Audio Tech Talk

[Sarong1]

System Noise Explained

I found this article interesting and relevant to car audio. Although its meant in home environment, I thought the terms used and the phenomenon described, can be translated into our car environment. Enjoy reading. This article is abstracted from Galen Carol Audio. http://www.gcaudio.com/resources/howtos/systemnoise.html

System Noise

I

Edited December 26, 2006 by Lukie102

[Lukie102]

DVD Audio

 

Have you listened to any album in DVD audio format? Or have you ever wondered why DVD Audio sounded nicer? Here is an article explaining in layman terms why DVD Audio should be the way to go for audiophiles and alikes. Abstracted from http://timefordvd.com/tutorial/DVD-AudioTutorial.shtml

 

You may have heard of DVD-Audio (sometimes referred to as just "DVD-A") while shopping for a new DVD player. So what's the deal with DVD-Audio? What is so good about the DVD-Audio format? How is it better than the popular audio CD format? Is it something worth looking into? What do you need in order to enjoy DVD-Audio to its fullest potential? For answers these questions and more, keep reading...

It's a DVD![/url]

 

The first the thing you need to know is that DVD-Audio is part of DVD specification (you probably guessed that already). The DVD specification dictates the physical aspects and data capacities of the DVD format, as described in our DVD Tutorial. Be sure to read that first, if you haven't done so already.

 

As you have guessed, DVD-Audio is the audio application format of the general DVD specification. DVD-Audio includes:

• Pulse Code Modulation (PCM)

• High Resolution Stereo (2-Channel) Audio

• Multi-Channel Audio

• Lossless Data Compression

• Extra Materials which include still images, lyrics, etc.

 

We will go into more detail about each of these DVD-Audio unique features.

 

Pulse Code Modulation (PCM)[/url]

 

There are a number of ways to represent an analog audio signal as a digital signal. Think of the analog audio signal as a continuously variable voltage that fluctuates in frequency and amplitude to represent the frequency of the sound and the loudness of the sound, respectively. By far the most common method of digitizing an analog signal (i.e., representing the analog audio signal as a stream of digital 1's and 0's) is Pulse Code Modulation (PCM). PCM is the same digital technology as used by the audio CD format. PCM works by sampling an analog signal at regular intervals and encoding the amplitude value of the analog signal in a digital word using. The analog signal is then represented by a stream of digital words.

 

Without trying to get too technical, you might be interested in knowing that digital sampling theory (the Nyquist criterion, to be exact) says that in order to reproduce an analog signal with a certain frequency, you must sample at least twice as fast as that frequency. Given that we humans can hear sounds with frequencies from 20 Hz to 20,000 Hz, we need to sample at least at 40,000 Hz (or 40,000 times per second) in order reproduce frequencies up to 20 kHz. That's why the CD format has a sampling frequency of 44.1 kHz (or 44,100 times per second), which is slightly better than twice the highest frequency that we can hear. Note, we emphasized the phrase "at least". While sampling frequencies twice that what we can hear is a minimum requirement, it can be argued (mathematically even) that twice is not fast enough to accurately capture the characteristic of these high frequency signals. That's why in PCM, higher sampling frequencies offer better accuracy in reproducing high frequency audio information. The CD format's 44.1 kHz sampling frequency is barely adequate for reproducing the higher frequencies in the range of human hearing.

 

The other part of PCM is word length. Each sample, or snapshot, of the analog signal is characterized by its magnitude. The magnitude is represented by the voltage value in the analog signal and is represented in the digital signal as a data word. A data word is a series of bits. A bit is binary digit that has a value of "1" or "0". The longer the data word, the wider the range of analog voltages that can be represented as well as finer gradations of values in that range. In other words, the longer the word length, the wider the dynamic range (i.e., the difference between the softest sounds and the loudest sounds) and the finer the nuances of sounds can be recorded. The CD format has a word length of 16 bits, which is enough to reproduce sounds with about 96 dB (decibels) in dynamic range.

Is the 44.1 kHz sampling rate and 16-bit word of the audio CD format adequate? While audiophiles and discerning audio enthusiasts would definitely say "no", we would guess that most "everyday consumers" would initially think "yes". Those who are into high fidelity music reproduction say that the audio CD sounds cold and exhibits occasional "ringing" effects in the upper most frequencies when compared to high quality analog recordings on the LP and studio master analog tapes. Many of these enthusiasts are right, and some of their claims can proven mathematically and empirically. That is why the consumer electronics manufacturers have designed the DVD-Audio format. We would hazard to guess that once the "everyday consumer" hears the new DVD-Audio format on a properly calibrated, good quality audio system, they would be able to readily hear the differences and the improvement over the CD format.

