slice tool (such as Kontakt) is speed and ease of use. If you manually slice the wave, you would then have to reconstruct your own MIDI file. With Kontakt (or a similar tool) the MIDI file is automatically created and you can keep tweaking each slice within the loop. Added to which, with a dedicated slice tool you lose none of the flexibility—once the loop has been sliced, each slice can be individually triggered and manipulated.

Editing Sliced Loops

Once you've got your slice markers set, you can start editing. We've covered most of the edits earlier in this chapter, but let me just point out a few changes you can work with and you can check back to the earlier material.

First, once your loop is sliced, you can apply an envelope (and this would be applied to the original sliced envelope where you have placed the markers, not to the individual slices).

To apply an envelope, with the sliced loop selected (and to emphasize: the loop, not the slices) go to the zone envelope tab in the wave editor and select volume or pan—this will apply a volume or a pan envelope. You can then adjust the level or the pan position of each slice (see Figure 4.14).

Figure 4.14 A volume envelope modifying each slice of a loop.

Earlier in this chapter we looked at the Basic Edits and Destructive Edits you could make with the sample editor in the Kontakt's wave editor. You can use these same tools to edit individual slices.

If you select an individual slice (click on the relevant slice in the mapping editor) and then go to the sample editor, that slice will be highlighted within the context of the wave. You can then drag over the wave and apply the transformation. Figure 4.15 shows an example edit where the first slice has been reversed.

Figure 4.15 Reversing a single slice.

Once you have your loop sample in slices, the number of changes is only limited by two factors: your patience and your creativity.

Applying the Edit

We've now looked at the main edits you are likely to perform. In the next chapter we will start looking at how we can we can apply these principles to create the results that people will listen to.

Chapter 5: From Samples to Sounds

So far we have focused on the raw samples and how those samples can be marshaled. In particular, we have looked at how to capture samples, how to deploy those samples in a sampler, and how to control those samples once they have been deployed in a sampler.

Options for Building Sounds

In many instances, all you will want to do is bring together your samples and deploy them across the keyboard and velocity ranges. But there are other choices and other options and none of these options is exclusive—you can mix and match to create your desired sound.

The High Fidelity Approach

The high fidelity approach is the approach we have focused on (since you can always turn high fidelity sounds into low fidelity sounds, but you can't reverse the process).

To create high fidelity sounds, you maximize the number of (high quality) samples and create detailed sound sets. The characteristics of these sound sets are small (usually single note) key ranges so there is no retuning of samples and many velocity layers. Each separate playing nuance is then captured in detail and separately deployed—often using key (or other) switching techniques to bring up the appropriate nuance.

For added realism and control, these sound sets use different microphone settings, so you can balance close mic'd sounds with ambient sounds.

If you are looking to faithfully reproduce the sound of an instrument, this is the approach for you.

The Layering Approach

With the layering approach you take two or more sample sets and layer them.

With this approach you have a lot of options:

You can layer the same instrument but with different playing techniques (for instance, as I suggested in the earlier chapters, you could layer a bass guitar so that at lower velocities you hear the fingered sound, at medium velocities you hear the picked sound, and at high velocities you hear the slapped or popped sound).

You can distribute different instruments, for instance, you could place a violin, viola, cello, and double bass across the keyboard—with or without each having an overlapping range. By placing the individual instruments across the key range, you can create a single playable string instrument.

Alternatively, you could layer completely different instruments, for instance, you could layer a piano and strings to benefit from the attack of the piano and the smooth sustain of the strings.

This approach gives you the ability to create new instruments and also allows you to work around the limitations of individual instruments (for instance, you no longer need to worry about the decay of a piano or the limited key range of a violin).

An advantage of this route—especially if you're building sound sets—is that you don't need to use so many samples. Since you're not creating real instruments, there is less need to include the detail and the nuances can come from the interaction between the different layered elements. However, if you want, you can still layer highly nuanced sound sets.

The Super Real or Enhanced Approach

The super real, or enhanced, approach takes the high fidelity approach and kicks it up a notch. This approach can also be applied in conjunction with the other approaches.

With this approach, you take your source material—your samples that you have deployed—and you enhance the sound. Typical enhancements will involve adding EQ, compression, or reverb (all of which are discussed in Chapter 8: FX.

With this approach, the aim is to enhance the sound but without making a noticeable change. There will be a change, but the point is that the listener will hear "perfection" without realizing that the original sound has been adjusted.

The Analog Synthesis Approach

In essence a sound created by an analog synthesizer has three main elements (see Figure 5.1):

the sound generator (an oscillator)

the filter (to shape the tone), and

the volume control (to control the level of the sound over time).

Figure 5.1 The basic signal flow in the sound creation process.

Once the sound has been created, it is then shaped over time—both in terms of the tone and the volume—to create and control the tone. This type of synthesis—which is seen in every analog synthesizer and analog-emulating synthesizer—is usually called subtractive synthesis: the sound generator creates a sound and the filter "subtracts" elements to reach the final sound.

Once the basic sound has been created, a range of control sources are employed to effect tonal changes over time. These control sources include:

envelopes, and

oscillators (often low frequency oscillators or LFOs).

One of the key differences between analog synthesis and subtractive synthesis applied in the context of sample-based synthesis is the sound source. With analog synthesis the sound source will be a (comparatively noisy) oscillator. With sample-based synthesis, the sound source is a sample or sample set. These samples—particularly if you have constructed a highly detailed sample set—may have a sound that is close to the tone you are looking to create and will almost certainly change in character both over the keyboard, over the velocity range, and over the duration of a note.

Accordingly, when deploying analog synthesis techniques in conjunction with samplers we are likely to want to be far gentler in our application of the tools at hand.

Depending on the extent of the shaping you intend to undertake you may not need a detailed sound set. Indeed, extensive use of analog techniques can cover shortcomings in a sample set.

We'll look at the application of analog synthesis techniques to sample-based sounds in greater detail in the following chapters.

The Sample Mangling Approach

All of the approaches so far have treated the samples with great reverence. Some approaches may have suggested a gentle shaping, but there has been nothing too radical.

However, some of the most striking results can be achieved by messing with the sample itself.

When you are messing with samples to an extreme extent, it is often an advantage to have a limited number of samples (if you're using a sample set covering 88 keys each with, say, 32 velocity layers, that could imply nearly 3,000 samples to be mangled... at some point the creative urge is likely to leave you).

Sample Mangling Techniques

When it comes to sample mangling, there are no rules.

None.

You can do what you want.

You can mangle your complex, detailed sample sets, but as a starting point (and to save time) I would suggest it is probably best to start with one or two samples and push them to their limits, and then a bit further.

First Steps: Messing with Pitch and Tempo

When it comes to mangling, the first step will usually be to mess with a sample's pitch.

You may be wondering about the reference to tempo in this section heading. In the context of a beat-related loop it makes sense, but at first glance, you may not see the relevance for conventional samples. However, remember that without taking any further action, if you deploy a sample and play it across your keyboard, it will play more quickly as the pitch rises and more slowly as the pitch falls.

As we saw in the last chapter, most samplers give us tools to keep the tempo of a sample constant when the pitch is changed. This therefore gives us four main options when we want to start mangling samples:

The first option is to maintain the pitch and maintain the tempo. This is fine... but looked at another ways, since you're not changing anything, you won't notice any sonic differences.

The second option is to change the pitch but maintain the tempo (in other words, maintain the length of the sample). This approach can make a sample playable over a much wider key range (since the notes won't be lengthened or shortened). However, the flip to this is that the character of the note will change (which may or may not be good).

The third option is to maintain the pitch but change the speed of the note—making it longer or shorter. This will have a significant effect on the character of the note, particularly with extreme changes when (especially if the note is slowed) the change will be very noticeable. Even if the listener can't identify what the problem is, they will usually be able to perceive that something isn't right (or that something is wrong but in a good way).

The fourth option is to change the pitch and the tempo. In essence, this is the behavior when a sample is loaded in the normal course of events.

There are many ways you can re-pitch a sample. The most straightforward is to map a single sample across a whole keyboard and play different notes. However, samplers also allow you to retune a sample and you can also set a key center outside of the key range. Each option gives a similar result—the route that's right for you will depend on how you then want to play the sample.

Editing for Effect

The other fast way to mangle a sample is to edit it. These edits can be temporary or permanent—so you can set movable trimming markers or you can destructively edit the wave.

