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Timbre (or Sound Quality)

The goal of this article is to help develop knowledge of basic acoustic principles. This in turn will help you to understand, and eventually master, the basic techniques of sound engineering and recording. Each section has a theme that is first defined in technical terms and then commented on in practical terms in respect to audio equipment.

Definition of timbre
Timbre (pronounced /tam-ber’/) is a sound’s identity. This identity depends on the physical characteristics of the sound’s medium (the matter or substance that supports the sound). Let’s take an A at 440 Hertz produced at 60 decibels: we can immediately tell if the sound was emitted from a violin, saxophone, or piano. Yet, even though the instrument is different, it’s the same note and the same amplitude. The difference is in the sound production: string, air column, etc.. Plus, the sound isn’t generated by the same “tool”: a bow for violin strings, a reed and an air column for the sax, and felt covered hammers that strike the piano strings. It’s the different physical characteristics of the medium and the « tool » that determine the characteristic sound waves in each case. Later we will also see how a sound chamber adds another dimension to this definition.

Waveform

The most basic waveform is a sine wave (sinusoid). It could be considered the atom of sound. Pure sinusoidal sounds are rare ( tuning forks, drinking glasses being rubbed) and were considered to have strange powers over human behavior at one time. Most sounds that surround us are of a more complex nature.
This means that inside a sound, that we perceive as being unique, there is a superposition of many sine waves that have, in a way, fused together to become one sound. It’s the nature of this superposition itself that determines the resulting waveform and that is responsible for its timbre. This is called a spectrum.

Spectral Representation

There are many ways of graphically representing sound. For instructional purposes we have chosen to use a spectrogram for its clarity and simplicity.

Horizontally: time in seconds. vertically: frequency in Hertz. A sine wave (sinusoid) at 100 Hertz is represented by a horizontal line at a height corresponding to 100. A harmonic sound at 100 Hertz is represented by superimposed lines corresponding to sine waves of 100, 200, 300: n x 100 Hertz. The length of the lines represent the length of the sound.

Noise

Let’s imagine a case where all sine wave frequencies that are perceptible to the human ear (from 20 Hertz to 20 kHertz) and having the same amplitude, are “mixed” into one sound signal. We get what is called “white noise”, or in other words “hiss”. If the white noise is very short we would perceive it to be a kind of short percussive sound. Consonants belong to this category, in the same way that a sound medium that receives the attack of the “tool” which “kick-starts” it, produces as noise. This noise corresponds to the time it takes for the sound wave to stabilize and take its final form. The “rubbing” of a bow on a string is similar to a hissing sound, while a hammer hitting a piano string is similar to a percussive sound. These notions will be dealt with in greater depth when we get to envelopes and transients. In the case where a series of noise frequencies is contained between certain limits we will refer to them as noise bands.

If a zone is particularly swollen in energy, then we can speak about colored noise around that zone. Pink noise is white noise with a power density that decreases by 3 dB per octave.

Harmonic Sound

Having already highlighted the superimposed or complex aspect of sound, we are now going to focus on a specific category of frequencies in a sound spectrum: harmonics. A harmonic sound is a sound which contains sine waves that obey the mathematical law called the Fourier series. This law translates as follows: A complex periodic signal is made up of a certain number of component frequencies that are integers of the fundamental frequency.

An example of a harmonic sound: a sound at 100 Hertz in which the component waves are 100; 200; 300 ; 400 ; 500 ; 600 Hertz. The perceived pitch is the lowest frequency: 100 Hertz. The following component waves (2 x 100, 3 x 100, 4 x 100, etc.) are calculated on integers and are called harmonics. The lowest frequency, on which they are based, is called the fundamental. The number , or “rank”, of a harmonic is the integer by which the fundamental is multiplied. For example the 3rd harmonic would be the one at 300 Hz.

The pitch of a harmonic sound is easily perceptible to the ear, and these sounds usually have an “in tune” quality about them. That’s why melodic musical instruments are designed with the goal of producing harmonic spectrums.
Noises, like those we referred to earlier, are aperiodic signals. They are characteristic of percussion instruments for example.

The distribution of energy in the spectrum

Regions of relatively great intensity in a sound spectrum are called formants. In the case of a band of consecutive frequencies it is referred to as a formant zone between x and y Hertz. This distribution of energy plays an important role in the perception of timbre, as do the number of components in the spectrum, their distribution, and its regularity or non regularity.

EQing on a console

It’s the EQ section of a console that will allow us to tweak or correct timbre. Depending on the model, the EQ section is more or less sophisticated and offers different possibilities of adjustment. We won’t be dealing with simple high/low EQ knobs or switches that you can find on hi-fi amplifiers or entry level mixers which are only meant to adapt a sound to a specific listening area. We’re more concerned with the EQ controls that are found on small modern digital models or part of most major recording software. We must keep in mind that EQ is mainly used for one reason…to correct, and not in the hope of improving the recorded signal: you can never turn a mediocre recorded sound (due to bad placement of the mic or even the quality of the mic itself) into a great sound by just using EQ. Equalizers split the audible frequency range( 20 Hertz to 20 kHertz…) into many sub-ranges. Thus one generally talks about highs, medium highs, low mids, and lows. The first thing to do, then, before tweaking any knobs, is to determine in which frequency range the problem lies, then after that, the nature of the problem. Is it due to too much coloring that wasn’t detected during the recording process, a parasite due to the environment, or a masked effect due to the presence of other instruments…

What does it look like?

