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Synthesis for dummies
Post by Kurups on Oct 14th, 2008, 9:27pm

Basically I'm going to just copy the tutorial from beatport into this thread as it's a lot easier for me personally to read it all on one page. I'm sure someone
somewhere will find this somewhat useful.

Images might be displayed as 'to the right' in the tutorial, most images are placed to the bottom of the segment.

*http://www.beatportal.com/ wrote this, not me i take no credit*

I. Subtractive and analog synthesis

Part 1: Introduction to Synthesizer Programming, Part One

Part 2: Introduction to Synthesizer Programming, Part Two

Part 3: Some call it analog: How subtractive synthesizers work

Part 4: Oscillators: Essential Waveforms

Part 5: Oscillators: Mixing and Blending

Part 6: Filters: The Wow Factor

Part 7: Filters: Going Deeper

Part 8: Understanding Envelopes, Part 1

Part 9: Understanding Envelopes, Part 2

Part 10: Essential LFO facts

Part 11: Grasping MIDI Controllers

II. Introduction to sampling

Part 1: Sampling Essentials – Part 1

Part 2: Sampling Essentials – Part 2

III. Introduction to effects

Part 1: Tutorial: How to use compression

Part 2: Tutorial: How to use a delay

Re: Synthesizers for dummies


Post by Kurups on Oct 15th, 2008, 03:07am

I.

Part One: Introduction to Synthesizer Programming

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To kick off our first series on music production techniques, we’ll begin by going deep into synthesizers. Sure, we all know that synths are musical instruments
that form the essence of all club music (a broad statement, but that’s a fact).

But did you know that there are upwards of five distinct types of synthesis technology that serve as the basis for most software and hardware synths?

When producers talk of a “synthesizer” they are generally referring to analog synthesizers like Moogs, Prophets and classic Roland gear, but that’s only one
type of synthesis.

In the pantheon of synthesis techniques, there are quite a few methods for creating new sounds that can be played via keyboards, drum pads, even MIDI-fied
wind instruments and guitars.

So, to get everyone warmed up, here’s a round-up of the most common sound generation tools available today.

Types of synthesis
Each type of synthesis has its own strengths and weaknesses.

Analog synthesizers are ill-suited for creating realistic pianos and orchestras, but are amazing for electronic basses and leads – and that’s just scratching the
surface.

FM synthesizers are also superb for creating exotic electronic textures, but have a more “digital” sound than the more common subtractive softsynths.

Then there’s physical modeling, which is theoretically capable of emulating the warmth and complexity of acoustic instruments, though the technology is still
relatively new (compared to analog, sampling and FM) and requires a fair amount of computational power to derive its sound.

Here’s a summary of these various sound generation systems, along with explanations of their strengths, weaknesses and applications.

Sampling
By now, pretty much everyone’s heard of sampling, as it’s probably the most common form of synthesis on the market – embedded into everything from
mobile phones to online gaming soundtracks.

Samplers can be found in a massive range of forms, from drum machines (like Akai’s legendary MPC series) to advanced software tools like Ableton Live,
which blurs the line between a performance instrument and a recording studio.

Akai’s vintage S1000 is a legendary sampler [on right] is favoured by countless producers for its easy-to-navigate LCD interface.

How it works: Samplers work by converting analog voltages (the audio signal) to digital information, which can then be manipulated to create loops, recreate
traditional instruments like piano, or generate single hits such as drums, percussion and acoustic sound effects.

Technically, the process is fairly straightforward, with two main criteria — sampling rate and sample resolution — that determine the quality of the digital
recording.

Sonically, the sampling rate governs the overall frequency range (low bass to trebbly highs) of the recording, whereas the sampling resolution, sometimes
called bit depth, determines the dynamic range (from quiet to loud) of the sample.

Pros: Sampling is the essential technology for looping “found” musical segments, as well as creating near perfect representations of acoustic instruments and
custom drum kits. It’s great for realistic sounds.

Cons: Not ideal for dynamic morphing textures, unless used in combination with other synthesis technologies.

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Subtractive
Subtractive synthesis is the oldest form of synthesizer technology and serves as the basis for the majority of both hardware and software based instruments.

For the record, all analog synthesizers are based on subtractive principles, but that’s not to say that analog is the only type of synthesis that relies on this
approach.

Generally, most of the other forms of synthesis — including FM and sampling — include a subtractive processing section to further modify the results of their
respective tone generation approaches.

How it works: Subtractive synthesis works by taking a harmonically rich oscillator source — often a simple waveform like sawtooth or square — then
subtracts specific ranges of frequency content via the use of filters.

Dynamic and morphing effects are created by changing the properties of the oscillators and filters by applying envelopes, LFOs (low frequency oscillators) and
MIDI controller data to their various parameters.

If this sounds really complicated, it’s not.

Here’s an analogy. Think of your CD decks as the original source (like the oscillator) and your mixer’s EQ as the filter section.

Taken a step further, some old school techniques like “transforming” (rhythmically moving the A-B slider of a mixer) have much in common with LFOs.
Approaching it from this perspective makes it easy to get a handle on the essentials with minimum head scratching.

Part of Reason’s popularity is the classic sound of its Subtractor synthesizer [right], which is capable of emulating many analog textures.

Pros: Classic analog and vintage sounds. Huge selection of capable hardware and software instruments. Easiest method of synthesis to master.

Cons: Not suited to realistic sounds like piano and orchestras (although analog strings are a notable exception).

FM (Frequency Modulation)
Slightly less common, but still with a devoted following is Frequency Modulation synthesis.

Originally introduced to the mainstream in the 1980s – thanks to Yamaha’s affordable line of DX synthesizers – FM has evolved into much more complex
tools, such as Native Instruments’ FM8 and Ableton’s Operator softsynths.

Modern FM synthesis fans rely on Native Instruments’ powerful FM8 softsynth [right], which also excels at recreating vintage Yamaha DX7 sounds.

How it works: The essential principle behind FM synths is extreme vibrato, like the quivering pitch of an opera vocalist, but much faster so that it operates in the

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audio range.

In an FM device, an array of interdependent oscillators — called an ‘algorithm’ — are tuned to different harmonics and/or frequencies.

From there, these oscillators (known as operators) interact with each other, generating harmonic spectra that can be used to emulate organic/acoustic
sounds or create bizarre digital textures.

Pros: Excellent for percussive, metallic sounds like bells, xylophones and plucked instruments, as well as punchy basses and exotic lead sounds.

Cons: Some producers dislike the sound, calling it digital and cold.

Re: Synthesizers for dummies


Post by Kurups on Oct 15th, 2008, 03:07am

Part 2: Introduction to Synthesizer Programming

Now in part two, we’ll continue by looking at the remaining types of synthesis, physical modeling and exotica, before jumping into the longest running debate in
the electronic music world - hardware synths versus software synths.

Physical Modeling
Sometimes referred to as PM, physical modeling is one of the most recent developments in synthesis, since it requires extreme computational resources to
generate its sounds.

PM forms the basis for software-based vintage keyboard emulations like AAS Lounge Lizard (electric piano), AAS String Studio (clavinets and guitars) and
Native Instruments’ B4 (tonewheel organs).

More complex virtual synths, like Apple Logic’s Sculpture, are also available.

Sculpture [right] is arguably the most comprehensive physical modeling synth for the mass market, so if you’re serious about physical modeling, then Logic’s
Scuplture is the only way to go.

It’s the deepest synth of its kind available today.

How it works: Physical modeling is unique in that it mathematically recreates the actual physics behind acoustic and mechanical instruments.

The behavior of vibrating strings, wind instruments, and struck metal objects can be converted into complex equations, which are then manipulated using
knobs and sliders.

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If you want to play a 20-meter glass flute or an iron tower struck with a hammer, this is the way to go.

Pros: Extremely complex acoustic-sounding instruments can be designed entirely in software.

Cons: Due to its complexity, PM is very hard to master. In fact, it’s possible to set up the parameters so that no sound is created, due to oddities in the
behavior of acoustical physics.

Exotica
In addition to the above forms of synthesis, there are several other less common methods that are used for various sounds and effects.

These include additive synthesis (Camel Audio’s Cameleon 5000, and the vintage Kawai K5), vector synthesis (Arturia’s Prophet V and Korg’s Wavestation),
and phase distortion (Casio’s vintage CZ and VZ synths).

Camel Audio’s Cameleon [right] is an extremely powerful additive softsynth.

As if this weren’t enough, monster softsynths like Native Instruments’ Reaktor can be used to create hybrid instruments that use any or all of the above
approaches.

Softsynths vs. Hardware

One of the longest running arguments in the electronic music world revolves around the perceived strengths and weaknesses of hardware synths versus
software synths.

Thankfully, there is no clear winner here, since thousands of amazing tracks have been produced using either approach — or more often than not, a
combination of the two.

Rather than take sides, here’s a quick rundown of the pros and cons of each.

Softsynths
Softsynths are plug-in applications that run inside a host application, most often a digital audio workstation (DAW) like ProTools, Ableton Live or Apple Logic.

Each DAW and associated computer platform (Mac, Windows) has its own set of criteria for plug-in compatibility.

The most common formats are VST, DirectX (Windows), AudioUnit (Mac OS X), TDM/RTAS (ProTools), and MAS (Digital Performer).

Pros: Extremely affordable. Total recall of a virtual studio configuration. Great sound. A single computer can host hundreds of different synths. Easier to use on
a plane than a Moog modular system.

Cons: When a DAW or operating system is upgraded, compatibility can sometimes break, forcing an upgrade and/or extreme inconvenience.

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Unless special care is taken when mixing, tracks made entirely from softsynths can sometimes suffer from a one-dimensional sound. Working with a mouse-
and-monitor interface can be cumbersome for certain users.

Hardware
Hardware synthesizers are for many, “the real deal,” and come in every synthesis flavor described above.

Some users prefer hardware for a variety of reasons, including stability (hardware synths rarely crash) and having a tactile, hands-on user interface.

Others find it inconvenient to dedicate an area of their living space to stacks and racks of geeky-looking electronic artifacts.

Pros: In the case of pure analog synths like Dave Smith Instruments’ new Prophet 08, there’s an extreme presence and punch.

Great for gigging live. Knobs make adjustments a fun and intuitive process. Generally very robust and stable. Compatible with any studio configuration.

Cons: Much more expensive than software. External MIDI control can create timing inconsistencies (due to the archaic transmission rate of MIDI data).

Vintage equipment requires periodic maintenance for tuning and such. Compared to laptops, not very portable.

A real Moog Modular [above] won’t fit on an airplane tray-table, but Arturia’s softsynth version will — and does an admirable job of recreating its sound.