Conventionally you edit a wave to achieve a good result. You make the edits cleanly and, if you are creating a loop, you make sure there are no glitches. However, there are no rules requiring this.

You could include glitches and odd cuts. Equally, you could edit the wave so that it plays well, but cut it so that only a very short section is heard—this way a single sample could create several different tones with different characters.

Sculpting Sounds with Samplers

Very often the key to creating a sound with a sampler is finding the right sample, but beyond that, there is no magic formula to creating a sound. However, I would suggest a few basic steps before you jump in:

Understand what each knob does.

In addition to understanding the function of each knob, understand what it does in the context of any particular sample. Edits will have a different effect depending on what source material they are working on.

Have a definite aim about the sound you're trying to program—don't just fiddle with the knobs and hope something useful comes along.

Starting to Shape

This chapter has looked at the basics of moving from samples to finished sounds. Let's move on and look at how those sounds can be shaped with filters.

Chapter 6: Filters

At its most basic you can think of the filter as being a tone control. A low-pass filter is like a treble control—turn it down and the sound gets "duller". However, a filter can do much more for you.

There are two key differences between a tone control and a filter:

Filters work on a per-note basis to individually sculpt each note. This is important because...

Filters can be animated in real time. If you think of a piano—when a note is struck it is bright, but the tone decays over time. By having filters on a per-note basis, you can hold a note and trigger a new note—the new note will have its own filter giving a much brighter sound which would not be possible if the note needed to use the other decaying note's filter settings.

There are many different types of filters, but the most common is a low-pass filter, so we'll look at the workings of that type of filter first.

Filter Parameters

Filters have two main control parameters: cut-off frequency and resonance, see Figure 6.1.

Figure 6.1 A low-pass filter with a resonant peak.

Cut-Off Frequency

The cut-off frequency is the point at which the filter starts to have effect. So if you have a low-pass filter and set the cut-off frequency to 8kHz, the sound spectrum above 8kHz will be progressively reduced. If you are using a high-pass filter (which is discussed shortly), sounds below the 8kHz cut-off frequency will be reduced.

Resonance

Resonance adds some bite to a filter. It works by boosting the sound spectrum around the filter cut-off frequency as Figure 6.1 illustrates.

Used in moderation, the effect is subtle and can make a sound appear brighter and/or slightly thinner (or less fat if you prefer). When used to the extreme the effect is noticeable—most dance records use filter sweeps with high levels of resonance to exaggerate the effect of the filter sweep. The effect can also give sounds a classic "squelch".

At very high resonance settings, some filters can exhibit quite extreme behavior, so take care with this parameter and turn down the output if you don't want to burn your ears off.

Filters in Samplers

Every sampler (or synthesizer using samples) will have some filter capability.

Kontakt has a wide range of filters on offer, including a number of filters specifically designed to work with samples (see Figure 6.2).

Figure 6.2 The sampler filters available in Kontakt.

In addition, Kontakt has a range of specialist synthesizer filters and effect filters. All of these filters are then fully controllable from Kontakt's interface (see Figure 6.3).

Figure 6.3 The control interface for Kontakt's 3x2 filter.

Specialist filters are made available for a number of reasons: the main purpose is to give different responses—different filters will behave differently and have different peaks and curves, giving a different tonal response. The success (or otherwise) of a specialist filter will depend on its interaction with the underlying sample material.

There is one key difference between how filters are used in analog synthesizers and how they are used in samplers. In analog synthesizers, the characteristic sound you are looking to create is often a product of the interaction between the sound source and the filter which then has its parameters controlled over time. Often that interaction is exaggerated to give the sound movement.

By contrast, samplers usually do not create their sound with a static noise source. Instead, they will often deploy a wide range of highly detailed samples. Accordingly, in order to preserve those tonal nuances, there is less need to deploy such heavy filtering. Indeed, the subtle tonal shifts in the samples will often be lost if the filtering is too heavy.

As a result, when filters are deployed in samplers, gentler filter slopes are usually used (I'll explain these in a moment), less filtering is used, and the resonance controlled is often not touched.

Filter Types

Let's have a look at the main filter types that are usually found in samplers in a bit more detail.

Low-Pass

The low-pass filter (or if you prefer, high-cut filter) allows low frequencies to pass through it, see Figure 6.4.

Figure 6.4 A low-pass filter.

When a low-pass filter is fully open, all frequencies can pass through it.

As the filter is closed, it progressively allows less sound to pass through—you will hear this as the sound becoming duller as the higher frequency elements of the spectrum are filtered out. When the filter is nearly completely closed only the very lowest elements of the frequency spectrum can pass—this sound will be close to the sound of a very low-pitched sine wave.

The effect that the low-pass filter will have on a sound will vary depending on the source material that is fed into it—by its nature it will be more aggressive in cutting bright source material rather than dull sounds.

High-Pass

The high-pass filter (or low-cut filter) is the reverse of the low-pass filter—it allows high frequencies to pass and progressively cuts out the lower frequencies, see Figure 6.5.

Figure 6.5 A high-pass filter.

While high-pass filtering may be noticeable if you play a patch on its own, in a mix any change to the tone of a sound is unlikely to be noticeable except with more extreme cuts. However, the net result of the low end filtering may be to give a cleaner/fuller low end when the bass elements are allowed to come through.

Band-Pass

The band-pass filter acts like a combination of a low-pass filter and a high-pass filter by cutting the frequency spectrum at both ends to only allow a narrow band of sound to pass, see Figure 6.6.

Figure 6.6 A band-pass filter.

The cut-off frequency control determines the center frequency where the full signal is allowed to pass—from that point outwards, the spectrum of sound is progressively cut. The resonance control has the effect of boosting the signal around the cut-off frequency.

At extremes of frequency, the band-pass filter will sound similar to either a high-pass or a low-pass filter.

Band-Pass filters tend to take a lot of energy out of the signal so you may have to boost the level after the filter.

Notch (Band Reject)

If the band-pass filter is the equivalent of burning the candle at each end, then the notch filter—or band-reject filter as it is sometimes called—equates to burning it in the middle. The notch filter cuts the frequencies at its current value, see Figure 6.7.

Figure 6.7 A notch filter.

The notch filter can be used for effect and it can be used surgically in the mix. If you're trying to mix two sounds and they don't sit well together it may be that they're both trying to operate in the same frequency range. In this case, you can "notch out" one of the sounds to allow the other to sit properly—you can do this in your patch design or, as many mixing engineers do, with EQ in the mix.

Comb Filters

Comb filters work by adding a slightly delayed version of a signal to itself. This causes phase cancellations and can give a slightly "chorused" or metallic type of sound.

The spectrum produced by these filters looks like a comb (see Figure 6.8), hence the name.

Figure 6.8 A comb filter.

Filter Slopes

A low-pass filter progressively reduces the volume of a sound above the cut-off point. The rate at which the sound wave is reduced above the cut-off frequency is determined by the slope of the filter. The slope reflects how aggressively the filter attacks the sound it is filtering.

If you have a 6dB/octave filter (which is quite a subtle filter and is sometimes called a 1-pole filter), it will reduce the level of the sound source by 6dB at one octave above the cut-off point, 12dB at two octaves above the cut-off frequency, 18dB at three octaves above the cut-off, and so on.

A 12dB/octave filter (sometimes called a 2-pole filter) will reduce the level of the sound wave's volume by 12dB for each octave above the cut-off point.

A 24dB/octave filter (sometimes called a 4-pole filter) will reduce the level of the sound wave's volume by 24dB for each octave above the cut-off point.

You will see some different filter slopes for a fixed cut-off frequency in Figure 6.9. This is an image drawn by hand and so is not to scale, however, you can see the differing result of the filter slopes. As you would expect, high-pass filters cut at the same rate, but below the cut-off frequency.

Figure 6.9 Different filter curves (note, these are not drawn to scale).

Using Filters in Practice

With analog synthesis, one of the key issues for making a filter sound realistic is the modulation that is applied to it. A filter which is not modulated will be, and will usually sound, static. The human ear is attuned to natural instruments which have a constantly shifting character and will expect similar characteristics in synthesized sound.

In the context of sample-based synthesis, the underlying samples will often be constantly shifting their tone and so any movement of the filter can be confusing or overpowering. However, if the sample is comparatively static, then you might want to add some movement in the filter.

Now you've got an idea about what filters can do, let's move forward and look at how they can be controlled using modulation.

