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Before we exactly tackle what the stereo is, let us have a peek at its modest beginnings. In the early 1930s, Harvey Fletcher, a scientist at Bell Laboratories, experimented with the formation of sound through stereophonic recording and came up with the concept of “Wall of Sound.” Hundreds of microphones are placed side by side in front of an orchestra, and act as loudspeakers for the benefit of listeners.

But it was during the late 1950s, when the idea of creating sounds was brought to new heights.

Monaural recording (a single-channel method of recording sounds) was introduced to the public. The method of simultaneous playing and recording sounds was still unheard because of the limited technology of the time. But still, it was a great invention, since it served as the fuel for manufacturers to further improve the system.

Later on, an assembly of amplifiers and speaker systems that played two or more channels at the same time was introduced to the public. The device is called a stereo. It can imitate quality sounds through multi-channel audio recording and playing.

A stereo is a device that reproduces sound through two audio channels or speakers. Depending on the way it is built and the features it comes equipped with, stereos can play music, record sounds or play and record – known as the dual channel sound reproduction system – all at the same time.

The basic elements of a stereo are a CD or cassette player, amplifier designed to deliver balanced electrical impulses to the speakers that come out as audible sounds. Speakers produce the sounds that you can hear. They come in various shapes and sizes — from the dynamic loudspeakers down to tiny earphones, tuners and equalizers.

The height of satisfaction and entertainment a stereo provides depends largely on the functionality of the stereo system that you have. Although, all stereos are designed to mimic a pleasant sound quality that you will enjoy listening to.

Considering the extent of avant-garde technology we have now, you can expect added features and functions incorporated in stereos that are all designed to further enhance your listening experience. Take for instance the different types of music systems like home theaters, digital sense-surround systems and audio technology designed for cars and those that come with USB ports and plug-ins.

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…

Conference, discussion or ‘push to talk’ microphones provide the best conference audio for applications beyond the immediate public address system.

Though sometimes not the best in absolute sound quality it is the ability for the microphones to be controlled by the delegate as well as the technician that maximises the chances that every word is heard.

For this reason conference microphones are particularly suited to recording meetings and in the provision of the ‘original’ feed to simultaneous interpreters.

In both these applications it is best to have as few microphones live as possible at any one time. Each ‘open’ microphone adds more back ground noise to the audio without enhancement.

The operation of a standard sound desk with just 6 conventional microphones on a top table is to leave all 6 microphones partially live so if someone does speak the technician need only increase the level of an already live microphone. This allows at least some audio, albeit low level to begin with, to be heard which is better than nothing. This is the practice because the technician does not know who is going to speak next and the speaker is unable to make their microphone live independent of the technician. There are two main reasons why this is not acceptable in a recording or interpretation environment.

Firstly the delay in a delegate speaking and in the microphone becoming live, as the technician recognises which microphone is in use and increases its level, introduces a pause into which whole sentences can disappear and critical meaning can be lost. Conference microphones can be operated by the speaker directly so eliminating this delay in making the microphone live or louder. This creates a freer flow of dialogue to take place without a word being lost. This is critical in the recording of important meetings and in the original language feed for simultaneous interpreters.

Secondly by having so many partially live microphones live you are introducing unwanted and distracting background noise into the audio. This is often not recognised by the delegates or by the technician who generally listen to the audio via the room acoustic in which these noises are already present. However other users like interpreters listen to the proceedings via headphones so they hear very close to the audio and can clearly hear the rustle of paper or the pouring of water. This unwanted audio is very distracting and masks the audio.

With conference microphones placed on lecterns you have the capability to control and eliminate these sounds without losing the critical audio needed by only having one or two microphones live at any one time. And this advantage is possible in a system for 50 – 100 participants yet using conventional microphones and a mixer is difficult on a top table with just 6 speakers in a panel discussion.

The advantages in using a conference microphone or discussion system are very clear (especially the audio).

Web Audio Plus is a software program that converts your audio files into a file that you can use on your web site that will stream. Web Audio Plus is a great little program that I like a lot and use almost every day.

This program is super simple to use and has a few easy steps to produce a finished streaming audio file complete with buttons ready to insert in your website.

Overview

Basically to use Web Audio plus you have to either use a microphone and record your audio into Web Audio Plus or load an mp3 or wav file into the program.

Upload your audio file the program creates to your web server.