High Resolution Stereo (2-Channel) Audio[/url]

One of the novelties of DVD-Audio is that it offers much higher resolution Pulse Code Modulation (PCM). DVD-Audio supports sampling rates of up to 192 kHz (i.e., the audio signal is sampled 192,000 times per second, or more than quadruple (4 times) the sampling rate of audio CD) and up to 24-bit word length. As we explained above, the higher sampling rate means more accurate and realistic reproduction of the higher frequencies. Is the 192 kHz sampling rate enough? At over nine times the highest frequency of human hearing, you would think so. But only careful listening tests of a high quality and well-balanced system will tell.

Though DVD-Audio supports up to 192 kHz sampling, not all audio program material has to be recorded using the highest rate. Other sampling rates are supported: 44.1 kHz, 48 kHz, 88.2 kHz, and 96 kHz. For two-channel stereo mode, high resolution audio usually means a sampling rate of 192 kHz.

 

The word length is increased up to 24 bits, meaning that a theoretical dynamic range of 144 dB can be achieved. We say theoretical because it is not currently possible to achieve such high dynamic ranges yet, even in the best of high-end components. The limiting factor is the noise level inherent in the electronics, due to thermal noise and other factors. The best signal-to-noise ratio that can be achieved in today's state-of-the-art components is about 120 dB. So it does seem that the 24-bit word length should be more than enough for the foreseeable future. Thought DVD-Audio can actually support word lengths of 16-bit and 20-bit, high resolution stereo usually means a word length of 24-bit.

 

Multi-Channel Audio[/url]

Another novelty of the DVD-Audio format is its capability for multi-channel discrete audio reproduction. That is up to six, full-range, independent audio channels can be recorded. Once you hear your favorite artists and albums recorded in the multi-channel format, you too may be convinced that it's wonderful experience and may realize that there's some missing in stereo recording when you switch back. There is real strong compelling reasons for multi-channel music. Among them, it allows us to hear not just the music, but experience the performance as though it's live in our own living room. No stereo music program, no matter how wonderfully recorded, can approach this feeling.

 

Usually, the sixth channel is used as the low frequency effects (LFE) channel, just like in a 5.1-channel home theater system, to drive the subwoofer. But the sixth channel is actually full frequency and can be used as a center surround channel (placed behind the listener, as in a home theater EX system) or as an overhead channel (placed well above the listener) for added height dimensionality to the soundstage. The application and placement of the six audio channels in a multi-channel DVD-Audio format is only limited by the imagination of the recording artist and the recording/mixing engineer.

 

Note that multi-channel DVD-Audio does not always mean 6 channels or 5.1 channels Sometimes it is just 4 channels (left front, right front, left surround, and right surround) or 3 channels (left front, center, right front). And in terms of sampling rates and data words, multi-channel DVD-Audio can use up to 192 kHz and up to 24-bit word length. But practically speaking, multi-channel DVD-Audio usually uses 96 kHz sampling, because of data capacity limitation of the DVD-Audio disc. Remember, 6-channel audio uses three times the data capacity of two-channel stereo when both use the same sampling rate and word length. Speaking of data capacity, DVD-Audio uses a form of data compression in order to fit the high resolution stereo and/or multi-channel digital information.

 

Lossless Data Compression[/url]

 

To store the massive quantity of digital audio information efficiently, the DVD Forum has approved the use of Meridian's proprietary lossless (i.e., no digital information is lost in the encoding and decoding process) encoding/decoding algorithm as part of the DVD-Audio format. It is appropriately named Meridian Lossless Packing (MLP) algorithm.

[Lukie102]

Damping Factor

Some one asked me recently a relevant question on damping factor, here is one article that explain it in a more layman manner. Abstracted from http://www.crownaudio.com/pdf/amps/damping_factor.pdf

[/left]

UNDERSTANDING DAMPING FACTOR

Loudspeakers have a mind of their own. You send them a signal and they add their own twist to it. They

keep on vibrating after the signal has stopped, due to inertia. That's called "ringing" or "time smearing."

In other words, the speaker produces sound waves that are not part of the original signal.

Suppose the incoming signal is a "tight" kick drum with a short attack and decay in its signal envelope.

When the kick-drum signal stops, the speaker continues to vibrate. The cone bounces back and forth in its

suspension. So that nice, snappy kick drum turns into a boomy throb.

Fortunately, a power amplifier can exert control over the loudspeaker and prevent ringing. Damping is the

ability of a power amplifier to control loudspeaker motion. It's measured in Damping Factor, which is load

impedance divided by amplifier output impedance. Let's explain.

If the speaker impedance is 8 ohms, and the amplifier output impedance is 0.01 ohms, the damping factor

is 800. That's a simplication. Since the speaker impedance and amplifier output impedance vary with fre-

quency, so does the damping factor. Also, the impedance of the speaker cable affects damping. Thick

cables (with low AWG) allow more damping than thin cables with (high AWG).

The lower the amplifier's output impedance, the higher the damping factor, and the tighter the sound is. A

damping factor of 1000 or greater is considered high. High damping factor equals tight bass.