Re: Synthesizers for dummies


Post by Kurups on Oct 15th, 2008, 03:08am

Part 3: Some call it Analog - How subtractive synthesizers work

The term ‘subtractive synthesis’ refers to the sound generation method used by many modern software synths, as well as all analog synthesizers – new or
old.

The fundamental principle consists of taking a sound source, called an ‘oscillator’, then feeding that signal into a filter.

The filter then subtracts frequency content from the oscillator signal, allowing it to be made brighter, duller, squeakier, and so on.

After the filter, the overall volume of the resulting sound can be adjusted to create percussive sounds, long swells, and so forth.

If this sounds too complex to grasp, fear not.

The secret to understanding subtractive synthesis lies in comparing it to other, more familiar, concepts from the DJ world.

A good analogy would be a classic DJ rig.

Two turntables (the sound sources), feed a mixer that contains an EQ (which correlates to the filter section), the result then feeds an amplifier.

If you have any experience with funky guitars, you can think of it this way: A guitar generates the music, which is then processed by a wah-wah pedal (which is
in fact, a filter), and the wah-wah output is amplified.

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This, gentle readers, is the underlying concept behind every analog synth.

The components for subtractive synths share familiar terminology, so here’s a summary of the most common components, along with a basic description of
their associated functions.

As this series progresses, we’ll dig much deeper into each module.

Oscillators
The oscillator is where the sound generation journey begins.

Most analog hardware and software models rely on two (sometimes even three or four) oscillators to generate the initial sound.

These oscillators are capable of generating an array of different waveforms, each with its own unique sonic fingerprint.

For example, sawtooth waves have a bright, buzzy character that’s great for in-your-face basses, pads and leads.

Square waves have a more hollow character that evokes both woodwind instruments and old school video games.

Other waveforms, like triangle and sine, have their own unique sound as well.

Since there are at least two oscillators involved, each one can be tuned independently and assigned its own waveform, then mixed together to create an even
more complex texture.

Common combinations include using the same waveform on both oscillators, then detuning each slightly for thick chords and trance-like leads.

Another increasingly fashionable option relies on tuning each oscillator to a different musical interval, like a fifth, then playing both notes from a single key.

The main riff from the D.Ramirez remix of ‘Yeah Yeah’ is a perfect example of this approach [listen to the track in the player below].

Arturia’s impressive Jupiter 8v softsynth [right] faithfully recreates Roland’s original dual-oscillator design.

Filters
The filter section modifies the mixed oscillator sound.

The most common type of filtering is called ‘low pass’ because it allows frequencies below a set frequency to pass unmodified, while frequencies higher than
the set frequency — called the ‘cutoff’ — are attenuated, or lowered, in volume.

Another parameter, called resonance, interacts with the cutoff frequency, creating squeaky/squelchy effects that are the hallmark of both TB-303 acid riffs and
old school funky synth basslines.

Other filter modes include high pass (which reduces low frequencies), band pass (kind of like a sweepable mid EQ) and notch.

Each mode has specialized characteristics that are useful for different types of sounds. We’ll explore these in more depth in future tutorials.

Ableton’s Sampler instrument [right] includes an extensive array of subtractive synthesis amenities like graphically adjustable filters and envelopes.

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Envelopes
In addition to filters and oscillators, classic analog synths also include tools like envelopes and LFOs — often called ‘modulators’ — that allow the oscillator
and filter characteristics to be modified dynamically as you play a sequence of notes.

Envelopes are modulators that are triggered when a key is played, thus shaping the character of each individual note.

For example, an amplifier envelope governs how the overall loudness of a sound changes when a key is pressed and held.

Some envelope settings create percussive sounds, other settings create long sustaining sounds, and so on.

The majority of modern synths include dedicated envelopes for the filter, amplifier, and oscillator pitch, though it’s sometimes possible to route the envelopes
to alternate destinations for even more exotic results.

LFOs
The acronym LFO stands for Low Frequency Oscillator, meaning an oscillator that operates below the range of human hearing (which incidentally, is 20 Hz to
20 kHz, for the supergeeks out there).

LFOs can be routed to control the most common synth parameters like pitch, amplifier, and filter cutoff, as well as other useful destinations like panning and
oscillator mix.

Since the rate of oscillation is so slow, it can be used to create repeating effects like vibrato (a quiver in pitch), tremolo (a quiver in loudness), or wah-wah (a
repeating change in filter character).

Like audio oscillators, LFOs also offer a selection of different waveforms, each with its own rhythmic properties.

Sine and triangle waves are commonly used for vibrato and tremolo, whereas sawtooth and square waves are terrific for pulsating or echo-like effects.

Also included in Ableton’s Sampler are three LFOs [right] that can be synchronized to tempo for creating powerful rhythmic effects.

MIDI Control
MIDI control is another critical component for creating interesting, evolving textures that build to a peak and/or transform in other ways over the course of a mix.

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One way to control that evolution is to have the various controllers on a MIDI keyboard affect different aspects of the synthesis engine.

Options such as note velocity (how hard you hit the keys), modulation and pitch wheels (knobs or levers at the far left end of a keyboard), or even foot pedals,
can allow specific parameters in the filter or envelope sections to morph as you adjust these controllers.

This really helps to make a repeating sequence come to life and give your tracks a lot more complexity.

With a little experimentation, you’ll find a better understanding of how the various components of a synth interact.

I strongly urge readers who do not already own a sequencer or digital audio workstation software to head over to www.propellerheads.se and download the
demo version of Reason.

Doing so will allow you to follow along with these lessons, experimenting as you go.

Re: Synthesizers for dummies


Post by Kurups on Oct 15th, 2008, 03:08am

Part 4: Oscillators - Essential Waveforms

Now that you have a handle on the signal flow of subtractive synthesizers, it’s time to dig into the specifics of the oscillator module.

To begin, let’s take a look at the most common waveforms you’ll find in an analog-style synth.

Each has its own distinct sound, so it’s important to understand their individual strengths.

Sine
The sine wave is the “atom” of the sonic universe.

Consisting of a single harmonic – often the fundamental, or root pitch – the sine wave generates the simplest sound of all the waveforms, and therein lies its
strength.

If you’re looking for that deep sub-bass that anchors many drum and bass tracks, start with a sine wave.

Similarly, you can use a sine wave is to add beef to any of the other waveforms by tuning it to the same pitch (or an octave lower for even more low-end
bombast), then balancing the oscillator mix until it suits your purposes.

Alternately, you can use a high-pitched sine wave to add a specific type of shimmer to bell-like textures.

Sawtooth
At the opposite end of the waveform spectrum is the sawtooth wave.

Consisting of all integer harmonics, the sawtooth has an extremely bright and buzzy sound that makes it the Swiss Army knife of synth waveforms.

Used in conjunction with a moderate lowpass filter setting, sawtooth waves are great for subtle pad sounds.

Open the filter up to maximum and add a slightly detuned sawtooth to the second oscillator and you’re on your way to creating classic trance leads.

Lower the filter cutoff to about 30-40% and add some filter envelope modulation and you’ve got the essentials for a timeless bass sound.

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Square
The square wave has become a bit trendy in dance music for the past several years.

Because of its Nintendo-esque hollow character, it lends itself to bright electro leads, glitchy embellishments and tech house bass lines.

The harmonic content of a square wave consists of only odd numbered harmonics, descending in volume in a linear manner.

That is, every successive odd harmonic is slightly quieter than the previous one.

Mixing square and sawtooth waves, especially in different octave ranges, can result in more complex harmonic combinations.

Alternately, you can use a square wave as the basis for emulated woodwind/reed sounds like flutes and clarinets.

Triangle
While its shape is quite different from that of a square wave, the triangle wave also contains only odd numbered harmonics.

The difference here is that these harmonics descend in volume in an exponential manner.

In plain English, this means that the triangle sounds like a more muted – or duller – square wave.

This makes it quite useful for supporting other waveforms without getting in the way, sort of like the sine wave but with additional harmonic character.

Pulse/Rectangle
The pulse wave, sometimes referred to as a rectangle, is a very flexible beastie.

Many synths have a pulse-width control for their square wave option, which is how this waveform is created.

In the case of a square wave, both sides of the cycle have 50% durations, hence the square shape.

If this duration is changed to 60/40, 30/70 and so forth, the harmonic structure shifts, giving the pulse wave a reedier character that’s quite useful in situations
when a sawtooth is a bit too overbearing.

What’s more, when you use an LFO to modulate the pulse-width continuously, it serves to thicken the sound and is great for pad and lead sounds.

Noise
Noise generators are often found in addition to the oscillators on a synth, as well as a waveform option.

White noise is created by generating every possible frequency at the same volume.

Some synths include a “color” control to vary the frequency content somewhat, creating three basic types of noise: white, pink and blue.

Pink noise has slightly emphasized lower frequencies, whereas blue noise is slightly tilted toward the highs.

In its raw form, white noise sounds like old school television static, but if you apply a filter to it, you can shape the sound to create everything from nature
sounds to vintage analog percussion effects.

Ocean waves and wind sounds are emulated by applying a filter to a noise source.

In modern club tracks, one of the most common uses for noise is to generate those giant whooshes that come at the end of a long breakdown, bringing the
track to its peak.

Hybrid and Digital Waveforms


Finally, many softsynths (notably, Reason’s Subtractor synth) include an array of additional digital waveforms that have more exotic harmonic content.

Some of these waves have vocal qualities, others have bell qualities, and so on.

These waves are excellent for creating more metallic, digital sounds, but often don’t have the same perceived “warmth” that the classic waves have.

That’s not to say they’re bad; it’s all a matter of your production objectives.

[PPG Wave, right image: The PPG Wave was the first mass-produced synth to rely on digitally generated wavetables for its oscillators. The originals are
collector’s items, but fortunately for us, Waldorf has released an affordable softsynth replica.]

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Speaking of Reason, I strongly urge readers who do not already own a sequencer or digital audio workstation software to head over to www.propellerheads.se
and download the demo version of Reason.

Doing so will allow you to follow along with these lessons, testing and experimenting as you go.

Re: Synthesizers for dummies


Post by Kurups on Oct 15th, 2008, 03:09am

Part 5: Oscillators - Mixing and Blending

Once you understand the harmonic characteristics of the basic oscillator waveforms, the real fun begins. By tuning and mixing various waves, you can create
entirely new sounds from scratch.

This week, we’ll cover the essentials of how to tune and mix oscillators.

Note: If you want to follow along at home, fire up your demo version of Reason, create a Subtractor synth and make sure oscillator two is turned on (the
orange box next to “Osc 2” should glow).

You can get Reason at www.propellerheads.se.

Detuning
Most oscillators include tuning controls for the octave range, semitone (note) and cents (detuning) of each waveform.