Copy and paste the code Web Audio Plus generates for you into your web page.

Steps To Take For A finished Streaming Audio File

Select Recording Device: The first thing you need to do is decide whether you are going to record your audio from within the software or import an already recorded file.

If you decide to record from within click next and hit the record button and start recording. When you are happy with the recording hit stop and play back your new recording.

Choosing a file you have already recorded to upload is as simple as browsing for your file and clicking open.

Which method you choose to get your audio into the program is up to you but I have found the medium quality works well. This is a good compromise between file size and sound quality.

Choose Buttons: Your next task is to choose the buttons you would like to use. You have lots of choices in color so pick a color and style that will look good with your website design. The color chart on the left is what color the button will be and on the right is what color the background will be.

There is a preview button near the bottom to see what your choice will look like.

Review Options: This screen will let you choose where to save your finished file and what size the buttons will be. The default on the button size is 200 by 73. I usually change this to 100 by 73 as I think it looks better.

Pick the location where you want to save your finished file. The default is your desktop, which is ok with me, then copy your html code by clicking the button. Click the make button and like magic your streaming audio buttons are ready for use.

Upload your finished files to your website and change the location in your code to reflect the path where you are storing the files and you are good to go.

Copy the code Web Audio Plus provides into your web page and your website should talk or sing.

Recording an MP3 allows musicians to sell music directly to their fans, without going through the “middle management” layer of a record company.

Now, you still have to have a good sound, and good production values to make an MP3 file work right – like anything with music recording, the studio time, and the post production time, is critical to your success. In case it’s been a while since you kicked it out in a studio, let’s run through the basics.

Quality input determines the quality of your sound output. Quality input depends a lot on the recording environment; MP3′s encoding means that a less than perfect recording will get significantly worse after digitization. So, make sure when recording an MP3, that the sound quality of your studio is as good as you can make it; it’s worth it to spend the time to do some sample recordings and stamp out the dead spots.

Next to sound absorbers and reflectors on the walls to kill unwanted echoes, it’s your microphone that drives your recording quality. Get good microphones; you’ll want a good condenser microphone in front of your lead singer, and two dynamic range microphones near your instruments. Like anything that has to deal with music, as you get more experience, your microphone collection will grow. (In particular, mic’ing your drum kit is going to be a fascinating experience.)

Placement of your microphones for a good recording is a mixture of science and acoustics, and art – and is something you’ll spend a lot time of trial and error. Take a look at general articles on studio setup to get the most out of this, and measure your recording area carefully. It’s amazing what a couple of feet of placement can do. No one ever said recording an MP3 was going to be easy, but believe me, in the end… will be well worth the effort.

Once the gear is set up, decide how many tracks you plan on laying down. 4 is the minimum; fortunately, modern digital recording gear lets you do this with a heck of a lot less hassle and fuss than you’d expect even from five years ago. Apple’s GarageBand software allows 12 tracks with trivial ease, and is a far cry from recording on 2″ tape. As always, it’s the time at the mixing boards that eats album creation. Look into setting your fades, and cross track and timing beats down early and work from within them.

Getting your recording from the studio equipment to your computer for MP3 conversions depends a lot on what you recorded it on. If you used MiniDisc, or most digital formats, then it’s already on a hard drive and you can just use digital audio in to move it over. You’ll either use an S/PDIF cable or a firewire cable for this, since the raw files are huge. If you’re stuck with an analog recording, most sound cards have analog input. You’re going to spend a lot of time cleaning up an analog signal going through your soundcard, and you need to be careful about clipping off the highs and flattening out the middles. It’s almost always better to go straight digital across the board. Saves you time, gets a higher quality output, and is easier to deal with.

Once you’ve got your digital music down, it’s time to rip it to an MP3. Now, MP3 as a format that doesn’t support multi-track audio beyond stereo sound, so be sure to convert down to that when you’ve got it to the stage you need it at. You’ll also need to select a bit rate, which is a measure of how much data is played per second, and directly impacts sound quality and file size.

If you’re trying to capture CD-quality sound, lots of track overlays and a complex arrangement, look at 192 kb/sec. It’s a large file size, but everything you want in the music will be in the download. Standard for most digital music is 144 or 128 kb/sec. If you’re recording an interview with just people talking, or people reading text (such as an audio book), you can (with a good, quiet recording area) make a lot of progress with 64 kb/sec bit rates to kind of give you a comparison.