How It Works

How does an amplifier control speaker motion? When the loudspeaker cone vibrates, it acts like a micro-

phone, generating a signal from its voice coil. This signal generated by the speaker is called back EMF

(back Electro Motive Force). It travels through the speaker cable back into the amplifier output, then

returns to the speaker. Since back EMF is in opposite polarity with the speaker's motion, back EMF

impedes or damps the speaker's ringing.

The smaller the amp's output impedance, the greater is the effect of back EMF on the speaker's motion.

An amplifier with low output impedance does not impede the back EMF, so the back EMF drives the loud-

speaker with a relatively strong signal that works against the speaker's motion. When the speaker cone

moves out, the back EMF pulls the speaker in, and vice versa.

In short, the loudspeaker damps itself through the amplifier output circuitry. The lower the impedance of

that output circuitry, the more the back EMF can control the speaker's ringing.

To prove it to yourself, take a woofer that is not connected to anything. Put your ear next to the cone and

tap on it. You might hear a low-pitched "bongggg" if the speaker itself is poorly damped. Now short the

speaker terminals and tap again. You should hear a tighter thump.

Damping factor varies with frequency. As you might suspect, damping factor is most important at low fre-

quencies, say 10 Hz to 400 Hz.

[Sarong1]

Lukie, thanks for the help...this thread will serve to be very beneficial to many people...

[Sarong1]

For the majority of us that are strictly sound quality fanatics, I find it important to be able to translate what we have heard and express ourselves in terms that are standardised and clearly understood.

 

In my attempt to share this sometimes difficult to master art of expression, I have abstracted some audiophile jargons or languages from the Hi-Fi Choice magazine. The next time round when you are invited to comment on each others setup, these terms will come useful.

 

Descriptive Terms

 

Aggressive Forward and bright sonic character.

 

Ambience The impression of an acoustical space, such as the performing hall in which a recording was made.

 

Analytical Highly detailed.

 

Articulate Intelligibility of voice(s) and instruments and the interactions between them.

 

Attack The leading edge of a note and the ability of a system to reproduce the attack transients in music.

 

Balance Essentially tonal balance, the degree to which one aspect of the sonic spectrum is emphasised above the rest. Also channel balance, the relative level of the left and right stereo channels.

 

Body Fullness of sound, with particular emphasis on upper bass. Opposite of thin.

 

Boxy The sound of a loudspeaker with audible cabinet resonances.

 

Bright A sound that emphasises the upper midrange/lower treble.

 

Dark A tonal balance that tilts downwards with increasing frequency. Opposite of bright.

 

Decay The fadeout of a note, it follows the attack.

 

Definition (or resolution) The ability of a component to reveal the subtle information that is fundamental to hi-fi sound.

 

Depth (of image) The perception of music being reproduced behind the loudspeakers and inhabiting a reproduction of the acoustic space of the original recording.

 

Detail The most delicate elements of the original sound and those which are the first to disappear with lesser equipment.

 

Dry A sound that is devoid of 'juice', which usually comes acrossas fine-grained and lean. Also a loss of reverberation as produced by a damped environment.

 

Dynamic The suggestion of energy and wide dynamic range. Related to perceived speed as well as contrasts in volume both large and small.

 

Euphonic An appealing form of distortion that generally enhances perceived fidelity, often ascribed to the harmonic elaborations of some valve amps.

 

Fast Good reproduction of rapid transcients which increases the sense of realism and 'snap'.

 

Focus A strong, precise sense of image projection.

 

Forward(ness) Similar to an aggressive sound, a sense of the image being projected in front of the speakers and of music being forced upon the listener.

 

Grainy A slightly raw, exposed sound which lacks finesse.

 

Grip A sense of control and sturdiness in the bass.

 

Hard Uncomfortable, forward, aggressive sound with a metallic tinge.

 

Harsh Grating, abrasive.

 

Imaging (stereo) The sense that a voice or instrument is in a particular place in the room.

 

Juicy Sound that has joie de vivre, energy and life.

 

Naturalness Realism.

 

Opaque Unclear, lacking transparency.

 

Open Sound which has height and 'air', relates to clean upper midrange and treble.

 

Presence A sense of an instrument or voice occupying a place in the listening room.

 

Speed A system with a good speed and transcient response can deliver the immediacy or 'snap' of live instruments.

 

Thick A lack of articulationand clarity in the bass.

 

Thin Bass light.

 

Timbre The tonal character of an instrument.

 

Transcient The leading edge of a percussive sound. Good transcient response makes the sound as a whole more live and realistic.

 

Transparency, transparent A hear-through quality that is aking to clarity and reveals all aspects of detail.

 

Tweak To tune a system or component in an attempt to get the best performance from it.

 

Tweaker Some one who enjoys this process.

 

Veiled Loss of detail due to limited transparency.

 

Warm A fullness in the lower midrange/upper bass.

 

Weight A sense of substance and underpinning produced by deep controlled bass.