Cents are the smallest musical tuning increment and are used for thickening a sound or making it sound “out of tune”.

While the term “cents” may seem a bit odd, it is used because there are one hundred cents in every semitone (also known as a half-step or note).

One of the most common sounds in epic trance is created by blending two sawtooth oscillators and detuning them by an equal amount in opposite directions.

Herein lies a secret to good sound design: By detuning each oscillator by an equal amount — positive and negative — you retain overall tuning coherence,
instead of leaning sharp or flat.

[Trance Lead, above image: This oscillator combination forms the basis for many classic trance leads. Note that in all the examples in this lesson, the filters
are wide open, so you can really hear what’s going on.]

Octaves
At the opposite end of the tuning spectrum are octaves.

If you’re not musically inclined yet, here’s an easy way to understand octave tuning.

When two oscillators are tuned to the same octave, playing Middle C generates two tones at the same pitch.

When one octave is lower, it’s like playing Middle C and the C below it simultaneously — and so on.

The track ‘Do It (Original Mix)’ by Cosmic Belt relies on a sample from the Duran Duran classic, ‘Save A Prayer’.

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The sound in Duran’s original hit was created by taking two square waves and tuning them apart by an octave, then applying filtering and envelopes.

In this example (see screenshot, above), we create the raw sound of ‘Save A Prayer’ before it is modified by the other modules, like filter and envelope.

By detuning oscillators by even wider amounts, like two octaves or more, we can create still more unique textures.

Nowadays, it would be prohibitively expensive to sample anything by the artist currently known as Prince.

However, we can easily recreate the synthy organ sounds used in many of his early tracks, like ‘Let’s Go Crazy’.

http://www.beatportal.com/uploads/news/TechBlogMinneapolisOrgan.jpg

The trick here lies in understanding that the essence of that sound is two sawtooth oscillators tuned two octaves apart (see screenshot, above) with the filters
wide open.

For more of that funky Prince goodness, try adding some triangle or sine wave vibrato (via an LFO) to the oscillator pitch.

Intervals
This is where things may get a little confusing.

Some synthesizers integrate interval tuning (AKA tuning by semitones) into the octave control by combining the functions and using a single control called
“coarse tuning” or something similar.

Other synths, like Reason’s Subtractor, use discrete controls for octaves and semitones.

The end result is identical; in many cases, interval tuning allows you to create sounds that play two notes with a single key.

In more refined examples, you can use interval tuning to highlight specific harmonics or frequencies.

Here are some examples of each.

Analog B3

By using interval tuning to emphasize certain frequencies, you can create entirely new harmonic textures that are useful for recreating specific instruments or
adding shimmery highs to a sound.

For instance, if you want to emulate a jazzy house organ, but don’t have access to a Hammond B3, just take two sine waves and tune them apart by one
octave plus seven semitones — or an octave and a fifth, if you’re musically trained.

The settings in the screenshot (above) display the specifics.

This recreates the sound of an organ with the first and third drawbars pulled out.

While this type of sound is ideal for chordal riffs, it also forms the basis for hundreds of classic house bass lines, notably the groove from ‘Show Me Love’ by
Robin S, which was recently covered by Mobin Master.

Analog Bells

Another cool interval trick involving two sine waves is the creation of ethereal, mellow chime sounds.

The approach here is to leave one sine wave untouched so it plays the fundamental (AKA the tonic or root).

Next, increase the octave and interval values on the second oscillator by a wide margin, then adjust their relative mix to taste (see screenshot, above).

If you’re familiar with the basics of envelopes, set a long release time on the amplifier envelope so the sound trails off like a real bell.

Experimentation is the key here, so have fun with it.

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That Yeah Yeah Sound


Earlier in the series, I mentioned that interval tuning was the secret to creating the lead sound in D. Ramirez’s massive remix of ‘Yeah Yeah’.

Now that you have a better understanding of what’s going on with oscillators, I’ll leave you with a step-by-step tutorial on using these techniques to duplicate
that sound using Reason.

Step 1: Fire up Reason and create an instance of Subtractor. Its default initial patch is actually set up nicely in the filters and envelopes, so you won’t have to
do much tweaking to get started, other than to make sure oscillator two is turned on (see the intro to this piece for directions).

Step 2: The core of this sound lies in the fact that the oscillators function as both lead and bass because of their extremely wide tunings. Keep the waveforms
set to sawtooth, then tune one oscillator down by two octaves, using a value of 2. Next, tune oscillator one up by a major third by raising its semitone value to 4
(see screenshot, right, for exact settings).

Step 3: Play a few notes and you should hear that we’re on the right path.

Step 4: Now, to get that seasick pitch sweep, you’ll need to add some portamento (also known as glide). Portamento causes the oscillators to slide from one
note to another like a trombone or slide whistle, rather than jumping in pitch like a piano or organ. To make the effect more consistent, reduce the polyphony
amount to 1, so that only one note can be played at a time. Using the screenshot (right) as a reference, set the portamento value to about 60-70% to maximize
similarity to the track’s lead.

Step 5: Now, play the following notes: F, F#, G, G#, then C, F#, F and D#. From there, you should be able to figure out the phrasing and rhythm of the lead.
The diagram (above) shows exactly which notes are used in the part, for those who are new to keyboards.

Re: Synthesizers for dummies


Post by Kurups on Oct 15th, 2008, 03:09am

Part 6: Filters - The Wow Factor

Once your oscillators are configured to generate the basis for your sound, it’s time to shape the frequency content via filters.

At first glance, filters appear to behave like an equalizer, but there are subtle differences that give them a very different sonic character.

The most significant distinction is the fact that filters subtract frequency content, whereas EQs can raise or lower the volume of entire frequency ranges.

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That doesn’t mean filters are less powerful than EQs. Far from it! Different tools for different jobs, that’s all.

In this tutorial, we’ll cover the four main types of filter and how each mode affects the sound generated by the oscillators.

Filter Types
The four basic types of filter are: low-pass, high-pass, band-pass and band-reject (also referred to as “notch”).

Each has a specific character and behavior that is useful for different types of sounds and effects.

Many modern synthesizers include a multi-mode filter, which can operate in any of these modes.

In addition, some filters also include options for toggling the rolloff slope of the filter between 12 dB per octave and 24 dB per octave.

In simple terms, a steeper slope (24 db/oct) creates a more pronounced effect.

Other implementations chain two filters in series.

Propellerhead Reason’s Subtractor, Malstrom and Thor all provide multiple filter paths, as does Ableton Live’s new Analog synth.

In these instances, the output of the first filter feeds the input of a second filter for more complex signal manipulation.

Low-Pass
The image below, taken from Ableton Live’s Auto Filter device, shows a low-pass filter with a slight increase in resonance (sometimes called Q or emphasis).

The horizontal axis represents the frequency range from low to high.

The vertical axis represents the amplitude — or volume — of the curve.

A low-pass filter passes the frequencies below the cutoff frequency and attenuates (or reduces the volume of) the frequencies above it.

Lowering the filter frequency makes the resulting output duller or more muted.

Raising it opens up the sound, allowing the full range of upper frequencies to pass.

If you’ve ever heard a breakdown that drops down to a low rumble (like the sound of music through the wall of an apartment or underwater) then gradually gets
brighter and clearer, then you’ve heard a low-pass filter.

Most older analog synthesizers relied exclusively on low-pass filters as their sole filter type, so if you’re wondering how classic 1980s era synthpop sounds
were created, you should start your sonic explorations using low-pass mode.

High-pass
In this image, the curve shows the frequency response of a high-pass filter.

High-pass filters are essentially the opposite of the low-pass type.

They work by passing the frequencies above the cutoff frequency and attenuate the frequencies below.

This can make the sound thin and buzzy.

Some older analog synths — notably those from Roland, like the Juno 6/60/106 and Jupiter 8 series — included a simple high-pass filter in conjunction with
the low-pass module.

High-pass filters are often used in remixes as a technique for introducing new patterns by starting them out trebly and tiny, then slowly making the sound larger
and fuller by allowing more low frequency content through.

Band-pass
While the image below may look like an EQ, the band-pass filter mode works in a somewhat different fashion.

Band-pass filters allow a specified range of frequencies to pass unaffected, while attenuating all frequencies on either side of that range.

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Sharing characteristics of both high- and low-pass filters, band-pass filters are useful for sculpting signal content in a manner not unlike that of the mid-band
EQ found on a DJ mixer.

On high-pass and low-pass filters, the frequency knob determines the cutoff frequency at which the filter begins attenuation.

On band-pass and band-reject filters, this parameter determines the center frequency of the filter’s operation.

Band-pass filters often sound quite good when the frequency is swept (continuously raised and/or lowered) either by automation, envelope or LFO.

The result is quite similar to a guitarist’s wah-wah pedal — and with good reason, since band-pass filters are commonly used in this type of effect pedal.

Band-reject (Notch)
he image below shows how a band-reject filter’s response is a mirror image to that of the band-pass.

A notch filter functions in an inverse manner to band-pass filtering.

It rejects the frequencies within a specified range, while allowing all others to pass.

The sound of a notch filter is quite subtle in practice, bearing a slight resemblance to a phaser when the center frequency is swept via an envelope or LFO.

Another good use for notch filters lies in the removal of problematic frequencies within a sample, much like a surgical grade EQ.

Next up…

In the next installment, we’ll go deep on the specific parameters that govern the behavior of filters.

Re: Synthesizers for dummies


Post by Kurups on Oct 15th, 2008, 03:10am

Part 7: Filters - Going Deeper

Last week, we looked at the four most common filter modes and how each affects the oscillators’ sound.

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This time out, we’re going to go a bit deeper into the specifics of the filter parameters.

As many readers have discovered, some synthesizers are extremely complex with elaborate functionality in each sound-processing module.

Other synths are more much straightforward.

To keep things consistent, we’ll be focusing on parameters that are common to all filter implementations.

Filter Parameters
Each filter includes an array of parameters that govern its overall sonic behavior.

The two main parameters are frequency (sometimes called the “cutoff” or “center” frequency) and resonance (also referred to as “emphasis” or “Q”).

Other common parameters include keyboard tracking, filter mode and modulation amounts for the LFO and envelope.

*Frequency

The aptly named frequency knob (or slider) determines where the filter begins operating on the oscillators’ frequency content — subtracting highs, lows or
mids, depending on the filter mode selected.

In the case of a low-pass filter, this knob serves to select the cutoff frequency, above which all higher frequencies are attenuated.

In a high-pass filter, the cutoff frequency has the opposite effect and all frequencies below the cutoff are lowered in volume.

In band-pass and notch modes, the frequency parameter sets the center frequency of the filter.