Chapter 7: Modulation: Real-Time and Player Controls

We've already briefly looked at modulation and real-time controls without applying the specific label of modulation. Now we're going to look at modulation in a bit more depth and bring in the additional dimension of modulation in the context of using analog synthesis tools in our samplers.

The principle of modulation is the same whether you're creating sounds with analog synthesizers or sample-based synthesizers (and frequency modulation synthesis too). However, depending on the source material you are working with—in other words, whether you are dealing with a raw oscillator, a complex sample set, or a detailed frequency modulation algorithm—your modulations are likely to be aimed at trying to achieve different ends.

We'll look at the differing considerations in a moment, but first, let's look at the basic principles of modulation.

Modulation: The Concept

The concept of modulation in a sampler is simple—something is changed by something else, and usually in real time.

Conventionally, a "source" will modulate a "destination". So if we think about vibrato, a low frequency oscillator would act as the source and the pitch would be the destination.

If you want more detailed expression, then the extent by which the source modulates the destination can be determined by a "control" (which in turn would be another modulation source, but its modulation destination would be the amount of modulation, rather than the destination).

So staying with our vibrato example, if we have a low frequency oscillator acting as the source to control the pitch of a note, then you could set a modulation wheel as the control to control the amount of vibrato in real-time.

Modulation in the Real World

Before we get to the detail, let's think about modulation in the context of an acoustic instrument such as the piano. Taking an over-simplistic view, the factors that have an effect on the sound are:

velocity

pitch, and

time.

Effect of Velocity on a Sound

With a piano, the velocity (the loudness) of a note has an effect in three main areas:

the volume—there is a direct link between how hard a note is struck and its volume

the tone—the harder a key is hit, the brighter the tone of the resulting note, and

the sustain time of the note—the louder the note when it is struck, the longer it will sustain.

Effect of Key Pitch on a Sound

Clearly the piano keyboard controls the pitch of the note played. However, there are other factors affected by the pitch:

the tone—the higher the note, the brighter it sounds, and

the sustain time of the note—the higher the note, the less time that it sustains.

Effect of Time on a Sound

The length of time that a note is held is also a factor in the tone of a piano. Over time:

the sound will get duller (however bright it may have been initially), and

the volume will decay until it finally reaches zero.

Combining the Effect of Velocity, Pitch, and Time

A piano note is far more complex than may be suggested here, however, in broad terms you can see that:

The initial volume of the note is affected by one factor only—velocity. After that, the player has no control over the loudness of the note as it decays naturally.

Tone is affected by three factors—the initial velocity of the note, the pitch of the note, and the length of time it is held. However, the tone increases in line with the increase in pitch (so the higher the note, the brighter it becomes), but the sustain time decreases in line with the increase in the pitch (so the higher the note, the less time it sounds for).

Sustain time (and also volume over time) is affected by two factors—the starting volume (which is in turn controlled by the initial velocity) and the pitch.

In the context of sample-based synthesis, we replicate these nuances by capturing and deploying a wide range of samples. However, there are still times when we want to mimic the characteristics of an acoustic instrument and our samples aren't up to the task. In these instances we're going to need to use modulation to control the sampler.

Differences between Analog Synthesis and Sample-Based Synthesis

The principle of modulation is the same whether you're creating sounds with analog synthesizers or sample-based synthesizers. However, depending on the source material you are working with your modulations are likely to be aimed at trying to achieve different ends.

With an analog synthesizer there is a single sound source—an oscillator. Once the waveform has been selected, the oscillator's output will not vary. Therefore to achieve any movement in the sound—in particular, to achieve tonal changes to account for changing velocity and to mimic the natural decay of a note—the analog synthesizer will bring several tools to control the cut-off frequency of the filter. These tools will make these adjustments both automatically and in response to the player's technique in real-time.

By contrast, with a detailed sample set, there will be many velocity layers and the sound will change over time since that changing tone will have been sampled. Therefore, with a detailed sound set there is usually no necessity to deploy a filter.

However, that is not to say you would never want to use a filter (as an example) in sample-based synthesis. Indeed, there may be many times when you want a filter, for example:

If you have a limited sample set—perhaps you have few velocity layers, each note has a wide key range, or your samples have been truncated and replaced with loop sections—then you will probably want to use a filter (in conjunction with your samples) to mask the shortcomings of your source material.

When you are bringing different sample sets together—whether you are layering different instruments or placing different playing styles in different velocity layers—you may want to smooth the transitions. Also with the layers you may find that you can make two sample sets sit together better by cutting the top end from one group and the bottom end from another and the two may fit.

You can use a filter for radical shaping of samples, but in practice, you're much more likely to use it as a tool to polish and enhance your source material, and any modulation of the filter will be far more gentle than would be the case for an analog synthesizer.

There are other modulations where you might be as radical with samples as you would with an analog synthesizer.

For instance, if you're using a volume envelope, you might make quite radical changes and use an envelope instead of editing a sample (or sample set). The benefit of using an envelope is that you can modify all samples together—you don't need to edit each sample individually.

Another place where you might get quite radical is with the LFO.

If you want vibrato, then you can sample the technique. However, in real life, a player will subtly adjust the frequency and the depth of the vibrato in real-time. These variations can be sampled, but would require a huge and complex sample set which would then be fiendishly difficult to deploy (since changing between vibrato depth and vibrato speed is hard to do while sustaining a sampled note).

As much easier approach is to use an LFO which allows the player to control the frequency and depth of the modulation in real-time.

Another aspect to remember is that sample-based synthesis is different from analog synthesis and as a result, there will be parameters that can be modulated in sample-based synthesis that aren't available in analog synthesizers.

So now we know that there are differences, let's move on and look at the basic principles of modulation and how modulations can be implemented in sample-based synthesizers.

Modulation Destinations

We will approach modulation in a slightly backwards way and look at modulation destinations before we look at modulation sources. Once we understand what can be modulated, we can work back to choose the best source to create the modulation.

I should point out that the following list of modulation destinations is, of course, not complete: it only covers the main destinations that you are likely to want to control. Different samplers will offer different amounts of control, so on any specific sampler you may be able to modulate more or fewer functions than are set out here.

Volume

Volume is likely to be one of the main destinations you modulate. You may want to:

control the volume of your whole sound—for instance to make the sound touch sensitive or to add a tremolo effect, and

control the volume of an individual element (for instance, if you are layering two sound sets) so you can cross-fade between two sounds.

Filter

In the context of sample-based synthesis, the main modulation destination in the filter block is the cut-off frequency. There may be a reason from time-to-time why you would want to modulate the resonance, but mostly, you will want to control the cut-off frequency to make a sound brighter or duller.

Pitch

The effects that can be obtained by modulating pitch include vibrato and trills as well as weird special effects.

Vibrato Speed and Depth

When a musician applies vibrato to a real instrument, both the speed and the depth of the vibrato increase—it therefore make sense that LFO speed should be a modulation destination (and the depth of the effect should also be controllable).

Sample Start Point

A sample doesn't need to start being played from its start and you may want to control the position from which it starts playing.

For instance, if you have a picked guitar or bass sound, you may want to control whether the sound begins before or after the picking. This approach could be applied to a number of sounds. As another example, you might control how much crack is heard when a drum is struck.

Pan

The classic use for panning is auto-pan type effects where the sound moves regularly between the left and right channels. There are however more subtle uses of panning, for instance panning can be related to the pitch of a note with low notes panned towards the left and higher notes towards the right as if you were sitting in front of a piano.

Envelope Speed

I'll talk about key tracking (where behavior is linked to pitch—for instance, in the way that higher notes sustain for a shorter period) in greater detail in the next section (Modulation Sources). One frequent modulation destination for key tracking modulation is the envelope, and in particular, the speed of the envelope where key tracking can give the sound a more natural quality. However, the envelope can be modulated for other reasons, for instance, the attack time of a note can be controlled by changing an envelope.

FX

The level of any FX units can be modulated to give prominence to an effect. If you want to get more complicated, you may want to change FX controls in real time—for instance tweaking the delay time.

Modulation Sources

Most samplers offer and array of modulation sources, each of which operates in a different manner. There are also external sources which can act as modulators—such as MIDI velocity, the pitch bend wheel, and the modulation wheel. These sources can then be used directly as modulation sources or as controllers to control the amount by which another modulation source is modulating a destination.

The modulation sources (or controllers) fall into four main categories:

Single-shot sources. An example of a single-shot modulation controller is an envelope—it begins, executes its level changes, and then ends (after perhaps sustaining for a while). Envelopes are discussed in greater detail later in this chapter.