Equalizers are…harmonic and partial filters. Their specificity lies in the fact that they not only can get rid of component frequencies, but that they can also amplify chosen frequency zones. Of course, if there isn’t anything in the signal in that range, only hiss will be added! Good EQ sections generally have 4 bands. Each offers at least 2 controls: frequency adjustment and gain. These are called semi-parametric. There’s often a third setting called the bandwidth or “Q” which has the purpose of enlarging or tightening the frequency range (bandwidth) of the filter. When this 3rd control is present, the Equalizer is then called a parametric equalizer. Frequency adjustment will be tweakable between the upper and lower limits of the sub-range of the filter (with software these limits no longer exist!).

How to Modify Timbre

You must always keep in mind that all EQing on an instrument will be destructive with respect to the recorded sound, just as the latter is also, in many cases, an imperfect copy, of the original. So one must be careful! Before touching anything, think about what you want to accomplish with EQing: I want a “warmer” sound, I want to cut the bass, I want my instrument to stand out in the mix, I want to get rid of that annoying resonance that came from the studio…

Audio effects, we all know what they are, sort of. They are used to manipulate audio in ways that are not available with traditional playing and recording techniques. If you’re like me, and enjoy dabbling in audio production, you’re probably familiar with all the basic effects. Reverb is one of them, and probably the most easy to explain; it adds space to your audio. Delay on the other hand, is a little bit more difficult to explain. Again, if you’re like me, you want to fully understand how these effects work, so that when you go to use them you know them inside and out. Today’s article we will be discussing reverb and delay, how they work and why they work the way they do.

Reverb

Sound produced in an enclosed space, reflects off of surfaces and blends together, creating reverberation (reverb for short). So, basically, reverb is the reflection of sound waves from a solid surface to our ears. It is most easily identified when the sounds stops, but you continue to hear the reflections as they decrease in amplitude. Large rooms or chambers are some of the best producers of natural reverb. There are a few different types of electronic reverberation mechanisms that produce reverb artificially. There types are:

1. Plate reverberators – This type of reverb uses large metal plates suspended by strings, which are in turn inside of damped cases to manufacture the effect. Transducers are used to apply a signal to the plates, and electronic pickups are then used to convert the plate’s vibrations to an electric signal.

2. Spring reverberators – These reverberators are similar to plate reverberators, except instead of using plates, springs are used instead. Spring reverberators are often integrated in instrument amplifiers, and are considered to be the most artificial sounding reverb types.

3. DSP reverberators – DSP reverb units use signal processing algorithms to create the reverb effect, using long delays, envelope shaping, and other processes. This type of reverb is the most widely used and the most flexible form of reverb.

4. Chamber reverberators – This is the most “natural” form of reverb, but can also be made artificially. Chamber reverb is basically a room with solid walls, a loudspeaker at one end, and microphones at one end. The audio is played through the loudspeaker, bounced off of the walls, and then recorded by the microphones.

Delay

The basic delay effect records an input signal, and then plays it back after a set period of time. The first wave of delay used reel-to-reel magnetic recording systems and tape loops to produce the effect.

1. Analog Delay – This was the first type of delay employed in the audio engineering field. One type of analog delay unit used magnetic tape as the recording and playback medium. Motors would guide the tape through the device, with different mechanisms modifying the effect’s parameters. The tape used in this type of delay would break down after a while, so the tape would have to be replaced from time to time to maintain fidelity of the audio. Other types of analog delay used magnetic drums, or spinning magnetic discs instead of tape as a storage medium for the audio information. The main advantage to these types was the increased durability of the storage medium.

2. Digital Delay – This type of delay unit became popular in the late 1970’s. But, at the time, were only available in the form of an expensive rack mounted unit. The BOSS DD-2 changed that in 1984, as it was now available in the form of an affordable foot pedal. Digital delay works by sampling the piece of audio being processed, recording the bit to a storage buffer, and then playing back the bit of audio based on the parameters set by the person using the unit. There are many different types of digital delay units that offer different digital signal processing options, so I can’t really expound on anything in that area. But in my opinion, digital delay effects units seem to be the most powerful and flexible of the two types. Many guitar players use this effect, although some people believe that digital delay sounds a bit artificial compared to its analog counterpart.

This is the first part in my continuing series on audio effects. I’ll be covering some of the more standard effects first, like today’s subjects, and then move on to the more advanced effects later on. I hope that this shed some light on the subject, making your next foray into audio recording or editing a little easier and more fun.