This is similar to the behavior of an EQ’s frequency knob.

In any of the modes, sweeping the cutoff frequency manually, or with a modulation source, will yield dramatic results.

*Resonance

The resonance parameter is sometimes referred to by different names.

On vintage Moog synthesizers, like the Minimoog pictured on the right, it’s called “emphasis”.

On Ableton’s synthesizer devices it’s abbreviated it as “Res”, whereas Ableton’s Auto Filter device labels it “Q.”

In high- and low-pass filter modes, the resonance knob emphasizes the cutoff frequency by increasing the amplitude of the signal at that exact frequency.

At medium values, this creates a squelchy effect that gives a synth that “boingy” sound.

At higher values, a whistling sound is superimposed on the signal content.

This whistling is called “self-oscillation” and is often associated with the TB-303 sound of acid house, as well as funky 1970s and 1980s sounds.

In the case of band-pass and notch filters, things can get a little confusing.

On many synths, the band-pass resonance knob works in a consistent manner, emphasizing the frequencies at and/or near the center frequency.

On others, it serves a somewhat different function, setting the width of the range of frequencies that are passed or attenuated by the filter, much like the Q
parameter of an equalizer.

The difference can be subtle, so when in doubt, use your ears!

*Keyboard Tracking

Sometimes abbreviated simply “keyboard” or even “kbd”, this parameter allows the filter frequency to track the pitch of the keyboard with higher notes raising
the frequency and lower notes reducing it.

Why is this useful?

Think of it this way.

Say you have a low, muted bass tone with a low-pass cutoff frequency tuned to around 260Hz (approximately middle C on a keyboard).

When you play your bassline, everything works as predicted, but if you jump above middle C, the sound becomes quieter.

If you play higher still, the sound fades away entirely.

This is because the filter cutoff frequency is set lower than the notes you are playing, so it cuts off those very notes!

By adjusting the keyboard tracking parameter, you “tune” the filter to the keyboard so that the higher you play, the greater the cutoff frequency.

This creates a more consistent behavior for certain types of sounds.

With high, self-oscillating resonance values, keyboard tracking can even be used to turn the filter into an additional oscillator.

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*Envelope Amount

The vast majority of subtractive synths allow the filter frequency to be modulated, creating dynamic motion in a sound.

The most common type of filter modulation is envelope modulation.

Ableton’s Sampler, shown above, has a graphically adjustable envelope for its filter.

Envelopes control how a sound changes over time.

For instance, to create that “wow” sound, you could use a filter envelope with a longer attack.

To create a percussive sound, you could create an envelope with a short attack and a rapid decay.

The overall intensity of the envelope’s effect on the filter is governed by the filter’s envelope amount control.

Note: Envelopes will be discussed in greater detail in a future lesson.

*LFO Amount

LFOs are a terrific source of repeating often rhythmic effects like vibrato and tremolo.

When applied to the filter frequency, they can create undulations that give a sound a lot more character and motion.

One of the most dramatic examples of this effect is Timo Maas’s legendary remix of ‘Dooms Night’, which has several LFOs routed to various destinations,
including low-pass filter frequency.

As with envelopes, the specifics of LFOs will be discussed in an upcoming tutorial, but in the meantime, I highly recommend exploring the effects of this
parameter.

Start with a low-pass with a low cutoff frequency, say 25-35%.

Then increase the value of the LFO amount until you hear its effect on the filter.

From there, you can experiment with various LFO settings.

Re: Synthesizers for dummies


Post by Kurups on Oct 15th, 2008, 03:10am

Part 8: Understanding Envelopes, Part 1

Now that we’ve covered the essentials of oscillators and filters — a.k.a. the audio signal path of subtractive synthesis — it’s time to delve into modulation.

The term “modulation” covers components like envelopes, LFOs and MIDI continuous controllers, to name a few.

These modulators allow sound designers to effectively automate various parameters and elements — such as filter cutoff and volume — within the audio
chain.

But there’s a lot more to it than just those two parameters.

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In this tutorial, we’ll begin by addressing the essential components of the envelope itself.

What’s an envelope?
My favorite analogy for what an envelope does is as follows: An envelope consists of a sequence of events that occurs every time you press a key.

For some, this may be an oversimplification, but it’s actually fairly accurate… Especially when you consider that many modern software synths include
sophisticated multi-stage envelopes that can even create rhythmic patterns each time a note is played.

For the purposes of this tutorial, we’ll focus on the most common type of envelope (pictured above): ADSR.

ADSR is an acronym for the four stages of a basic envelope: attack time, decay time, sustain level and release time.

Some envelopes – such as those on hardware synths or their modern software emulations — are controlled by a knob or slider type interface.

Newer softsynths – like those from Native Instruments, Rob Papen and Ableton – feature graphically editable envelopes that display the shape of the envelope
itself, instead of purely numeric values.

The ADSR configuration


Attack time

The first stage of the envelope is the attack (pictured below).

This parameter determines how the sound begins.

When a key is first pressed, the attack segment is activated and proceeds to the envelope’s peak (maximum value), which is set via the envelope amount
parameter and/or the “peak level” on some synths.

Fast attacks are used for percussive sounds like pianos, drums and plucked sounds.

Slow attacks are useful for softer sounds like woodwinds or strings, as well as long sweeps and swells that are often used during breakdowns.

Decay time

After the attack segment is completed, the envelope proceeds to the decay stage (pictured below).

This component determines how long it takes for the sound to transition to the sustain level, or if the sustain is set to zero, fade to silence.

Fast decays are appropriate for percussive sounds, whereas longer decays are useful for sounds that fade out over a longer time, like pianos and cymbals.

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Sustain level

After the decay completes, the sustain parameter (below) determines the overall level or volume that the envelope will maintain if a key is held for a longer
duration.

For example, long string parts or chordal pad progressions require sounds that remain at a specified level for several bars.

By setting the sustain level to any value higher than zero, the sound will continue indefinitely until you lift the key.

Release time

The release parameter (below) determines the amount of time it will take for the envelope to return to zero — or silence — after the key is lifted.

Long releases are great for cymbal crashes, transitional sweeps, and ambient textures.

Short releases are ideal for any sound you want to stop immediately when the key is lifted, such as stabs, percussive sounds and gated, rhythmic effects.

Additional envelope segments

Some envelopes are slightly more complicated than the ADSR type, while retaining the same basic properties.

Common additions to the ADSR envelope include the following segments.

Delay time

Delay is an additional segment that precedes the attack portion, causing the envelope to wait for a given period after the key is pressed before starting the
attack.

This can sound a bit odd when used on a volume envelope, as it just feels like severe latency.

That said, it’s quite useful for modulating timbre and pitch parameters in unusual ways, like adding a filter swell long after the key has been pressed.

Peak level

As previously mentioned, the peak level parameter specifies the maximum level an envelope will reach after the attack portion before moving on to the decay
segment.

Hold time

This segment is rather unusual but quite useful for adding a bit of punch to an envelope.

Essentially, the hold time parameter will cause the envelope to sustain for a bit at after the attack peak, before moving on to the decay segment.

Note: According to legend, early Moog synths had a non-adjustable hold time of approximately 30 milliseconds immediately following the attack.

This is often associated with the overall impact that characterized vintage Minimoog sounds.

Regardless of its veracity, if your envelope supports hold times, try adding a bit of this to percussive and lead sounds as it really does work.

Sustain time

Some envelopes also include a second embedded decay that reduces the sustain level over time.

Used creatively, this can create two consecutive decay segments for more complex envelope shapes.

Curve

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Non-linear curves are another great way to add impact and enhanced percussiveness to a sound.

Some envelopes (pictured above), such as those found in Ableton’s Sampler device, allow the curve of each segment to be varied from linear to exponential to
inverse exponential, simply by dragging the blue Bezier dot on each segment.

While this sounds extremely math intensive and obtuse, it’s actually quite simple to manipulate and immediately hear the results.

At first listen, the effect can be quite subtle, so it’s useful to create a short repeating sequence and tinker with these parameters as the sound plays.

Re: Synthesizers for dummies


Post by Kurups on Oct 15th, 2008, 03:10am

Part 9: Understanding Envelopes, Part 2

This time around, we’ll touch upon envelope looping as well as more complex multistage envelopes.

After that, we go deeper into the sound design possibilities of modulating various synthesizer functions like filter and pitch.

Envelope looping
Some products include a feature that causes the loop to repeat, allowing you to create complex rhythmic effects similar to those of LFOs.

Certain softsynths also include the ability to synchronize this looping to various note values, like quarter-note, eighth-note and sixteenth-note.

Looped, tempo-synced envelopes were tailor-made for club music, so if your synth supports this function, it’s a deep source of creativity and inspiration worth
exploring.

Multistage Envelopes
Multistage envelopes go beyond the ADSR format, allowing for complex shapes that swoop and undulate every time a key is pressed.

The first commercially available synth to include multistage envelopes was Yamaha’s legendary DX7, which included rate and level parameters for each
stage, allowing for extremely intricate envelopes.

Modern softsynths, like Native Instruments’ FM8 and Massive, include multistage envelopes (pictured above) that are capable of complex morphing and
rhythmic effects.

Once you have a mastery of basic ADSR envelopes, the complexity of a multistage envelope may become irresistible.

Fortunately for Ableton Live users, the sequencer’s clip envelope feature provides much of this same functionality in a very intuitive and flexible manner.

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Envelope Destinations
The three most common destinations for envelope modulation are amplitude/volume, filter cutoff and pitch.

Amplitude/Volume

Every synth includes some sort of volume envelope, otherwise each sound would sustain indefinitely!

As the name suggests, this type of envelope controls how the loudness of a sound will change over time.

Filter envelope

Many synths also include a dedicated filter envelope, which controls the behavior of the cutoff frequency.

By manipulating the segment times and overall envelope amount, it’s possible to create swells, wah-wah effects or enhance the percussiveness of a sound.

Raising the resonance of the filter and applying envelope modulation to the cutoff, results in more dramatic, funky and/or squeaky effects.

Some filter envelopes also include a bi-polar or invertible amount control.

If an envelope is inverted, it will begin with a decay, followed by a rise (second attack) to the sustain level, and finally, when the key is lifted, a final rise to the
original cutoff value.

Note: When working with the filter envelope, it’s important to keep in mind that the envelope will add or subtract from the current cutoff frequency setting, so if
the cutoff is set at maximum and the envelope amount has a positive value, you will not hear its effect since the cutoff is already at full value.

Conversely, if your cutoff is set to minimum and the envelope amount is a negative value, then the envelope will have no effect for the opposite reason.

Pitch envelope

Sometimes a third envelope is included for pitch or other modulation possibilities.