 

[Lukie102]

small matter

[Sarong1]

CAR ACOUSTICS

 

Folks in ICE will always complain about bearing with different car acoustics and defeating cancellation as key requisite to achieving good music reproduction in car. No matter how much of sound proofing and dampening can be done on your car, there still remain many furniture in the car that cannot be totally sound proved or eliminated.

I came across this article from http://www.saecollege.de/reference_material/pages/Acoustics.htm

://http://www.saecollege.de/reference_...ics.htm

://http://www.saecollege.de/reference_...ics.htm

://http://www.saecollege.de/reference_...ics.htm

It summarises well what are the good elements in car acoustics which some people said are a better choice over many other internal listening environment.

[/url]

The main factors in a car are:

• The Shape: There are no parallel walls in a car and what walls there are are thin and curved.

• The Speakers. In a car, the speakers are almost always flush mounted. i.e. They are mounted into a flat surface like below the rear window or in the side door panels. As a result there are no out of phase signals coming from the rear of the speaker. Also the rear speakers are mounted in a big cabinet - the boot.

• High Frequencies: In the car the windows are the main high frequency reflectors but they are all at angles (approx. 6 - 12 degrees)and are usually curved as well. The highs also get diffused evenly throughout the cabin by the dash board. Also the ceiling, sides and floor are covered in high frequency absorption.

• Mid Frequencies: The seats, door panels and passengers are all low mid to high mid absorbers. Modern cars have deep pile carpet on thick underfelt which also acts as a mid frequency absorber. Most of the car's acoustic treatment for cutting down engine and road noise is also on the inside and acts as acoustic treatment for the car stereo.

• Low Frequencies: The beautiful thing about cars is the bottom end response. With a couple of hundred watts a side, a sub-woofer under the seat and the loudness switch on the bottom end thumps away and sounds great. Actually most of the low end goes straight through the walls and disappears, consequently it doesn't hum around the internal body causing phase problems. Any vibration is dampened by the foam lining and carpet and as far as the low end is concerned the car is equal to open air. Next time you play CD in your car get out and listen to what actually leaves the car (most of it!! especially if the windows are open)

What do you think? Is car acoustic really an ideal listening environment?

[Sarong1]

This is one good article on understanding just how drums are recorded in studio environment. Abstracted from http://www.saecollege.de/reference_material/

 

MICROPHONE PLACEMENT

Let's take a look at a standard drum kit.

 

Looking from the top this is the standard layout of right handed drummer. What we have to do now is Mike it! So let's start by putting two mikes over the top (generally called the "Overheads"). But where? We have to start thinking in terms of a stereo image right from the start.

The stereo Image is the picture of sound created between the two speakers. When a signal is placed equally in each speaker the sound appears to come from a phantom centre speaker, this is what we call the stereo centre All balancing is based around this point - e.g. the bass, kick and snare are normally panned in the centre whilst the hihats, toms and cymbals are spread across the speakers from left to right. The easiest way to experience the stereo spread is to listen through headphones where the effect is more obvious.

 

The standard panning setup in drum recordings is:

• Kick - Centre
• Snare Centre
• Hats - Half right/right
• Cymbals - left - right
• Toms - left - centre - right.

But if you look at a kit it isn't really setup like it should sound. The snare is to the right, the toms have no spread etc. In fact if you were to put up two overheads left and right the stereo image would have the kick centre and the snare to the right and the toms going from centre to left.

Sound Picture Created

Yet on recordings if you imagine looking at the kit from the front it looks (sounds) like the snare is centre and the toms spread from left to right. (You will notice that I refer to the imaging as a picture - well that's what it is!) So can we get the overheads to paint a picture like this? Have a look at this setup which is based around drawing an imaginary line through the kit which lines up the kick and snare etc.

The microphone placement places a stereo image similar to where you want to go as far as the placement of the different components.

The mic placement will look like this"

You can now start to hear a stereo image of the kit as you will want to hear it in the mix. If you were to put the mikes together in a stereo pair but aimed each side of the dividing line you would get a stereo image but a narrow one. The width increases as you move the two mikes away from each other. The placement in the drawings above are about normal with enough spread to make the kit have some width. With practice and careful placement of these two mikes you should be able to get a good balance of the kit. If you added a kick drum you would have a real, open sound of the kit.

The next step is to mike the individual components so that their position and individual sound can be emphasised.

KICK DRUM

There are three ways of setting up a kick drum

• Front and rear skins on.
• Front skin with rear skin with hole in it
• Front skin only.

These three set-ups create three differing sounds. First you must tune the kick as per the directions in the tuning drums page.

The first setup with both skins is a thick, solid, round sound with a decay as the drum decays. I believe the best way to mike up this setup is to use two mikes. One over the pedal and one at the other end like this.