Continuous sources. An example of a continuous source is a low frequency oscillator—the modulation source repeats its cycle continually modulating the destination. LFOs are also discussed in greater detail later in this chapter.

Fixed. Some modulation sources are fixed once a note has been played. The main example of this is velocity—once a note has been struck, the velocity is fixed. In the context of sample-based synthesis, these fixed modulations are often used to switch between sample sets or layers within a sample set.

Variable. Other modulation sources are constantly variable after the note has been struck. One example of a variable source is the modulation wheel which can be continuously tweaked in real-time.

A second aspect to think about in relation to modulation sources is polarity:

Unipolar. Some sources (for instance velocity, the modulation wheel, and conventional envelopes) are unipolar. This means that their output within a single cycle is either positive or negative.

Bipolar. By contrast, bipolar sources (such as LFOs and the pitch wheel) are capable of generating both positive and negative signals within the same cycle.

Let's have a look at some of the modulation sources in a bit more depth. I've excluded two very common modulation sources—envelopes and LFOs—from this list as we'll look at them in greater detail immediately after this list.

Velocity

When creating sounds, velocity is often used to control:

the loudness of a note and

the tone of a note.

In the context of sample based synthesis, these nuances can be in two ways:

first, by selecting a different sound set, and

second by modulating the filter cut-off frequency.

Velocity can also be used to control the envelope attack time where the envelope then controls volume or tone. With this modulation, you could use velocity to determine whether a sound begins immediately or gently fades in.

Velocity can either be applied directly to modulate the destination, or it can be used as a modulation source controller to control another source (for instance, an envelope parameter) which, in turn, is modulating the destination.

Switches

There are times when you will want to switch between different sound sets (in other words, to call up different groups of samples in real-time. There are two main switching tools:

Specific notes or specific ranges. If you trigger a certain key, that will call up one sample set but when you trigger another specified key, it will call up a different sound set.

MIDI CC (MIDI control change) messages can also be used as switches. Usually when a MIDI CC message is between two specified settings, then a corresponding sample set will be called up.

MIDI Control Change messages can be used as a source or as a source controller.

Pitch Bender

The pitch bender is often hardwired to the pitch bend function. However, most samplers also allow the pitch bend signal to be assigned to control other functions.

Modulation Wheel

Conventionally the modulation wheel is wired to the vibrato depth and so acts as a modulation controller.

However, it doesn't have to be so—the modulation wheel can also be used as a modulation source (for instance, to control the filter cut-off frequency directly).

Key Tracking

Key tracking is the process where changes are made according to the pitch of a note.

For an example of key tracking, if you pluck the bottom E string of an acoustic guitar, the sound will last for over 10 seconds. However, if you play the top E string at the highest fret, the sound will barely last for 3 seconds. With samples that detail is automatically picked up through the sampling process. However, if your samples are looped (as in there are sample loops meaning that a note can sustain infinitely—I'm not talking about beat loops here) then you might want to use key tracking to control the length of an envelope to reflect the correct behavior of the acoustic instrument.

Most samplers allow you to set pitch as a modulation source. This will then apply the modulation in a linear fashion across the keyboard and key tracking can be applied in many ways—for instance, the cut-off point of a filter can be controlled by key tracking so that higher notes are brighter than lower notes.

The most obvious use for key tracking is to get the pitch of notes right—in other words, you want to know that when you hit a C key that a C note will sound or when you hit an E key that an E note will sound, and so on. This implementation of key tracking is so obvious that we often don't think about it. However, particularly with percussion, but also when you're getting creative, this assignment may not always be appropriate.

Aftertouch

If your keyboard (or other MIDI controller) gives aftertouch messages, then the pressure on the keys while a note is sustained can be used to control your patch. You could, for instance, use aftertouch to control volume to give a swell to a held string chord.

Aftertouch can either be applied directly to modulate the destination, or it can be used as a modulation source controller to control another source which is modulating the destination.

Expression Pedal

Often the expression controller is hardwired to a synthesizer's volume. As with the modulation wheel, it can also be used as a modulation controller.

Envelopes

One of the main modulation sources are envelopes. Envelopes can have a dramatic effect on a sound, and in particular, they can have a significant impact on a sound's character.

So what is an envelope? Well, in short, it is a level that changes over time. Envelopes don't make any sound, but what they do is to change other parameters that affect the sound—such as the pitch, the filter, and the volume—as the note is played. This in turn both animates and controls the sound of the underlying sample.

Envelopes are a one-shot modulation source. When triggered, the envelope begins, executes its level changes, and then ends. If you want continuous modulation, then rather than choose an envelope, an LFO (which is discussed in the next section), may be a better choice of tool.

Volume Envelopes

The most immediate use for envelopes is controlling volume.

Think of a note played on a piano. When a key is struck, the note goes from silence to the maximum volume instantaneously. From this peak, there is an immediate reduction in the volume and then the note reaches a level from which it gradually fades to nothing. A picture of the volume of a piano note over time is set out in Figure 7.1.

Figure 7.1 A (simplified) volume envelope of a piano note.

Now if you think about a violin note which slowly fades in, stays at its maximum volume until the note ends, and then gently decays, this volume level may look like the image in Figure 7.2.

Figure 7.2 A typical sustained string envelope.

Volume envelopes are captured when you capture a sample, so when you're dealing with detailed sample sets you don't necessarily need to pay so much attention to volume envelopes.

However, there will be many times when you want to use an envelope in conjunction with samples, for instance, if you want to smooth off the start of a sample to give the sound a slower attack. Alternatively, you might want to fade the level of a sample set over time so that it works together with another sample set.

One thing to remember about a volume envelope is that its effect is to cut the volume—it cannot compensate for elements that were not captured in the original sample. So for instance, take a look at Figure 7.3 which shows a sample in a wave editor (that's the blue bit) with an intended envelope overlaid (show in red).

Figure 7.3 A piano envelope laid over a sample that fades in. The likely effect of this envelope would be to weaken the sample by cutting the peak at the end of the fade—it wouldn't create a fast attack time.

In this instance, the envelope will not convert the sample from having a slow fade-in to having a fast attack. To create that sort of change you need to edit the start of the sample (to cut the fade-in).

What Else Can Envelopes Do?

An envelope changes a level over time, but its effect will depend on how it is applied. For instance:

An envelope could control volume. Depending on the architecture of the sound, the envelope may control the level of a group of samples or the level of a whole sound.

An envelope may control a filter. If an envelope controls a filter, it will (usually) control the cut-off frequency and so change the brightness over time (usually making the sound duller over time, in other words, the sound will start bright and then get duller).

Different synthesizers are designed in different ways and have different envelope options. Let's look at some of the more common types of envelope.

ADSR Envelopes

The ADSR envelope (see Figure 7.4) is the classic synthesizer envelope. A variant of this envelope is used on virtually every synthesizer and sampler.

Figure 7.4 The ADSR envelope.

There are four main controls in the ADSR envelope:

A: Attack time. The attack time governs the time it takes for the level to reach its maximum after the note is triggered. Using the example of a piano, the attack time would be zero—in other words it would take no time for the level to go from silence to the maximum volume. For a string type sound the attack may be slower.

D: Decay time. The decay time controls how quickly the level drops (to the sustain level) after it has reached its maximum at the end of the attack phase. Again, using the example of the piano, the decay time would be fast, but it would be longer than the attack time.

S: Sustain level. The sustain level is the level of the envelope while a key is held. This level stays constant until the key is released.

R: Release time. The release time is the time it takes the level to decay to zero after a key is released.

You will notice that with this envelope:

Once the sustain part of the envelope has been reached the note sustains indefinitely—it does not decay to zero until the key is released. However, if your sample has a finite length (for instance, if you have sampled a piano and the sample does not include a loop section), then if your sample ceases to sound/reaches its end, the effect of the envelope will be irrelevant (since it won't have any source material to work on).

There is no function in the envelope to determine for how long the note sustains (the only control over this is by releasing the key). Following the point above, the underlying sample can still have an effect on how long the sound will be heard for.

Another weakness with this type of envelope (and this applies to most envelopes) is that real sounds do not necessarily increase or decrease in a linear manner. Take the example of a slow swelling violin—in practice the attack of the note is likely to have two phases:

first the note will go from nothing to a very quiet level very quickly, then

the note volume may increase more slowly.