Alternately, some synths allow the filter envelope to also modulate pitch.

In either case, a pitch envelope will affect the tuning of one or both oscillators.

With a moderate amount setting and fast segment times, a pitch envelope can enhance the attack and/or add an organic quality to a sound.

More extreme settings can be used to create those electro sound effects that run the gamut from squeaky to wild pitch sweeps.

If electro is your cup of tea, I highly urge you to start with a single sawtooth oscillator, set the lowpass filter cutoff to maximum, then go crazy experimenting
with the pitch envelope options.

You’ll be pleasantly surprised.

Other parameters

In some synths, like the Prophet 08, Reason’s Thor, and Ableton’s Sampler, the third envelope can be routed to many other destinations besides pitch.

Here are a few possibilities.

FM amount: If an oscillator or filter can be modulated via FM (frequency modulation), you can create extreme effects by using an envelope to shape this
amount over time, resulting in percussive or sweep effects.

LFO amount: Sometimes, it can be cool to use an LFO to add a touch of vibrato to the beginning of a note or have it fade in as the note evolves, this can be
done with an envelope.

Hard sync: Classic hard sync sweeps – an effect that sounds a bit like an extreme flanger – can be created by activating hard sync and modulating the pitch of
the slaved oscillator only.

Not sure which oscillator to modulate?

Just activate oscillator sync, create an extreme envelope with a medium attack and decay, and apply the envelope to each oscillator individually (with high
amount values).

If the envelope sweeps the overall pitch, that’s the wrong destination.

If you hear a crazy “ripping” sound, that’s the correct oscillator, so make a note of it.

So there you have it, the essentials of envelope modulation.

Next up… LFO madness!

Re: Synthesizers for dummies


Post by Kurups on Oct 15th, 2008, 03:10am

Part 10: Essential LFO facts

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In the analog arsenal of modulation resources, low frequency oscillators – a.k.a. LFOs – are right up there with envelopes when it comes to usefulness and
flexibility.

While the most common uses for LFOs are vibrato, tremolo and wah-wah effects, there’s a lot more that can be done with LFOs — if you’re crafty about it.

In this installment, we’ll cover the specifics, along with a few more exotic applications to get your juices flowing.

What's an LFO?
The term “low frequency oscillator” is derived from the fact that audio oscillators operate within the range of human hearing, which is approximately 20 Hz to
20 kHz, whereas low frequency oscillators generally function below the frequency of 20 Hz.

While some LFOs can operate in ranges higher than that, we’re going to focus on the more traditional implementation described above.

Since they use slowly cycling waveforms, LFOs can be used to modulate the value of another parameter repeatedly.

The most common example of this type of modulation is vibrato: An undulation of pitch like that of an opera singer or violin.

By applying a sine or triangle wave LFO to the pitch of one or more oscillators, this “quivering” effect occurs.

In small amounts, the vibrato effect can be used to add a quasi-organic quality to accented notes in a sequence.

Larger amounts of pitch modulation can be used to emulate sirens or create wild electro sweeps.

Now that we’ve covered the essential concept, let’s move on to the specifics.

Common LFO Parameters


Rate

This is the speed of the LFO modulation.

Some manufacturers measure the rate in Hertz (cycles per second), whereas others use arbitrary numbers.

Almost all software synths include an alternate mode – labeled “sync” – that allows the LFO to be synchronized to the sequence tempo at various note values.

Depth/Destination

This is the overall amount of LFO modulation.

Some synths include the destination (cutoff, pitch, amp, etc) in the LFO section.

Others locate the LFO amount in the section of the modulated parameter(s).

Waveform

This selects the LFO waveform, which defines the overall character of the modulation.

If the waveform selection parameter is absent, this often means that the LFO will operate in sine or triangle mode only.

For example, Reason’s Subtractor includes an LFO – LFO 2 to be specific – that functions exclusively in sine/tri mode.

A list of common waveforms can be found later in this tutorial.

Delay

Delay – alternately referred to as “attack” – causes the LFO to start after a specified amount of time passes once the key is pressed.

This allows the unmodulated note to play for a time before the effect begins.

In practice, you can use this feature to allow staccato notes to play normally, while sustained notes receive the LFO treatment.

It’s also great for creating wiggly funk vibrato that engages after the note is held for a short time.

Keyboard tracking

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Sometimes abbreviated as “kbd” or “key”, keyboard tracking is a unique feature that allows the LFO speed to increase (or decrease, if negative amounts are
implemented) as a user plays higher notes on the keyboard.

When used with slower LFO rates, keyboard tracking can yield somewhat idiosyncratic results, but if the LFO can function in the audio range, it’s useful for
tuned ring modulation style effects.

Retrig

Old school analog LFOs often cycled continuously, regardless of whether a key was pressed.

As synthesizer technology progressed, it became possible to reset the LFO cycle every time a new note was played.

This is called “retriggering”, “key trigger” or “key sync”, and is implemented as a simple on/off switch on some synths, including the Prophet 08, various
Ableton devices, and Reason’s new Thor softsynth.

Offset

Offset is somewhat more exotic parameter that allows the user to define the point in the waveform that the LFO cycle begins, which is especially useful for
refining the sound of tempo-synced rhythmic effects.

Some manufacturers refer to this parameter as “phase”, which is fine as it makes sense as well.

Common LFO waveforms


The waveform of an LFO determines the overall character of the repeating modulation.

As described above, vibrato relies on sine or triangle waves, so applying a different waveform will deliver very different results.

Here’s a list of common waveforms and practical applications for each.

Most of these waveforms (taken from Ableton’s Sampler) are pictured to the right of each description.

Sine

Sine waves [first image, right] are best suited for organic, classic effects.

In addition to vibrato, sine waves can be used for tremolo when applied to amplitude (loudness) or cycling wah-wah sounds when applied to filter cutoff.

In conjunction with slow rates and high modulation depths, sine waves can be used to create sirens and alarms.

Triangle

Many LFOs – especially on older analog synths – rely on triangle waves [second image, right] instead of sine waves, since the end result is often
indistinguishable.

Accordingly, either wave can be used for the applications described in the sine wave section.

Square

Since the shape of a square wave [third image, right] consists of a jump between two specific values, square waves are used to create trill effects (an

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alternation between two notes, much like a UK police siren) when applied to pitch.

Applying a square wave LFO to amplitude or filter cutoff is a quick and easy way to create gated or pulsed effects, especially when synced to tempo.

Sawtooth

Sawtooth LFO waveforms [fourth image, right] come in two different flavors: rising and falling.

Rising sawtooth waves, when applied to pitch, create “woop-woop” effects like a car alarm.

If you slow this effect down to the minimum possible rate, with a huge pitch depth setting, then process the synth output with a long delay, you can create
those near-infinite pitch rises that are all the rage with the cool kids.

On the other hand, falling sawtooth waves applied to pitch are great for creating retro “booo-booo” sounds and laser zaps, if cheese is your weapon of choice.

Lactose-intolerant programmers will be more inclined to use falling sawtooths for rhythmic effects.

In fact, if you sync a falling saw LFO to eighth-notes and apply it to filter cutoff and/or amp volume, you can emulate the sound made popular by an artist such
as Deadmau5 with minimal effort.

Random

Random waves [fifth image, right] – sometimes referred to as “sample & hold” or “S&H” – shift randomly from value to value at the specified tempo.

Applying this waveform to amplitude adds a certain jitter to the volume.

When used on filter cutoff with a touch of resonance, it’s a resource for retro-style percolation.

Routed to pitch, especially with huge depth settings, random LFOs will create those hyper-goofy 1970s “computer” sounds from old sci-fi movies.

The secret to getting useful results lies in subtlety, creativity and avoidance of these clichés.

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Noise

Some LFOs allow you to use white noise as a waveform.

Applied to pitch in small to medium amounts, noise modulation results in distortion-like effects.

Applied to highly resonant filters, noise is an excellent resource for both glitchy crackling effects and organic sounds like rain and sizzling bacon (really).

Experimentation is key with this waveform, as it has much potential as a timbral resource.

Working with multiple LFOs

If your synth includes more than one tempo-synced LFO, you’re in luck.

By combining different note-values and assigning each LFO to a timbre or amp related destination, you can create polyrhythmic patterns with a minimum of
effort.

Here’s one possible configuration for a synth with three LFOs and key triggering set to “on” for all.

Assign a falling sawtooth LFO to filter cutoff using eighth-notes, then lower the overall cutoff frequency and increase the LFO depth to taste.

Assign the second LFO to oscillator pulse width, FM, or another waveform-related parameter, using a quarter-note triangle wave with fairly strong modulation
depth.

Assign the final LFO to amplitude/volume with a sixteenth-note square wave.

From there, experiment with different rates and destinations for each LFO and use your creativity to develop a sound that fits your style.

Re: Synthesizers for dummies


Post by Kurups on Oct 15th, 2008, 03:11am

Part 11: Grasping MIDI Controllers

In our previous tutorials, we covered the specifics of synthesizer audio modules such as oscillators and filters, as well as modulators such as envelopes,
LFOs and such.

These are the essentials for preset design, but for adding character to a performance, the next step lies in creative use of MIDI controllers.

In this tutorial, we’ll cover some of the techniques that allow you to morph the character of your synth parts over time.

Essential MIDI Concepts

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In terms of keyboard performance, there are two primary areas of MIDI control included on every synthesizer, be they soft or hard.

These are the MIDI note event and continuous controllers.

The MIDI note event is the data sent by a keyboard controller every time a key is pressed.

Continuous controllers are MIDI resources that can be assigned to various parameters so that you can change their value as you play.

For example, the pitch and modulation wheels are continuous controllers that allow performers to bend pitch (like a guitarist) or modulate the value of common
properties like filter cutoff or LFO amount via the mod wheel.

Other controllers include the damper pedal and aftertouch, as well as the various knobs and sliders found on popular keyboards like the M-Audio Oxygen 49.

A comprehensive list of MIDI continuous controllers can be found on the Borg website.

The MIDI Manufacturers Association website is an incredible resource for learning almost everything about MIDI.

Wikipedia also has an excellent description.

The MIDI note event


Every time a key is pressed, the keyboard sends out a MIDI note event.

This consists of three components.

First is the MIDI channel number, which can range from 1 to 16.

For hardware rigs, the channel number determines which synthesizer will respond to the note event.

For most current software synths, this is irrelevant, since the assignment is handled within the sequencer.

Next is the MIDI note number.

This is the number of the key being pressed.

For example, the MIDI specification defines middle C as number 60, so when you press this key, the note number sent is 60.

Finally, the velocity is sent.

This is how quickly (not how “hard” as some information mistakenly purports) the key is pressed and uses a range of zero to 127.