This setup allows you to balance the attack sound of the beater with the decay of the front skin. This miking setup also brings up and important factor in recording:

Microphone Phase Relationships

So which mike should you phase reverse??. If you look at the microphone over the beater it is pointing downwards like all the other microphones on the kit will do whereas the microphone on the front skin of the kick drum is facing the opposite way. Therefore the front skin mike should have the phase reversal. As you can see it is a good idea to reverse the phase of your kick mike even when you are not using two of them as the normal kick mike setup places the kick mike out of phase to the rest of the kit mikes.

Similarly, when we get into miking toms and snares top and bottom the bottom mike will require a phase reversal.

The next setup is where the kick drum has a front skin on with a hole in it. Because of the hole you can access the front skin - thus the attack sound - without having to use a beater mike.

Here the mike is placed inside the drum pointing to where the beater hits so as to get the full impact of the beater. Note that the mike is still out of phase to all the downward facing mikes on the kit so a phase reversal is preferred. The mike is also placed off centre within the shell.

Another factor effecting the kick sound is the beater the drummer uses. Beaters vary from soft to hard. Hard beaters (usually wood) have more impact sound than the softer beaters. Experiment with each and you will hear the difference. How close to the centre of the skin the beater is placed also varies the sound. Similarly the size of the drum sticks the drummer uses will also effect the sound - thin sticks aren't going to go boof! no matter how much you EQ them.

Sound Pressure Level:

It should be noted here that the SPL (Sound Pressure Level) created by drums is extreme so you must select a microphone that can handle high SPL and even then it will output a high voltage into the console. Therefore a Microphone PAD should be inserted in the console to prevent the front end of the microphone preamplifier distorting. If your console doesn't have a mike pad switch you should insert one in the microphone lead. Like the phase reversal plug you can purchase mike pad plugs from your local dealer. A pad of anywhere from 10db - 20db will be required.

A note here about mike pads.

When purchasing a console check that the microphone pad is in the proper section of the circuit. Some manufacturers put the pad after the transformer and before the preamp. Unless the transformer is an extremely high quality one it will distort so make sure the pad is before it in the circuit.

Here's a circuit diagram for all you techo freaks who understand circuit diagrams. Nowadays all the front ends are electronically balanced circuits that are capable of handling the high voltages that modern mikes produce but if you are into retro equipment its worth checking your old console out because a lot were made with the pad in the wrong place.

TOMS

The toms are similar to the full kick drum miking in that there is a mike on the impact skin that gets the full attack of the stick when it hits the drum plus you can also add another optional bottom mike to get the hang of the the drum. You must again remember the phase relationships here. If you wish to add a bottom mike to the toms you must reverse it's phase.

RACK TOMS FLOOR TOMS

If your drummer doesn't have a bottom skin on the toms you can use either a top mike or both mikes or you can opt for just one under mike with a phase reversal naturally. The advantage here is that the under mike is inside the tom which isolates the mike from the other drum sounds and improves separation.

CYMBALS AND HIHATS

The Cymbals

These are basically covered by the overheads but you might find that the ride cymbal needs a mike of it's own if the drummer rides it a lot through the chorus. Basically you want the crash cymbals to have a loose sound yet the ride often is the main drive as it replaces the hihat for the 8 a 16 feels. You must consider this factor when setting up the overheads. Drummers also accent using the bell of the ride cymbal that can be extremely loud so beware of miking too close to the bell of the ride cymbal or it will dominate the sound field. Some engineers mike the ride from underneath. In a complex drum setup with lots of splash and crash cymbals you might like to spot mike certain cymbals but I reckon that if you've setup your overheads correctly they should cover the full cymbal range.

The Hihats

Like the overheads the hihat also requires a mike with a clean top end so it's usually a condensor mike. I like to hide the hihat mike from the snare by placing it in a position that is pointed at where the drummer impacts it with his stick but the hihat is physically between the hihat mike and the snare.

Good separation between the hihat and the snare is desirable so consider the snare when you place the the hihat mike. Another factor of the hihat is the sound made when they are snapped together. I like to aim the mike so it is pointing at a point that gets the stick impact as well as the pointing at the edge of the hats as that is where the closing sound emanates. N.B. If you get too close you will get wind distortion from the hats as they close.

One of the problems you can get is where the drummer has the hihat low to the snare and the toms also low to the snare. This creates separation problems as well as making it hard to isolate the snare from the tom. There's not much you can do other than ask the drummer to change. This is not as awesome as it sounds, some drummers have never considered this aspect of their kit layout and on making the change actually say it's OK and find they easily got used to it and now prefer it. The same problem can occur with the ride cymbal - some drummers have their ride cymbal almost touching the floor tom which makes separation hard - I recently had a drummer like that and when I mentioned it he agreed to change. After the session he remarked that he actually liked the change and would do it in future. Moral of this story? - don't be afraid to ask!!