This behavior is not easy to replicate with a synthesizer envelope.

The ADSR envelope is the "standard" envelope. Let's move on and look at the envelopes that are available in Kontakt. As you will see, they are similar to the ADSR envelope, but there are differences and these differences are included so that the envelopes can be used in specialized situations.

AHD Envelopes

The AHD (attack-hold-decay) envelope is intended for use with percussion samples. It is a comparatively simple envelope making it fast to use, but it balances the simplicity with enough power to control samples. Figure 7.5 shows the AHD envelope in Kontakt.

Figure 7.5 The AHD envelope section in Kontakt.

There are three controls in the AHD envelope (see Figure 7.6):

Figure 7.6 The AHD envelope.

A: Attack time. The attack time is the time it takes for the level to reach its maximum after the note is triggered.

H: Hold time. The hold time is the time—after the level has reached the maximum at the end of the attack phase—for which the level remains at the maximum. This setting can be useful for giving emphasis to percussive elements.

D: Decay time. The decay time controls how quickly the level drops (to zero) from the maximum level at the end of the hold phase).

AHDSR Envelope

The AHDSR envelope (see Figure 7.7) is the main workhorse envelope in Kontakt.

Figure 7.7 The AHDSR envelope in Kontakt.

There are five main envelope controls in this envelope (see Figure 7.8):

Figure 7.8 The AHDSR envelope.

A: Attack time. The attack time is the time it takes for the level to reach its maximum after the note is triggered.

H: Hold time. The hold time is the time—after the level has reached the maximum at the end of the attack phase—for which the level remains at the maximum. If the hold time is set to zero, then this envelope behaves as a conventional ADSR envelope.

D: Decay time. The decay time controls how quickly the level drops (to the sustain level) from its maximum (at the end of the hold phase).

S: Sustain level. The sustain level is the level of the envelope while a key is held. This level stays constant while the key is held.

R: Release time. The release time is the time it takes the level to decay to zero after the key is released and the hold time stage has completed.

DBD Envelopes

The DBD (decay-break-decay) envelope (see Figure 7.9) is another specialist envelope intended for pitch modulation. It can be used for other purposes, but its intended use is pitch modulation.

Figure 7.9 The DBD envelope section in Kontakt.

There are three controls in the DBD envelope (see Figure 7.10):

Figure 7.10 The DBD envelope.

Decay 1 time. The decay 1 time is the time it takes for the level to reach the break point level after the note is triggered.

Break point level. The break point level determines the maximum effect of the envelope. If the envelope is used for modulating pitch, the break point level can be set to a maximum of plus or minus 36 semitones (in other words, three octaves).

Decay 2 time. The decay time controls how quickly the level return to zero from the break point level.

This envelope is unipolar—in other words its effect is either positive or negative, not both within one cycle. However, you can set the break point level to be a positive or a negative value in order to achieve positive or negative pitch modulation.

Multi-Stage Envelopes

The envelopes we have looked at so far have all been fixed: you get a certain number of parameters that you can adjust, and that's it. This approach works well—it is simple and straightforward to understand.

Kontakt also offers a multi-stage envelope where you can add as many breakpoints as you want. For each breakpoint you can set the level, set the time from the previous breakpoint, and you can adjust the curve. If you want to add a loop section, you can. If you want to synchronize the envelope (in other words synchronize the breakpoints) with the beat of a track, you can (and you might want to do this to create rhythmic envelopes).

With this envelope you can recreate an ADSR envelope (as Figure 7.11 illustrates) or create something far more complex.

Figure 7.11 A multi-stage envelope in Kontakt—in this instance mimicking an ADSR envelope.

This envelope offers the musician a detailed level of control and appropriate feedback about the settings. In Kontakt the envelope is also comparatively intuitive to use:

To add a breakpoint, right-click (ctrl-click on a Mac) where you want to add a node (breakpoint).

To remove a breakpoint, right-click (ctrl-click on a Mac) on it.

To move a node, click-and-drag it. If tempo sync mode has been engaged, then the node will snap to the nearest beat divider.

To change the slope between two nodes, click on the blue circle at the midpoint between two nodes and drag it.

The envelope will sustain while a key is held. As the envelope sustains, its level will continue to loop, following the level between the two vertical orange lines. These vertical orange lines can be dragged to change the loop start and end points (and you can have nodes inside and outside of these sustain points).

When you click on a breakpoint, the selected node and the total number of breakpoints is shown in the top row (next to the # sign). For that selected node, the row gives details about its setting and some further control:

The Mode selector determines how the other breakpoints are affected when that node is moved. If SLD is selected, all breakpoints to the right of that node are shifted when it is shifted. If FIX is selected, only that breakpoint is moved and every other node remains fixed.

Abs Time displays the time (in seconds) from the start of the envelope to the selected node. This is a display which cannot be adjusted.

Delta Time shows the time (in seconds) between the selected and previous nodes.

Level shows the level of the node.

Slope displays a value for the curve.

Envelope Curves

I have mentioned envelope curves a few times.

If you check back to Figures 7.5 or 7.7 you will see a curve control. With most envelopes the changes are linear, in other words, there's a direct line from one breakpoint to another.

The curve control allows you to adjust the transition of the attack phase. You can slow the curve so that the initial changes in the level are minimal but become more dramatic during its course or you can set a fast curve which makes a large proportion of the level change in the initial stages, but gets slower as it reaches its destination level. And you can set the control anywhere between those two extremes to get the right transition that works with the underlying samples.

In Kontakt's multi-stage envelopes, each segment can have its own curve.

LFOs and Specialized Modulation Sources

A low frequency oscillator (or LFO as it is more frequently called) is a level that changes over time. As with envelopes, LFOs don't make any sound, but what they do is to change other parameters that affect the sound—such as the pitch, the filter, and the volume—as the note is played.

In addition to working as a vibrato source, the LFO is also often used in sample-based synthesis to modulate volume, to give tremolo effects. When trying to create synthesizer-like sounds, an LFO may also be used to modulate the filter cut-off, to give wah-wah or gating/stuttering type effects—especially when a square or pulse wave is selected as the modulation source.

You can, of course, modulate many other destinations with an LFO.

Envelopes are a one-shot modulation source. When triggered, the envelope begins, executes its level changes, and then ends. By contrast, an LFO is an oscillator and so repeats its cycle continually modulating the destination. Because of this repetition, the effect of LFO modulation is often more prominent in sounds and therefore needs to be controlled more carefully.

Main LFO Controls

The main controls on any LFO are wave shape and depth.

Wave Shape

Most samplers offer several LFO shapes; these are usually the same basic wave shapes that are available for audio oscillators. The most frequently used LFO shapes are:

Sine. The sine wave (see Figure 7.12) produces an even, rounded modulation source.

Figure 7.12 The sine wave.

Triangle. The triangle wave (see Figure 7.13) has virtually the same effect as the sine wave, but the changes are constant. This may give a crisper sound when compared to the sine wave. Both the sine and triangle are ideal waves for vibrato, tremolo, or wah-wah effects.

Figure 7.13 The triangle wave.

Sawtooth. Depending on the polarity of the wave, the sawtooth provides either a rising modulation source which drops abruptly after reaching the maximum value or a falling modulation source, swiftly rising to a peak and then gently falling (see Figure 7.14). The wave is often used for modulating the filter cut-off.

Figure 7.14 The sawtooth wave—a rising sawtooth on the left and a falling sawtooth on the right.

Square. The square wave (see Figure 7.15) is either at its maximum or its minimum. This wave tends to be used to give rhythmic effects. It can also be used to modulate the pitch to give trill type effects. The pulse wave is similar to the square wave, but the two sides of the wave are not balanced.

Figure 7.15 A square wave (on the left) and a pulse wave (on the right).

Random. Some samplers offer a random or more commonly, sample-and-hold wave (see Figure 7.16). The output from the a random wave is a random value, and for a sample-and-hold wave, the level of the wave steps from level to level—there is no smooth transition meaning that this is an excellent choice for creating rhythmic effects.

Figure 7.16 A sample-and-hold oscillator which is especially useful for creating rhythmic effects.

These are the most common shapes—many samplers offer a much wider range of options.

Frequency

The frequency control affects the speed of the LFO. If an LFO is used as a vibrato source, then the frequency control will affect the speed of the vibrato. Most LFOs have a range of around 0.1 cycles per second to about 20 cycles per second.