The more quickly the key is struck, the higher the value.

When a key is lifted, a velocity of 0 is sent, which turns the note off.

Using velocity in performance


The note velocity is an incredibly useful tool for adding dynamics to a sound.

Most commonly, velocity controls the volume of a sound, but there are quite a few alternate destinations that can be used quite musically.

Assigning velocity to control the cutoff frequency of a lowpass filter makes the sound brighter when the key is struck harder.

This is great for percussive sounds and drum samples.

Another filter destination for velocity control is the amount of envelope modulation for the filter cutoff.

Incidentally, the default preset for Propellerheads Subtractor synth (in Reason) has this destination pre-assigned, as well as a dedicated section (pictured
above) for routing velocity to a variety of useful destinations such as oscillator phase, FM amount and oscillator mix.

Some synths allow even more exotic velocity routings for the adventurous.

In this case, other useful parameters suitable for velocity include oscillator pitch (especially in conjunction with hard sync) and filter resonance.

If in question, check your synthesizer manual for more information on its capabilities.

Continuous controller applications


Continuous controllers are an even more powerful resource, as they allow for smoothly transitioning changes for multiple parameters, resulting in morphing
effects.

The most common controller for this type of application is the modulation wheel (pictured at right), while some synthesizers, such as those from Roland and
Korg, use a lever for this function.

In Reason’s Subtractor synth, the mod wheel can be simultaneously routed to cutoff, resonance, LFO amount, oscillator phase and FM amount.

This allows Subtractor to sound one way with the mod wheel down and another when the mod wheel is at maximum, delivering smooth morphing effects with
minimal head scratching.

The above diagram also shows Subtractor’s modulation options for other controllers like aftertouch, expression pedal, and the inimitable (and obscure) breath

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controller.

Ableton Live features the most comprehensive and intuitive implementation of continuous controller assignment on the market today.

By simply clicking the MIDI button in the upper right hand corner of Live’s interface, producers can instantly assign any controller to any parameter within Live –
including effects, synthesizer parameters and sequencer functions – simply by selecting a parameter “knob” and moving the desired slider or knob on their
hardware device.

As a DJ, I use this feature to create complex controller routings so that I can manipulate multiple parameters like filters, bit-crushers, delays and such as I
perform my sets.

We’ll discuss those possibilities in a future tutorial.

The bottom line here is to explore the options for real-time control on the synthesizers and software in your arsenal.

With a little experimentation, it’s possible to create incredible textures that can transform a breakdown from a necessary breather on the dancefloor to a peak
moment of the evening.

Re: Synthesizers for dummies


Post by Kurups on Oct 15th, 2008, 03:13am

II.

Part 1: Sampling Essentials

Now that we’ve covered the ins and outs of subtractive/analog synthesis, we’ll begin exploring other popular methods of tone generation.

This week, we’ll tackle the specifics of sampling.

Since most samplers also rely strongly on subtractive synthesis techniques, the material in the previous tutorials is equally relevant here.

If you haven’t already checked out the earlier lessons, you might want to do so before reading further.

On the other hand, if you’re already up to speed, it’s time to dive in.

What’s a sampler?
In simplest terms, a sampler is a musical instrument that records audio, stores it digitally, then allows you to trigger the recording and/or change its pitch in
real-time via MIDI using a keyboard, sequencer, drum pads, etc.

A sampler - or any digital audio recorder - converts audio to binary information via something called an analog-to-digital converter.

The a-d converter samples the voltages in an analog audio signal and converts those voltages into numerical data that the computer can store in memory or
on its hard drive.

From there, the data is played back via a digital-to-analog converter and the result is a replica of the original sound.

Much of the conversion process - and overall recording quality - is governed by two variables: sampling rate and resolution.

Sampling Rate

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The sampling rate is the number of times per second that the voltage is sampled.

In the same way that film records moving images by taking a series of still images and playing them back in rapid succession, a sampler records voltages at
the sampling rate, then plays them back in rapid succession, thus recreating the original waveform.

As with video and film, the greater the sampling rate, the better the sample quality.

In fact, the minimum acceptable sampling rate for musical audio is 44.1 kHz.

That’s 44,100 times per second.

44.1 kHz may sound extremely fast, but that’s because the highest frequency that can be accurately recorded is half of the sampling rate.

Since the human ear can detect frequencies all the way up to 20 kHz, 44.1kHz allows for frequencies up to 22.05 kHz to be accurately recorded.

If you’re a fan of math and physics, you can check out the Nyquist-Shannon sampling theorem for the intimate details on sampling rates.

Of course, there are those who feel that higher sampling rates can be detected by the average listener, but there are also several studies to the contrary.

In my ever-so-humble opinion, unless you have world-class monitors and an acoustically perfect space (and let’s be realistic, most clubs do not qualify in this
area) then 44.1 is a fine sampling rate for all but the most detailed solo acoustic recordings.

Sample Resolution

Sample resolution, sometimes referred to as bit depth, is a slightly different story.

The number of bits that are used to store the value for each sampled voltage determines the resolution.

That is, the more bits you have, the greater the range of numbers that can be used to describe a sound, since each bit increases the range of values
exponentially.

For example, one bit has two values, two bits have four possible values, three bits have eight possible values and so on.

Here’s another analogy: If you’ve ever used a CD deck for DJing, you’ve probably noticed that the BPM counters aren’t always correct.

One of the reasons for this inaccuracy comes from the fact that the counters do not include decimal places, so if you have one track that’s 125.1 BPM and
another that’s 125.4 BPM, the CD player’s detector will round the tempo of both tracks to 125 BPM.

Trainwreck time.

Sampling resolution behaves in a similar manner.

When the original audio fluctuates slightly between two values, a sampler has to round these values to the nearest available number, so the more numbers
you have available, the more accurate that rounding function will be.

What this means in plain English is that higher bit depths have better dynamic range, which means that the distance between the loudest sounds and the
quietest sounds can be even greater, thus the quality of a 24-bit recording is perceived as being “better” than a 16-bit recording.

Of course, all of this goes right out the window as soon as the track is mastered for a club, since compressors, limiters, maximizers and such actually reduce
the dynamic range of a recording so it sounds “louder” overall.

The video below does an excellent job of explaining how the mastering process has “evolved” over the past twenty years:

PART II

Part 1: Sampling Essentials

Now that we’ve covered the ins and outs of subtractive/analog synthesis, we’ll begin exploring other popular methods of tone generation.

This week, we’ll tackle the specifics of sampling.

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Since most samplers also rely strongly on subtractive synthesis techniques, the material in the previous tutorials is equally relevant here.

If you haven’t already checked out the earlier lessons, you might want to do so before reading further.

On the other hand, if you’re already up to speed, it’s time to dive in.

What’s a sampler?
In simplest terms, a sampler is a musical instrument that records audio, stores it digitally, then allows you to trigger the recording and/or change its pitch in
real-time via MIDI using a keyboard, sequencer, drum pads, etc.

A sampler - or any digital audio recorder - converts audio to binary information via something called an analog-to-digital converter.

The a-d converter samples the voltages in an analog audio signal and converts those voltages into numerical data that the computer can store in memory or
on its hard drive.

From there, the data is played back via a digital-to-analog converter and the result is a replica of the original sound.

Much of the conversion process - and overall recording quality - is governed by two variables: sampling rate and resolution.

Sampling Rate

The sampling rate is the number of times per second that the voltage is sampled.

In the same way that film records moving images by taking a series of still images and playing them back in rapid succession, a sampler records voltages at
the sampling rate, then plays them back in rapid succession, thus recreating the original waveform.

As with video and film, the greater the sampling rate, the better the sample quality.

In fact, the minimum acceptable sampling rate for musical audio is 44.1 kHz.

That’s 44,100 times per second.

44.1 kHz may sound extremely fast, but that’s because the highest frequency that can be accurately recorded is half of the sampling rate.

Since the human ear can detect frequencies all the way up to 20 kHz, 44.1kHz allows for frequencies up to 22.05 kHz to be accurately recorded.

If you’re a fan of math and physics, you can check out the Nyquist-Shannon sampling theorem for the intimate details on sampling rates.

Of course, there are those who feel that higher sampling rates can be detected by the average listener, but there are also several studies to the contrary.

In my ever-so-humble opinion, unless you have world-class monitors and an acoustically perfect space (and let’s be realistic, most clubs do not qualify in this
area) then 44.1 is a fine sampling rate for all but the most detailed solo acoustic recordings.

Sample Resolution

Sample resolution, sometimes referred to as bit depth, is a slightly different story.

The number of bits that are used to store the value for each sampled voltage determines the resolution.

That is, the more bits you have, the greater the range of numbers that can be used to describe a sound, since each bit increases the range of values
exponentially.

For example, one bit has two values, two bits have four possible values, three bits have eight possible values and so on.

Here’s another analogy: If you’ve ever used a CD deck for DJing, you’ve probably noticed that the BPM counters aren’t always correct.

One of the reasons for this inaccuracy comes from the fact that the counters do not include decimal places, so if you have one track that’s 125.1 BPM and
another that’s 125.4 BPM, the CD player’s detector will round the tempo of both tracks to 125 BPM.

Trainwreck time.

Sampling resolution behaves in a similar manner.

When the original audio fluctuates slightly between two values, a sampler has to round these values to the nearest available number, so the more numbers
you have available, the more accurate that rounding function will be.

What this means in plain English is that higher bit depths have better dynamic range, which means that the distance between the loudest sounds and the
quietest sounds can be even greater, thus the quality of a 24-bit recording is perceived as being “better” than a 16-bit recording.

Of course, all of this goes right out the window as soon as the track is mastered for a club, since compressors, limiters, maximizers and such actually reduce
the dynamic range of a recording so it sounds “louder” overall.

The video below does an excellent job of explaining how the mastering process has “evolved” over the past twenty years:

http://www.youtube.com/watch?v=3Gmex_4hreQ

Again, a noisy venue and a bunch of illicit substances will limit your perception of these distinctions, but you should still understand all of this stuff anyway,
since it applies to the techniques used to edit and process samples.

Again, a noisy venue and a bunch of illicit substances will limit your perception of these distinctions, but you should still understand all of this stuff anyway,
since it applies to the techniques used to edit and process samples.

Re: Synthesizers for dummies


Post by Kurups on Oct 15th, 2008, 03:14am

Part 2: Sampling Essentials

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In our previous installment, we tackled the specifics that govern the sample recording process, including sampling rate and sample resolution.

This time around, we’re going to cover how samples are edited and prepared for playback via a software or hardware sampler.

From there, we’ll go over how to make the most of your sampler’s resources.

Editing basics

Once a sound is converted to digital information, the creative possibilities expand exponentially.