SNARE DRUM

Once again, the snare can be miked from the top and the bottom, in fact it is one most often double miked. The bottom mike on a snare can give the snare more depth but it also gives you control over how much snare crack sound is in the overall sound. (The snare is actually the stretched wires across the bottom skin and gives the snare it's sound - otherwise it's just another tom). The snare mike is normally squeezed in between the hihat and the first rack tom and like the tom mikes is aimed at the main impact area in the centre of the snare.

Side Stick: Often you have a drummer playing a lot of side stick. I have used a separate mike specifically for the side stick. The side stick action is for the drum stick to hit the rim of the snare drum and the main impact is on the right side of the snare. As your normal snare mike is placed on the left side it doesn't always pick up the side stick clearly. Not only does this give you a mike closer to the side stick action it also allows for different EQ and effects for the side stick sound. You can either track it to a different recording track or you can watch the drummer and switch mikes during record.

AMBIENCE MIKES

Drums miked close-up don't actually sound very real as their real sound is a combination of various factors. You actually have to get away from them to get the full sound. A close mike on a snare doesn't really sound like a snare drum (thus the importance of the overheads) so some engineers add Ambience mikes to allow the freedom to add the distance sound of the kit when mixing. Naturally the drums must be in their own room for this system to be used or the ambience mikes will pick up everyone else in the room. Basically ambience mikes are a stereo pair of mikes placed at a distance (room size limited) from the kit. They can be setup as a crossed pair or moved apart to gain a more ambient spread. You might like to try using a MS Stereo mike setup. Ambience mikes can also be Gated - so they only open when the snare is hit for example- and you must have plenty of recording tracks to allow for another stereo pair. Ambience mikes effect all the kit and push the drum kit back in the sound field so if you want a round tight kick sound and an ambient snare sound you have to gate them so they are closed for the kick and open for the snare. An engineer I know used to hang a very directional shotgun mike high above the kit aimed at the snare and use the under snare mike to trigger a gate that opened it whenever the snare was hit. He would then mix it in with the snare sound and it gave the snare a natural ambience and was extremely effective.

So now we have set up all the mikes we are ready to start balancing and equalising them

Microphones for drums

• The Kick. What are we looking for in a kick drum mike? Firstly and most importantly it must be capable of withstanding high sound pressure levels!! When a mike is only inches away from a kick drum beater the sound pressure levels are extremely high at low frequencies. The kick drum mike must be capable of handling the extreme transients involved. Secondly it must be capable of reproducing very low frequencies.
  The two most popular kick mikes are - The AKG D12 and the Beyer M88. The M88 is my favourite. Both these mikes have an extended bottom end response and can handle the high sound pressure levels associated with kick drums. On the other hand if the drummer is not hitting too hard you can't beat the Neuman U87 or 49, which are high quality condenser and have large diaphragms (good for low frequencies) and smooth low end response. Other mikes are the Shure SM57/58 and the old RE20 which are both capable of withstanding the load.
• The Snare. Here we are looking for a mike that will withstand extreme high end transients and has a tight pattern so as to keep out the high hats and the adjacent toms. The most common snare mikes would have to be the Shure SM57 and the Sennheiser MD421, followed by the Neuman U87/89 and the AKG 414EB. Others are the Sennheiser MD441 or the Neuman KM84. I'm always amazed at how many engineers still use the Shure SM57 even though there are lots of other mikes around. The main advantage of the SM57 is that it's a tight mike with a tight pattern that keeps out the spill from the hi/hat and the toms. They are also extremely reliable and don't mind being hit by a wayward drummer. I should note here that the difference between the SM57 and the SM 58 is that the SM58 has a permanent wind shield - the microphone section is identical. You can buy a wind shield for the SM57 (Note the two microphones next time you see a press conference from the White House.)
• The Toms. The two main mikes used for toms are the Sennheiser 421 and the Shure SM57. In the studio I like to use Neuman U87's as they have a beautiful warm bottom end. The Shure SM 57's don't have a lot of bottoms but if you're tight miked the proximity effect compensates for it and as with the snare their tight pattern helps.
• Overheads. Good condensor mikes make the best overheads. There are three main overhead mikes, the Neuman U87 for warmth, the AKG 414EB and the AKG 451 for crystal clarity. The AKG C1000 and the Roden are also a good budget condensor overhead mike except I find that both have a slightly tinny top end compared with the more expensive models. I would say the AKG 451 with a CK1 capsule and 10db pad is the most popular overhead mike.
• Hihats. Condensor mikes with a tight pattern make the best Hihat mikes like the AKG 451 or the Neuman KM84. Both have a 10db pad option which is handy as the high end transients from a hihat are extreme.
• Ambience Microphones. Usually high quality condensor mikes are used here.

Edited January 1, 2007 by Sarong1

[Sarong1]

RECORDING PIANOS & ORGANS

The Grand Piano
The piano is really just a guitar (or more accurately a harp) lying on its side. It has strings stretched between two bridges, a striking area where it is hit with a soft hammer, and a sound board below.

The drawing above shows the main areas of concern when recording a piano. The mikes can be placed in any of the position A - D as well as underneath. The sound holes give you access to the sound board below the strings. So lets look at each position.