It is also very common to allow an LFO frequency to be synchronized with the beat of a track so that the time it takes the LFO to complete a whole cycle is a subdivision of the track's tempo. This option is often used in conjunction with rhythmic effects.

Depth

The depth setting of an LFO controls the extent to which the LFO affects the modulation destination.

When using an LFO to introduce vibrato you will probably have to apply less vibrato than a player would apply to a natural instrument. LFO vibrato usually has a fixed frequency (but this too can be modulated) and so the effect will quickly sound mechanical. Reducing the depth of the effect helps to stop some of the mechanical feel becoming too prominent.

This setting needs to be carefully balanced so that an appropriate range of its effect is controlled. For instance, if you are using an LFO to give some vibrato, you probably want a range which is less than (plus or minus) one semitone.

Delay or Fade In

Delay or Fade In is the time it takes from when the note is struck for the LFO's depth to reach its full effect.

Sync

When a note is triggered, any LFO that is applied to that note could be at any stage of its cycle. Usually that's not a problem—indeed, this random element (with a free running LFO) may be desirable for ensemble type sounds—however, there will be times when you want to control where the LFO is in its cycle at the point when a note is triggered. For instance, if the LFO is modulating pitch, then you are likely to want to know—with certainty—the starting pitch when the note is triggered.

Sync is also an issue when you start adjusting the phase of an LFO.

Phase

When we talk about the "phase" of an oscillator we generally mean the position in a wave's cycle. For any control over an LFO's phase to be meaningful, you will want to synchronize your LFO (otherwise you're simply adjusting something that is random in order to create something that is still random, but random in a different way).

By combining sync and phase, you can set an LFO to start playing at any specific point in its cycle when a note is triggered.

Monophonic/Polyphonic LFOs

LFOs can be monophonic or polyphonic.

Monophonic LFOs. With monophonic LFOs, a single oscillator is used whenever that LFO is invoked. So if you are using an LFO to add some vibrato to a string section, when you play a chord all of the notes will have the rise and fall of their pitch synchronized (since all notes will be modulated by one LFO). The effect may not be particularly natural in this instance.

Polyphonic LFOs. With a polyphonic LFO, each separate note has its own LFO which will start its cycle at a different time from any other LFO (assuming that you haven't engaged the sync function, which would defeat the purpose of using a polyphonic LFO). Taking our string section example again, using a polyphonic LFO to modulate a chord will mean that some notes will have their pitch raised while others are having their pitch flattened. This may give a more desirable result for a string section, but is less use for rhythmic sounds.

Specialized Modulation Sources

Beyond the LFOs, Kontakt has two specialized modulation sources.

Envelope Follower

The envelope follower (see Figure 7.17) takes the level of the currently playing sample and converts it into a level that can be used as a modulation source.

Figure 7.17 Envelope follower in Kontakt.

There are four controls in the envelope follower:

Attack. The attack control sets the time it takes for the envelope follower to react after the level increases. With higher values, the response is smoothed out.

Decay. The decay control sets the time it takes for the envelope follower to react after the level falls.

Gain. The gain control adjusts the sensitivity of the unit to the incoming audio signal.

Adapt. The adapt control adjusts the response for the envelope follower.

Step Generator

The step generator (see Figure 7.18) is a modulation source with a number of steps (up to 32). Each step is evenly spaced (which makes this tool particularly effective when the steps are synchronized to the beats of a bar.

Figure 7.18 Step generator in Kontakt.

The level in each step can be set individually (hence each beat can have an individual level reinforcing the rhythmic element). The step generator is then attached to a modulation destination (typically a filter cut-off or a volume amplifier) and off you go.

Setting Up Modulations

Every sampler will set up its modulations in a different way. While the individual knobs to be clicked will change, the basic process is usually similar: select the modulator, connect it to the destination, adjust the amount of modulation, and then add a control (if necessary).

By way of illustration, let me show you the process in Kontakt to modulate the cut-off frequency in a filter with an envelope. The process goes something like this: First, add a filter (which in the case of Kontakt means making decision about the filter type and slope and loading the filter as an FX unit) and set its basic parameters. In Figure 7.19 you can see that I have selected a 2-pole low-pass filter.

Figure 7.19 Setting a filter in Kontakt.

With the filter set, the next step is to call up a modulator—in this case an envelope—by right-clicking on the control to be modulated (in this case the cut-off frequency) and selecting the appropriate envelope from the drop-down menu (see Figure 7.20).

Figure 7.20 Selecting the modulation source—in this case an envelope.

The envelope will then appear in the modulation section at the bottom of the interface (where you can adjust its settings, see Figure 7.21). You can also see that under the filter settings there is a new Mod row added where you can adjust the depth of the modulation.

Figure 7.21 The modulation envelope and the modulation parameters.

This modulation assumes a positive polarity (in other words, it assumes that you want to open up the filter with the envelope). If you want to close the filter (so that the sound gets duller rather than brighter) click on the Invert button on the modulation depth parameter bar.

You could take this set-up a stage further and add some velocity control so that at low velocities the envelope will have little effect, but at high velocities, the envelope will have a significant effect on the filter's cut-off. In the envelope in the modulation section, click on the mod button which will reveal an add modulator drop-down menu. From this drop-down, select velocity which will give you a row of parameters immediately under the envelope (see Figure 7.22) with a slider to set the depth.

Figure 7.22 Adding velocity modulation to the envelope.

In this example I have used MIDI velocity as the control. You could, of course, use any other source—and in particular any external MIDI hardware controller—to control your modulation. Equally, the controller can be applied directly to control the modulation (so in this example, you could set the modulation wheel to control the cut-off frequency instead of using an envelope).

If you want to remove this modulation, you can right-click on the envelope and choose delete.

Kontakt has one approach to apply modulations—Battery uses another approach. As you can see from Figure 7.23, Battery has a modulation page which applies the modulations on a per-cell basis. As well as having two LFOs and an envelope, you will see that there is the modulation router section where sources and destinations can be set, and the depth of the modulation can be readily controlled.

Figure 7.23 The Modulation page for each cell in Battery.

Now Polish!!

We've now looked at applying modulation to our sample-based sounds. The final stage in polishing the finished product is to add some effects which we'll look at in the next chapter.

Chapter 8: FX

We've looked at the elements used to create a sound using sample-based synthesis techniques. Now we'll look at how to polish and control those sounds and how to extend your sonic palette further using effects units.

The FX units you find on today's samplers are generally used in two ways:

as an integral part of the sound for instance, a distortion unit, or

to enhance a sound, for instance, reverb to give a sound some space and depth or to make up for any shortcomings in the source material.

We are going to look at both options for shaping your sound.

Deployment of FX

FX units are deployed in one of two ways: as insert FX units or as send FX units. There is a practical difference between the two.

Insert FX

With insert FX the whole of the audio signal passes through the FX unit (see Figure 8.1).

Figure 8.1 The signal flow through an insert FX unit.

If you want to add a distortion effect, then you are likely to want the whole signal to be distorted by the fuzz box. EQ and compression are other typical examples of insert FX.

Master FX

Some samplers offer master FX units. There is nothing intrinsically different about these units or the signal architecture: they are insert FX units inserted immediately before the sampler's output so that the processing is applied to the whole signal.

Send FX

With send FX, you send part of your signal to the FX unit and then mix an effected signal back with the dry sound (see Figure 8.2).

Figure 8.2 The signal flow with a send FX unit.

Examples of FX units that are grouped under the send FX heading are modulation effects (chorus and so on), delays, and reverbs.

Kontakt offers a row of slots for insert and send FX units. It also has a row of slots for Group insert FX (see Figure 8.3). This allows you to apply an insert FX unit to a sub-group of samples within a sample set.

Figure 8.3 The FX slots in Kontakt.

Battery offers an insert FX section for each cell (see Figure 8.4).

Figure 8.4 The insert FX pane for each cell in Battery.

To the right of the pane are two sends: Delay and Reverb. These send FX units are located on the master pane (see Figure 8.5) which—apart from the delay and reverb—contains a number of master FX units.

Figure 8.5 The Master pane in Battery where the master FX units are located.

Distortion

Most people are used to hearing distortion used in an extreme way. However, depending on the amount of control you are given over the distortion, it can be used in a much more subtle manner.

In the context of sample based synthesis, gentle amounts of distortion can bring some coherence to a sound, taking it from sounding like a bunch of samples to feeling like a real instrument played in a real context.