At the simple end of the spectrum, you can slice a sample into tinier bits by adjusting the start and end points, then use it percussively or musically - either in
its raw form or processed by the sampler’s synthesis tools.

As your skills advance you can apply sample-looping functions to make a sound sustain indefinitely and/or zero in on exotic harmonic spectra that would be
difficult to generate using conventional means.

Sound designers and advanced users will eventually explore the possibilities of multisampling, which allows the creation of detailed instruments that can
capture the essence of both acoustic and electronic instruments, as well as serving as a method for creating drum kits and the like.

Start and end points


Once you have your sample loaded into your sampler, the first task is to set the start and end points.

These function as expected, with the start point determining where a sample begins playing and the end point determining where it stops.

This enables you to zero in on exactly the material you want to play.

What’s more, you can also create interesting transients by setting the start point in unorthodox ways.

Pictured above is a nifty example: You can add a clicky attack transient to any sound - especially 808 kicks and toms - by setting the start point to the first peak
in the waveform.

This works especially well with muted analog sounds, so if you’re after glitchy and/or minimal textures, this is a good place to get your feet wet.

Looping

While looping beats and musical recordings is a well-understood process, looping within a sampler is a very different approach.

When samplers first hit the mainstream in the early 80s, RAM was a luxury that few could afford.

Even by the mid 90s, a mere 32MB of sampler RAM was still considered downright extravagant.

In order to have sounds like strings or woodwinds sustain indefinitely, sound designers would use looping to take a fragment of the original recording and loop
that section, thus creating the illusion of a sustained note.

If you listen closely to single notes in older workstations and early softsynths, you can actually hear slight imperfections - like bumps or cycling – within the
sustaining portion.

These are loops.

The above picture shows a looped sample, with the medium green segment being the part of the sample that plays before the loop and the lightest green
segment being the loop itself.

Since most modern samplers include a pre-recorded set of sampled instruments, you no longer need to dive into the complexities of hiring an orchestra and

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meticulously sampling every note and articulation, then looping it.

But that doesn’t mean that looping isn’t a skill worth mastering.

One nifty looping trick for creating exotic textures can be accomplished within Ableton Live’s basic Simpler instrument.

Take any sound (really, any sound will work for this, even a snare drum or cymbal crash) and set the loop points for just after the beginning of the sound.

Next, create the smallest loop possible.

If you’re doing this correctly, you should hear the initial attack portion of the sound, followed by a buzzy, bright sustaining segment.

Now, slowly increase the length of the loop.

The pitch of the buzzy section should drop in pitch until you start to hear a rhythmic bumpiness.

Now, experiment with different loop lengths until you hear something you like.

Keep in mind that you may also have to adjust the transposition and detuning of the sound in order to keep it in tune with the rest of your track.

Once you’ve tackled the above experiment, you’ll have a better understanding of how looping works.

From there you can begin to explore more advanced looping functions.

Here’s a list of some other looping features and their uses.

Forward looping: This is the default mode for looping samples, with the loop playing forward until it reaches the end point, then starting over from the beginning
of the loop.

Bi-directional looping: Instead of starting over at the beginning, the loop reaches the end, then moves backward to the beginning of the sound, then forward to
the end and so forth. This is sometimes referred to as back-and-forth looping.

Crossfade looping: This parameter works well with longer loops by blending in a bit of audio from before and after the loop start and end points, which can help
to smooth the loop transition for certain sounds.

Snap to zero-crossing: More often than not, the best loop points are located at the point where the waveform crosses above or below the horizontal axis. By
turning on “snap” the sampler will auto-detect these crossing points, which often helps to remove clicks and pops from certain types of looped material.

Multisampling

One of the most intricate tools in a sampler is the multisampling function, which allows you to map many different samples to different ranges of a keyboard.

Most commonly, this is used to map different drum samples to various keys so that you can create a single preset for an entire kit.

In professional sound design, multisampling is used to create authentic sounding acoustic instruments.

Why?

Well, if you’ve ever tried to sample your friend’s guitar or violin, you’ve probably discovered that sampling a single note doesn’t necessarily translate across the
entire keyboard range.

This is because, like vinyl or tape, playing a sample at a higher pitch changes its speed and thus its formant content.

This is great for deep techno vocals or munchkin voices similar to those in Bass Kleph’s “Helium”, but not so great for accurate instrument sampling.

Accordingly, manufacturers have implemented multisampling – sometimes called “keymapping” or “zones” - which allows you to assign multiple samples to
specific ranges across the keyboard.

Pictured above is a five zone multisample from Ableton Live’s Sampler instrument.

Most modern samplers also allow you to also create zones for various velocity levels, so different samples are triggered depending on how hard you hit a key.

By sampling different articulations and pitches across the entire range of an acoustic or electric instrument, it’s possible to create convincing pianos,
orchestras and so on.

One final tidbit on multisampling.

Each sample that’s assigned to a zone has something called the root key.

If you look closely at the above zone image, you’ll see tiny little r’s in each region; these are the root keys for each zone.

The root key is the key at which a sample plays pack at its original pitch.

Since most samplers default the root key to C3, you will often have to adjust the transposition of this note to reflect the actual key being played.

Just a heads up…

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Tips for making a large sample consume less memory

Having a ton of sounds in your sampler – especially if it’s a hardware-based device – frequently requires the management of memory so all your samples can
fit into a single program or sequencing session.

Here are some approaches for helping conserve memory.

Truncate or crop unused data. If you’re sampling a tiny bit of audio from a longer recording, there’s probably a ton of data that’s simply not being used. Most
samplers allow you to delete the information from before and after the start and end points via tools labeled trim, crop or truncate.

In the above image, the green section represents the range of sound data that is being used, while the black data outside that range can be safely deleted to
free up some memory.

Note: To be on the safe side, move your start and end points outward in either direction before cropping, so you have some margin for error when fine tuning
your material.

Sum stereo samples to mono. Since a stereo sample consumes twice as much memory as a mono sample, it’s often worth the extra time to convert the
sample to mono (usually in a separate audio editor) before loading it into your sampler.

Sample at a lower sampling rate. As discussed in the previous tutorial, sampling at rates higher than 44.1 kHz (or 48 kHz if it’s a video source) really doesn’t
yield that much of an improvement unless it’s a pristinely recorded acoustic instrument. It also consumes more CPU and RAM due to the fact that more
samples are taken at a faster rate.

Sample at a lower resolution. More bits consume more memory and CPU, so again, unless the sample requires intimate detail, you can usually get away with
16-bit samples – and if you’re into distorted lo-fi samples, feel free to sample at 8-bit if your sampler supports it. Just be aware that once the bit depth is
lowered, there’s no going back to a cleaner sample.

Once you have your samples properly edited, you can then use subtractive synthesis tools to further shape the sound.

Next up? The arcane world of FM synthesis.

Re: Synthesizers for dummies


Post by Kurups on Oct 15th, 2008, 03:15am

III.

Part 1: How to use compression

In our first installment on audio processing tools, we’ll tackle one of the most basic – yet seemingly arcane – tools in a producer’s arsenal.

The compressor.

Correctly applied compression is one of the most useful techniques for bringing a track to life, adding impact and character to everything from bass and drums
to an entire mix.

The underlying principles are actually pretty easy to understand, once you familiarize yourself with the various parameters and terminology.

So let’s do that.

How compressors work

Compressors affect the most basic aspect of sound: Volume.

In studio parlance, we use the term “dynamics,” which is just a fancy word for how a sound’s volume changes over time, like the immediate crack of a snare
or punch of a kick.

Another example might be a chord progression that varies slightly in level as the sound evolves.

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By applying a compressor to these signals, we can modify their dynamics to bring them in line with our production objectives.

The essential concept in understanding this process is that a compressor lowers the volume of a signal when that signal’s volume exceeds the level
determined by the compressor’s threshold setting.

The amount and rate of that volume reduction is determined by various other parameters with names like ratio, attack, and release.

Finally, many compressors offer different options for modifying the character and performance of the process so that it can be fine-tuned for a wider variety of
production applications.

With names like peak, RMS, soft knee, hard knee, lookahead and sidechain, it’s easy to see why so many producers just turn the knobs till things sound
“good.”

But actually knowing what each knob does is the domain of the pros, so here’s an overview to get you started.

Compression parameters
Threshold

This is the key parameter that governs the overall behavior of the compressor.

Threshold determines the volume at which compression kicks in.

Simply put, when a signal’s level exceeds the set threshold, the compressor activates and begins lowering the volume.

Ratio

Next in importance is the ratio parameter.

This determines how much the compressor will lower the volume once the threshold is exceeded.

For example, if a compressor’s ratio is set to 6:1, then when the threshold is exceeded by six decibels, only one decibel will pass above the threshold.

Extreme settings, like 10:1, allow only one decibel to pass for every ten.

Note that for many producers, any setting above 10:1 is considered “limiting.”

That is, at the highest settings, a compressor turns into a limiter, which serves the same essential function, but in a much more extreme manner.

Attack

As with synthesizer envelopes, the attack parameter determines how quickly a characteristic changes.

In a compressor, the attack parameter controls how quickly the processor lowers the volume after the threshold is exceeded.

Here’s a practical use: If you’re using a compressor on a drum and you want some of that drum’s original attack to cut through, set a slower attack so the
compressor doesn’t clamp down too quickly.

Release

Once the signal passes back below the threshold, the release parameter determines how long it will take for the compressor to stop manipulating the volume.

At this point, you should take a few moments and experiment with all of the above parameters, paying close attention to the attack and release amounts with
high ratio settings.

If you’re not careful with these parameters, is all too easy to take a wonderful, lively sound and squish it into a lifeless mess, so spend some time learning how
these parameters affect a wide variety of sounds.

Use drums. Use bass. Use vocals, pads, or any other instrument that’s part of your style.

Once you have a basic grasp of your compressor’s sound, the next round of parameters will make a lot more sense.

Peak/RMS

Many compressors include a Peak/RMS switch.

This affects the behavior of the threshold control.

In peak mode, the threshold reacts quickly and strongly to sudden volume changes, which can be quite handy for drums and percussion.

RMS mode causes the threshold parameter to adapt to the overall average of the volume shifts, which is useful for more consistent signals like voice, pads
and even full mixes.

Knee

Another common parameter that shapes the sound of a compressor is the colorfully titled “knee” parameter, which is so named because it changes the shape
of the threshold response – looking a tad like a knee.

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The screenshot above shows Live 7’s compressor with a softened knee setting.

Hard knee compression is the usual default, but soft knee mode is a much more pleasing sound for many producers.

Here’s why.