Position A

Position A is the most typical mike position used in studio music recording. It's a stereo pair that is about 150cm(6") apart, placed over the hammers with one pointing to the lower strings and the other directed toward the high strings. They should be about 150cm(6") above the strings. They are placed just behind the music stand. If there is no music involved the music stand can be removed giving a cleaner access to the strings. If these two mikes are placed correctly you can achieve a really good stereo image where the low strings appear from the left and the notes follow to the high notes on the left. Because the mikes are over the hammers the notes are bright and have a nice attack.

Position B

Position B utilises the hardness of the bridge and can be used to emphasise the low strings. I often add a small amount of it to the left of the image to accentuate the bass strings. Great if you have a 7 or 9 foot grand!!

Position D

Position D is the traditional Classical way of recording a grand piano and is still used today when recording grand pianos with an orchestra. It can also be used to add body and warm to a position A setup.

Position C

Position C is a position that accesses the sound board. The level coming off the sound board is quite high so it is a good position if you are caught having to record a grand piano in a studio with other instruments and you want separation from the other instruments. Two mikes places in the sound hole s allow you to lower the piano lid to the lower stand and with a couple of blankets or sleeping bags thrown over the lot you will get good separation yet a clean sound that will sound even better with a bit of high shelving added. Another way of accessing the sound board is to place a mike under the piano pointing straight up. This is often used in TV where they don't want the mikes to show.

PZM Mikes

There is one more way of recording a grand and that is to use 2 x PZM mikes fixed to the lid and then the lid closed. This produces a beautiful clean sound and is also great if you have spill problems, hence there use on stage shows. Give me two good Neumans or AKGs and I'll go with them anyday though. The AKG 451 is my favourite.

Abstracted from http://www.saecollege.de/reference_material/

[Boring]

Here's a breakdown of the IASCA cd and what are the thing to look out for in each track taken from talkaudio:

 

This is for the iasca official sound quality referance cd which was used in part of the 2005 events and will be used exclusively in the 2006 events

 

Track 1 Don Dorsey "Ascent"

Not officially used as a judging track but is a very good track to listen to as regards to hearing if a system is capable of the full range of frequencies. There is some extreme low frequency content in this track which is perfect for trying out subwoofers. The voice and introduction at the beginning of the track is that of Mary Ellen Papadeas whom is the iasca presidents wife

 

Track 2 Left and right channel verification (OFFICIAL JUDGING TRACK)

Without doubt one of the most important tracks on the disc, please take time to listen to the WHOLE of this track not just the first few seconds. Just because you get "left channel left channel" correct doesn't necessarily mean you're going to get the "right channel right channel" bit correct also.

 

Track 3 Phase verification track

This track is not offically used as a judging track but the judge may sometimes play this track. If he hears something which is out of the ordinary he or she may also put comments on your score sheet. This is an excellent set up track if used correctly for setting up mid and high frequencies so as voices appear as natural as possible. Simply listen to the track and listen to the voice both in and out of phase. There should be a difference in your perception as to where the voice is originating from and how tight the image focus is. Simply keep changing the phase (positive and negative) on your speakers until you get the best focus you can get.

 

Track 4 Instructions for the tonal accuracy and spectral balance part of the judging section

 

Track 5 John Williams Star Sars "Throne room and title ending" (OFFICIAL JUDGING TRACK)

The reproduction should leave you with a felling of being in a massive music hall with an equally massive orchestra. Around nine or ten seconds into the track there should be quite a bit of low frequency information which should be felt as well as heard. The notes should be distinct and should both start and stop not run into one another. Around 29-30 seconds into the track there should be a french horn at the back of the stage on the right hand side these should echo around and bounce of the edges of the soundstage. At around the same time there are some high frequencies played from the xylophone, they should be clean and crisp but not overpower any of the other instruments.

 

Track 6 John Williams Superman "the planet krypton" (OFFICIAL JUDGING TRACK)

This track is an excellant test of the ability of the system to play extreme low frequency information. The judges will mainly be listening for around 1.09 seconds into the track where there are a succession of eight or nine very low notes all sustained for around two seconds. They should descend in order, the fourth note in succession is the 18hz note when this is played correctly the music should be felt as well as heard, if done correctly you should feel a small vibration through the car and the sub may appear to flutter. All this should be done without the sub ever losing control of the notes and producing just one long continuous note. Right at the end of the track there is a crescendo from the tymphany drums the strings and the organ these should increase in volume but not get so loud so as the subwoofer is the dominant speaker heard.