However, gentle—virtually imperceptible—distortion is not the only option and you will find that distortion units come in different flavors—in broad terms, digital distortion and analog-emulating distortion.

Old School Overdrive and Analog-Emulating Distortion

Analog-emulating distortion is (usually) designed to give a gentle overdrive. At lower levels its effect is heard as a warming/fattening of the sound, rather than as distortion.

Most distortion units have fairly simple controls, usually with only a drive control and occasionally with a tone control.

As an example, Kontakt offers several varieties of this kind of unit with increasing levels of destruction: Saturator, Distortion, and Skreamer (which offers a wide range of tone and distortion controls).

Digital Distortion

By contrast to analog-emulating distortion, digital distortion is (often) quite harsh and cold, lacking the warmth/fattening of digital distortion. You will find these units with a number of labels; typically they are called digital distortions, decimators, lo-fi, bit crushers, and sometimes alias units.

To understand what these units are trying to achieve, you need to understand digital audio. We covered this in chapter one, but let me recap. There are two factors that determine the audio fidelity of a waveform generated by a sampler:

the bit rate—the number of times that the signal is generated per second, and

the sampling rate—the number of levels of loudness that are allowed.

The two factors in combination affect the quality of a digital waveform. Take audio on a CD as a practical example of this: the sound you hear has 65,536 (2^16) levels of loudness and each level of loudness is recalibrated 44,100 times per second. If you reduce the bit rate and the sampling rate of a CD, then the quality would fall as Figure 8.6 shows.

Figure 8.6 As the bit rate and/or sample rate are cut, the resolution of the wave—in other words, its audio fidelity—will fall. This drop in fidelity is perceived as distortion which, with care, can be controlled for creative effect.

Digital distortion units will allow you to control (within the FX unit) either or both of the bit rate and the sampling rate, and sometimes also the tone of the output. Generally the distortion that these units create is pretty harsh and hard to control, but if you want a really brash overdrive, these are the units to head for.

If you do want more control (and your sampler allows you to move the effect within the signal chain) then put the digital distortion unit before a filter, so that the filter can smooth off some of the rough edges.

Compression

Compressors were originally used to prevent an analog signal overloading an analog input, and indeed, compressors still are used for this purpose (and you may use hardware compressors when you are capturing your raw samples).

In essence, a compressor is an automatic gain control: when a signal gets to a certain level the compressor restricts how much louder the signal can get (see Figure 8.7).

Figure 8.7 A compressor controlling an input signal.

Most compressors allow you to control:

Threshold. The threshold is the level above which a signal is attenuated.

Ratio. The compression ratio controls the extent to which a signal above the threshold is allowed to increase in level. So for instance, with a 4:1 ratio if the input signal increases by 4dB above the threshold, the output signal will only increase by 1dB above the threshold.

Gain. When a signal has been compressed, it will be quieter than the input signal. The make-up gain allows the output level to be rebalanced to an appropriate volume.

In practice, you're quite likely to want to use the compressor in a number of ways including:

to catch some of the loudest peaks

to make a sound rounder and/or smoother

to enhance the perceived loudness of the sound, and

to give the sound more punch.

The disadvantage of using a compressor is that the resulting sound can tend to dominate a mix. With more extreme compressor settings, the sound will usually have a more uniform level and will fill a broader frequency range. Both of these factors can make it harder to mix a highly compressed sound. Another downside of compressor abuse is that the resulting sound tends to be less interesting—the delicate harmonic shifts of a detailed sample set can get covered up as all of the harmonics take a similar level.

You may also come across limiters. In simplistic terms, these are compressors with a high ratio (often fixed). With a high ratio, limiters prevent a signal from passing a certain level.

EQ

The purpose of equalization is to make certain elements of the sound spectrum louder and other elements quieter. You can use EQ creatively or surgically. However, remember that you still need to get your sound to sit in a mix, so go carefully with the EQ, and try not to use the effect to cover the faults of poor samples (unless you really need to...). Also, remember that you may not hear the full effect of any EQ tweaks unless you are using full-range speakers.

The most flexible form of EQ is the parametric EQ where you can set:

the frequency to be boosted or cut,

the amount of boost or cut, and

the bandwidth (or Q) of the boost or cut. The bandwidth determines how wide the boost or cut is, much like a constantly variable slope control for a filter.

A simplified form of EQ is the shelf EQ. A shelf EQ will cut or boost:

the whole signal above a specified frequency in the case of a high shelf, or

the whole signal below a specified frequency in the case of a low shelf.

Take a look at Figure 8.8 which shows a low shelf EQ (on the left), a high shelf EQ (on the right), and a parametric EQ in the middle.

Figure 8.8 A low shelf, parametric, and high shelf EQ.

Filter

We looked at filters in Chapter 6: Filters.

Many samplers include filters in their FX sections—these usually operate in the same manner as those described in Chapter 6, however, there is one other aspect: the location of filters in the signal chain. Often you can place them at a more appropriate point than the main filters, for instance, you can place the FX filter immediately after a distortion effect.

Modulated Delay Effects

Modulated delay effects take many different forms (chorus/flanger/phaser, and so on) but they all do a similar thing: in essence they mix a delayed, pitch-modulated signal with the original signal to give a fuller/warmer/richer/whooshier (insert whatever description you want) sound. The delay time for the delayed signal is so short that you don't hear it as an echo, simply as a cohesive changed tone.

While the operating principles are the same, the sound the different units produce can be markedly different. Although the delays are so short the delayed signal cannot be separately identified, the delay time has a major effect on the sound.

Used sparingly, these effects can give an element of spark or richness to a sound. Used to excess and you will create a sound that quickly becomes passé.

Phaser/Flanger

Words cannot readily describe the difference between the sound produced by a phaser and the sound produced by a flanger, however, to my mind a phaser gives a more nasal sound where a flanger gives a more whooshy type sound. One of the reasons for the phaser sounding more nasal is that the delayed signal passes through a notch filter (which has its frequency modulated).

Flangers and phasers are often used on percussion sounds and with plucked-type sounds (such as guitars and electric pianos). The sound is very distinctive and is hard to use in a subtle way, so you are likely not to want to use the effect too frequently. The sound also has a 1970s retro quality to it—whether you love or hate that era is likely to have an influence on how you feel about phasing and flanging.

The main controls you are likely to find on a phaser or a flanger may include:

Delay. The delay control will adjust the time of the delayed signal.

Feedback. The feedback control sets the number of times the delayed signal is repeated. This has an effect on the perception of depth of the effect.

Rate. The rate control adjusts the frequency of the low frequency oscillator modulating the pitch of the delayed signal.

Depth. The depth control adjusts the extent to which the low frequency oscillator modulates the pitch of the delayed signal.

Chorus

Chorus units also mix a delayed signal with the original signal, however, they use a longer delay than would be typical with a phaser or flanger. This delayed signal is then modulated by a low frequency oscillator.

The result can be a much more lush/smooth sound—at extremes however, it can sound like vibrato played too fast or just bad tuning.

Used in moderation, the effect can either add sparkle or thicken up a sound without noticeably changing the sound's characteristics.

The main controls you are likely to find on a chorus are similar to those on a phaser or a flanger and may include:

Delay. The delay control will adjust the time of the delayed signal.

Feedback. The feedback control sets the number of times the delayed signal is repeated. This has an effect on the perception of depth (or out-of-tuneness) of the effect.

Rate. The rate control adjusts the frequency of the low frequency oscillator modulating the pitch of the delayed signal.

Depth. The depth control adjusts the extent to which the low frequency oscillator modulates the pitch of the delayed signal.

Tone. The tone control will adjust the tone of the delayed, modulated signal to take away any harshness or metallic ringing.

Delay

Virtually every sampler with built-in effects will offer some sort of delay.

Usually there will be two ways to set the delay time:

The delay may be set as an absolute amount, usually defined in milliseconds.

Alternatively, the delay time may be linked to the tempo of the track, being defined as a fraction of the beat (for instance, with delay of 1/4, the signal is delayed by a quarter note).

Often several different types of delay unit are available—the most common are mono, stereo, and cross delay.

Mono Delay

A mono delay adds a simple delay to the input signal. A mono delay can be achieved in a stereo delay unit if the delay time is equal for both channels.

Stereo Delay

With a stereo delay, there are two separate delay lines which may (or may not) have different delay times. Conventionally these delay lines are hardwired left and right.