When soft knee mode is active, the compressor gradually kicks in as the signal approaches the threshold rather than simply engaging instantly as the
threshold is crossed.

Again, there are valuable uses for hard knee mode, like drums and compression.

But when you hear producers talk about “transparency” there’s a good chance they’re referring to a soft knee mode – often in conjunction with the RMS mode
mentioned previously.

Lookahead

Lookahead is a parameter that’s found exclusively in compressor plug-ins, since implementing it in hardware would defy the laws of physics.

What lookahead does is analyze the sampled audio before it arrives at the compressor, thus giving the compressor a “heads up” that a loud signal is on its
way.

Like an audio crystal ball, of sorts.

Ableton Live’s Compressor device includes a lookahead function that can be set to 1 millisecond, 10 milliseconds, or “off.”

For what its worth, I just leave it in 1 millisecond mode and forget about it.

Maybe I’m just lazy

Sidechain

And now, the moment many of you have been waiting for… Sidechaining.

Sidechaining is one of the coolest features on any compressor, since it allows a second signal to control the compressor’s behavior as it processes the
primary signal.

What the heck?

Here’s the most common use for sidechaining: kick and bass.

Since many kick drums share the same frequency range as the bass line, it’s quite common for the two elements to get in each other’s way.

By using your compressor’s sidechain function, you can use the volume of the kick drum signal to lower the volume of your bass sound every time the kick
drum hits, thus creating room for both instruments in a push-pull manner.

In doing so, you’ll immediately be rewarded with a sound that you’ve heard on countless tracks to great effect.

We highly recommend spending time experimenting with this specific technique, so to get you started, check out the screenshot to the right.

And don’t just stop with bass, either. Try using a kick drum as a sidechain input on sustaining sounds like pads, as this will give those tracks that “bounce” that
defines tons of trance and progressive house classics.

Now go compress something.

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Re: Synthesizers for dummies


Post by Kurups on Oct 15th, 2008, 03:18am

Part 2: : How to use a delay

Right up there with compression and EQ, delay-based effects are arguably the most ubiquitous tools in a producer’s arsenal.

Most musicians associate the word “delay” with repeating echo effects, but echo is just one of the many applications for delay.

In this tutorial, we’ll reminisce a bit about the history of these tools, then take a closer look at the underlying technology and its applications for producers.

History of the delay

The first use of delay was pioneered by the French artists who led the Musique Concrète movement way back in the early 1950s.

This high-concept - and utterly revolutionary - musical movement made extensive use of tape loops to repeat audio recordings, creating rhythms and drones
out of seemingly ordinary sources like water droplets and nature sounds.

Sound familiar?

Many historians agree that Musique Concrete was the first true electronic music, using tape in much the same way that we now use samplers – but over half
a century ago!

Later in the 50s, ambitious inventors like Ray Butts and Mike Battle turned the tape loop concept into usable products like the Echosonic and Echoplex
(pictured below).

These first delays were composed of a small tape recorder that played a continuous loop of tape.

By varying the position of the record and playback heads, as well as changing the tape speed and varying a few other parameters, intrepid producers could
adjust the delay time, number of repeats and other signal characteristics.

These early tape-based delays have such a unique character that they are still highly sought after on eBay, often fetching hundreds – even thousands - of
dollars.

In the late 1970s, simple analog circuits called “bucket brigade devices” simulated tape delay with equally unique results, but this was quickly superceded by
digital circuitry, resulting in the delay as we know it today.

How a digital delay works

While tape and analog delays have their own unique character and depth, digital delay technology ratchets up the entire concept by an order of magnitude,
especially when you factor in the massive amounts of memory available in current computer technology.

Here’s how it works.

A digital delay functions as a simple audio recorder by capturing an audio signal, holding it in memory for a specified time, then playing it back, thus creating a

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single “echo. “

One of the keys to the echo process lies in blending the delayed signal with the original, unaffected audio.

The signals are known respectively as wet (delayed) and dry (unprocessed).

Without the presence of the dry signal, there would be no echo - just the original sound played back a little later.

To make the echoes repeat multiple times, a parameter known as “feedback” is implemented.

Feedback sends a bit of the delayed signal back into the input of the delay, causing it to continually repeat at progressively quieter volumes until the echo fades
to silence.

Adventurous producers sometimes set the feedback level at extremely high amounts, creating a wash of echoes that fade out over a much longer period of
time.

This technique is often used during breakdowns, adding tension and drama to a track’s overall story.

Types of delay effects

Since delay times can range from fractions of milliseconds to several minutes, there’s an extraordinary range of results available.

Here are some of the more common effects that can be created by simply varying a few parameters.

Resonators. Sometimes referred to as a “static flange,” you can create metallic, ringing pitches by setting the delay time to very short lengths and increasing
the feedback to near maximum, thus causing the delayed signal to repeat so quickly that it generates a pitched tone.

If you’re mathematically inclined, you can calculate the specific pitches that the delay produces in this manner, here’s how.

When a delay is set to one millisecond (one thousandth of a second) with a very high feedback amount, the delayed signal repeats one thousand times per
second.

Since frequency is measured in Hertz – a measurement that correlates to one cycle per second – a delay that’s repeating one thousand times per second will
have a frequency of 1000 Hz, also known as 1 kilohertz (kHz).

To drop that pitch by an octave, just do a bit more math and double the delay time to two milliseconds.

Since two divided by one thousand equals five hundred, a two-millisecond delay will result in a pitch at 500 Hz.

For those who aren’t as mathematically inclined, just fire up Ableton Live’s Resonators device (pictured above).

The Resonators device applies these same principles, but mercifully, it allows users to dial in multiple pitches using standard musical terminology, making the
entire process a relatively painless affair.

Flange/Chorus. It may come as a surprise to some, but these classic modulation effects are created by simply using an LFO to modulate the delay time as
the signal passes through the effect.

The main difference between a flanger and a chorus is that flangers use high feedback and extremely short delay times, usually under ten milliseconds but
sometimes as much as twenty milliseconds.

Chorusing and flanging share overlapping delay times, so there is no clear point at which one becomes the other – it’s more of a morphing thing.

That said, chorus times can range as high as forty milliseconds, after which the effect veers into a territory that is sometimes referred to as “doubling” by
some producers.

In a future tutorial, we’ll go into much greater depth on the applications of various modulation effects like chorus, flanging, phasers, and such.

Slapback. Fans of old school reggae and dub genres will be familiar with an effect called “slapback,” which is essentially a very short echo with a lot of
feedback and a touch of filtering.

To recreate this effect, try tinkering with delay times in the 90 millisecond range and vary the feedback amount.

If your delay plug-in supports it, spend a bit of time fiddling with its EQ or filter, as this will allow a wider range of textures.

As you increase the delay time, the slapback effect will become more distinct until it transforms into the echo effect we all know and love.

Other delay variations

spectrum of the echo via a simple EQ/filter tool.

Multitap. Multitap delays up the ante further by giving the user access to even more individual delay lines, each with its own panning, feedback and often,

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filtering characteristics.

Pictured at the top of this article is Logic’s Delay Designer, which is currently the most sophisticated delay to ship standard with any DAW software.

Some multitap delays include complex routing options that allow each of the individual delays to feed into and thus interact with the others.

A multitap delay gets really deep, really fast, so it’s wise to master the basics before taking the plunge – unless you’re fine with sticking with presets until you
get up to speed.

Reverb. The ambient effects created by reverb are produced by extremely specialized and complex delay algorithms, optimized to simulate the acoustic
characteristics of various rooms and spaces.

Since the reverb process is a world unto itself, we’ll be devoting an entire tutorial to its subtleties in the not-too-distant future.

Until then, we’ll leave you with one extremely important tip to contemplate.

Since repeating delays actually add additional audio signals to your mix, it’s all too easy to turn your track into a mushy mess by using too much or too many
on a number of tracks.

Instead, we recommend limiting the use of echo effects to only one or two specific instruments that you want to highlight and enhance.

This way, you make the most of what a delay has to offer without overwhelming the other parts of your track.

Until next time…

Re: Synthesizers for dummies


Post by Kurups on Oct 15th, 2008, 03:30am

Saved for fucking fun

that shit took way too much longer than i thought it would

Re: Synthesizers for dummies


Post by Heraclys on Oct 15th, 2008, 06:17am

You're a hero!

That's a big contribution you just did to the forum dude...

Thanks man!!

Re: Synthesizers for dummies


Post by slinky on Oct 15th, 2008, 10:27am

Take that Taylor!

Re: Synthesis for dummies


Post by lukas. on Oct 15th, 2008, 6:49pm

hey could you get us the link to download this as a pdf or something? thanks

Re: Synthesis for dummies


Post by Kurups on Oct 15th, 2008, 7:46pm

Yeah i'll convert it all to a PDF later on tonight

Re: Synthesis for dummies


Post by Alpha5 on Oct 19th, 2008, 4:48pm

Bump

Re: Synthesis for dummies


Post by Kurups on Nov 8th, 2008, 11:02am

does anyone know of any good videos / tutorials that go further into the use of LFOs and envelopes - cause that shit blows me away as far as how hard it is to
figure out

Re: Synthesis for dummies


Post by Kyran on Nov 8th, 2008, 11:47am

What don't you understand?


Attack is the time it takes to get to the peak after you first press the key.
Sometimes there's a hold time, where it stays at 100% for a bit.
Then there's a sustain level. This is what the signal levels off to after the decay time.
Once you release the key, the release time is how long it takes to get back to 0%

Sometimes there's an envelope amount which affects the scale of the whole envelope. (ie. only makes the peak go to 50% or reduces the depth of the
envelope to half) This is useful with filter envelopes.

If it's an amp envelope, after the attack time, the signal will be at full volume, and then will decay to the sustain level, if it's less than 100%

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For a filter envelope, it's the same thing except it's the cutoff frequency, not the volume. You can change the range of frequencies it sweeps between with the
env. amount control and the sustain level.

LFOs just cycle between high and low. Use them on pitch to get vibrato, or on a filter for a continuous sweep. Sometimes LFOs have an attack time too, which
means they don't kick in right away. With this you would play a note and then it would apply vibrato after holding it for a little while.

Re: Synthesis for dummies


Post by Kurups on Nov 9th, 2008, 01:16am

i'm definitely asking the wrong question.

on the es2 synth i just looked, it has 'vectors'

those are i'm pretty sure what I'd meant to say.

Re: Synthesis for dummies


Post by arob on Nov 10th, 2008, 2:12pm

Kurups,

Router tutorial: http://www.youtube.com/watch?v=98wmzd6EKPg

Vector Tutorial: http://www.youtube.com/watch?v=M8zGcujsgd8

SFLogicNinja lays it out better than anyone else on the net.

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