Cross Delay

Like a stereo delay, a cross delay has two channels. However, the output from the first delay is fed into the second and the output from the second is fed into the first. This can give the effect of spreading the delays across the stereo spectrum or ping-ponging the delays between the two channels.

Common Delay Controls

The most common controls that delay units have are:

Delay time. As noted, delay time is often (but not always) synchronized to the host's tempo and expressed in terms of beats.

Feedback. Feedback controls how many times each delay repeats (quite literally, it controls how much of the delayed signal is sent back to the input).

EQ, damping or a filter. EQ, damping, and a filter may be used to control the sound of the echo. Typically these controls will be used to cut the high frequency elements of the delayed signal to make the sound duller, resulting in a more natural sounding echo.

Reverb

Reverb can give a sound some spatial context.

With reverb units, there is significant interaction between the controls, so even more than normal it is necessary to balance several parameters when shaping the sound. The most common controls offered by reverb units are:

Pre Delay. This controls the time between the original signal and the reverb taking effect. Smaller values give the perception of a smaller room while longer delays give the perception of larger rooms.

Size (or Room Size). This controls the size of the room being simulated by the reverb unit and has a direct effect on the decay time of the reverb (in other words, the time for which the reverberation effect sounds). Occasionally some units will offer a separate decay control to adjust the decay time.

Damp. Damping works by cutting the higher frequencies in the reverb signal. One of the reasons why electronic reverb sounds artificial (that is clangy and metallic) is because it has too much high frequency information. The damping control can help to create a more realistic simulation.

Width. If a width control is present, it will adjust the stereo width of the reverb.

What Next?

If you're wondering where to find out more about making sounds with samplers and synthesizers, then check out the next section, Further Resources, which points you to a range of further materials (including free downloadable books and free sound design videos).

Further Resources

I hope you have found this book interesting and useful. If you want to know more about sound design and making music with synthesizers, then I suggest you head over to my website (noisesculpture.com) and check out my music-related books and videos.

If you're browsing the site, I suggest you look at the following books and videos first.

How to Make a Noise Series of Books

There are five books in the How to Make a Noise series (this one, plus the four others listed here). Each book looks at a different aspect of synthesis, focusing on the particular strengths of that technique.

While each book is separate, there is (as you would expect) some overlap (for instance, a filter will always act like a filter, whatever the synthesizer—but the way that the filter is used will vary).

How to Make a Noise: Analog Synthesis

(published by Coombe Hill Publishing, ISBN: 978-0955495502, available in electronic format only, noisesculpture.com/analog)

How to Make a Noise: Analog Synthesis looks at creating sounds using analog synthesis techniques.

Analog synthesis works by taking a sound source and shaping it with a filter. This is a simple but powerful technique which allows you to create the warm, rich, energetic, and commanding sounds, and with some care you will find tones with detail, complexity, and delicacy.

These are the techniques used to create the sounds that you associate with many classic synthesizers such as the Moogs (Mini Moog, Moog Modular, Taurus, Little Phatty), the Oberheim synths (OB-8, OB-X), the Rolands (SH-1000, Juno-6, Jupiter-4, and Jupiter-8), Prophet-5 from Sequential Circuits, and the ARPs (ARP 2500, Arp 2600).

You will have heard these techniques used on many recordings and will immediately recognize the sounds they produce. After reading the book you will be able to create and control these classic sounds when you apply the techniques to hardware or software synthesizers.

How to Make a Noise: Frequency Modulation Synthesis

(published by Coombe Hill Publishing, ISBN: 978-0955495502, available in electronic format only, noisesculpture.com/fm)

Frequency modulation synthesis came to prominence in the 1980s with the DX7 synthesizer which dominated the sounds in the charts for years.

FM synthesis adopts a completely different approach to creating sounds when compared with either analog or sample-based synthesis. Instead of starting with a bright sound and making it dull and/or shaping it with a filter, FM synthesis starts with a simple wave—usually a sine wave—and distorts it (usually with another sine wave). The interaction between the two (or more) waves creates and shapes the sound over time.

How to Make a Noise: Frequency Modulation Synthesis shows you how to create a broad, detailed, highly nuanced range of controllable tones. These principles can then be applied to any synthesizer with frequency modulation capabilities.

How to Make a Noise: iPad Synthesizers Edition

(published by Coombe Hill Publishing, ISBN: 978-0955495502, available in electronic format only, noisesculpture.com/ipad)

The fourth book in the series, How to Make a Noise: iPad Synthesizers Edition doesn't focus on a particular technique, but instead, looks at creating sounds on five synthesizers that are available for the iPad (BassLine, iMS20, iELECTRIBE, ReBirth, and NLogSynth Pro).

Many of the techniques discussed in the three previous books are covered in iPad Synthesizers Edition, but with particular focus to their application to the five featured iPad synthesizers.

How to Make a Noise: A Comprehensive Guide to Synthesizer Programming

(published by Coombe Hill Publishing, ISBN: 978-0955495502, noisesculpture.com/comprehensive)

There is another book in the How to Make a Noise series—the original hard copy (paper) book. The four electronic-only books are parallel editions to this existing book.

How to Make a Noise: A Comprehensive Guide to Synthesizer Programming was first published in 2005 and revised in 2007 and so is now past its first flush of youth. However, it is perhaps the most widely read book about synthesizer programming (and there's a free downloadable PDF version, too).

This book focuses on building sounds (over 300 sounds are built) and looks at building these sounds with six synthesizers (Cameleon 5000, Rhino 2, Surge, Vanguard, Wusikstation, and Z3TA+) which were on the cutting edge when the book was first published, but (in most cases) have now been superseded.

However, the techniques outlined are still applicable and the book also sets out other synthesis options, such as additive synthesis and wave-sequencing (which are not covered in any of the other publications).

Jumpstart Your Music Career

(published by Course Technology PTR, ISBN: 978-1435459526, noisesculpture.com/jump)

You will have noticed that this is not a book about synthesis. However, once you can make sounds, then might to want to make some money, and this is where Jumpstart Your Music Career comes in. This book is for every musician who wants to earn a living—and to sustain a long-term career—making music.

It has never been so easy to have a career in music: the barriers to entry have never been lower, the access to and ways to find fans have never been more straightforward, and the options to generate income have never been more plentiful.

The hard part is developing a strategy that's right for you and then executing it. Jumpstart Your Music Career looks at how you can build and sustain a successful career. It is the book that should be on every musician's bookshelf.

Cakewalk Synthesizers: From Presets to Power User, second edition

(published by Course Technology PTR, ISBN: 978-1435455641, noisesculpture.com/cakewalk-synthesizers)

Cakewalk Synthesizers: From Presets to Power User, second edition is the ultimate guide to understanding and using all of Cakewalk's synthesizers and learning about synthesizer programming.

The book features the complete range of Cakewalk (and rgc:audio) synthesizers, including Rapture, Rapture LE, Dimension Pro, Dimension LE, Beatscape, Session Drummer 2 and Session Drummer 3, Z3TA+, and Pentagon I.

There is also a package of downloadable material to accompany this book. This package includes additional chapters and is available for free to everyone (you don't need to have bought the book). This package also includes sound design master classes from several leading synthesizer programmers and so is well worth checking out.

Becoming a Synthesizer Wizard: From Presets to Power User

(published by Course Technology PTR, ISBN: 978-1598635508, noisesculpture.com/becoming-a-synthesizer-wizard)

Becoming a Synthesizer Wizard: From Presets to Power User is for:

People who are new to synthesis and who want to understand the basics and take their first steps in creating and controlling sounds with synthesizers.

Musicians who are familiar with synthesis, but who want to solidify or expand their knowledge so they can use their tools more fully.

Anyone who wants to build, share, and perhaps sell their own synthesizers.

The book looks at synthesis from first principles. It focuses on the elements of synthesis and the building blocks, and it highlights its examples using four leading modular synthesizers (Zebra 2, KarmaFX Synth Modular, VAZ Modular, and SynthEdit). From these building blocks, the book looks at a range of sound building techniques that can be used with the featured synthesizers, or can be readily applied to any other synthesizer.

The book is also a great introduction to SynthEdit for anyone who wants to create—and then share or sell—their own synthesizers.

Synthesizer Boot Camp

(noisesculpture.com/synthesizer-boot-camp)

Synthesizer Boot Camp is a series of videos which looks at specific synthesizer techniques that can be applied in creating sounds. All of these videos are freely available on my website and can also be accessed on the Noise Sculpture YouTube channel.