 

Track 7 Jaques Loussier "Gavotte in d minor" (OFFICIAL JUDGING TRACK)

This track is primarilly used for testing midbass and midrange frequencies. Right at the beginning of the track the double bass is very prominant, the notes should be clear and concise not drone on from one to another. At around 40 seconds the double bass plays a sustained low frequency note which should last a while, if it doesnt improvements can be made in this area. One of the other instruments being scrtinised is the kick drum which again should be quick and have weight behind it. Without being overdominant the kickdrum has two sounds, one being the hammer hitting the skin then the hammer coming off the skin. The judge may occasionally comment on these. At around 23 seconds the piano begins to play it should sound larger than life and not compressed in the slightest. It should sound as if coming from the front of the vehicle not the passenger side footwell. At 45 seconds the piano plays a low sustained note which should last a for a few seconds, if it doesnt again improvements can be made

 

Track 8 instructions for the staging evaluation

in the next three tracks the judges will be looking for information regarding THE LISTENING POSITION ,STAGE HEIGHT , STAGE WIDTH, STAGE DEPTH AND AMBIANCE

 

Track 9 Henry Mancini "Theme from the pink panther" (OFFICIAL JUDGING TRACK)

Theres a lot going on in this track but at the beginning the piano at far left is an excellent reference as to the stage width. On that side of the car with the triangle on the right hand side but slightly in a bit, so bear this in mind. The saxophone is also quite prominant in the recording, it should sound a lot deeper and further into the mix and should be positioned around left of centre to centre. The brass and drums will be even deeper into the stage at around the 42 seconds mark. The orchestra finally gets moving and from here on in you should be able to determine the size of the room and hear the reflections off the walls it should sound large and lush.

 

Track 10 Lincoln Mayorga "Camarillo" (OFFICIAL JUDGING TRACK)

This track is excellent for judging stage height and also stage width. SET UP TIP ...right at the begginning of this track there is a bass playing at left of centre right at the front of the stage pay particular attention to this. As every car sounds different

simply follow the bass line and see where you think it originates from and where it moves. 23 seconds into the track the bass play a particular low note to see if you have things set correctly it should not move if however it moves forwards and backwards then you have a slight imbalance of the midbass speakers and subwoofer. If it jumps from one note to another from left to right in the car then you have a speaker imbalance from left to right . In the car a well set up system the bass will not move and will appear to all come from the front of the car with the subwoofer undetectable. The tambourine at the beginning of the track should be right on the stage boundary on the right hand side a little deeper than the bass. The saxophone should be central with the drums slightly deeper in the mix. Right at the front of the stage is the piano and it should sound large and bold as its the closest instrument.

Both guitars are the outermost instruments on the track and again can be used to determine the outermost points of the stage

 

Track 11 Carl Orff "Fortune empress of the world " (OFFICIAL JUDGING TRACK)

At the begginning of the track only a certain section of the choir are singing and if you listen carefully a piano is playing very gently right at the front of the stage. At 41 seconds into the recording (first timpani strike) the sound should change and the whole of the choir are in full voice and the sheer size of the room should be very evident. If you cannot get the impression of a huge cavernous hall with a symphony orchestra and full choir then there are a few improvements which can be made to your system.

 

This track is excellent for judging stage depth, pay particular attention to the timpani strikes

 

Track 12 staging evaluation track technical track "left centre and right narratives"

(NOT USED IN COMPETITION)

Can be useful for set up purposes or checking tonality on voices

 

Track 13 instructions for imaging track evaluations

Different judge will use differant references throughout the evaluations the end result being the same the two main parts to imaging are "relative position" and "focus"

 

Track 14 imaging evaluation #1 "Blues stay away from me" Harry James band

(OFFICIAL JUDGING TRACK)

In this track generally the two instruments which are used for reference are the trumpet at slightly inside far left and the baritone sax at far right. The trumpet is slightly inside far left and this must be taken into consideration. The judges will listen for these two instruments and then wait to see if anything catches them out. TIP.. generally the things which makes imaging either move or briefly change in size is down to intensity and tonality changes between the left and right channel so conentrating on these things in set up can greatly improve this portion of evaluation

 

Track 15 imaging evaluation #2 "I will find you there" Micheal Ruff (OFFICIAL JUDGING TRACK)

Theres lots going on in this track but the main things to listen out for are the three vocals which are at centre the male vocal which is far right and the female vocal which is far left. The female vocal is located further back in the stage than the centre image. There are also two instruments to concentrate on the guitar which plays infrequenctly at left of centre and the trombone at right of centre.

Again concentrate on relative position and focus of each individual instrument or voice

 

Track 16 imaging evaluation #3 "Too close " Clair Marlo (OFFICAL JUDGING TRACK)

The main thing to concentrate on with this track is the female centre vocal which should be exactly in the centre and not move when either low or high notes are sung. Other reference which can be used on this track are the accordian at left of centre and the guitars at just inside far left and far right

 

Track 17 imaging evaluation #4 seven drum beats

Generally this track is not used by iasca uk judges but several other countries in europe do use this track for imaging evaluation so I'll run thorugh it anyhows

Tip.. when using this track listen to the tonality first before the positioning. Listen to the track a couple of times whilst going across does the tonality change at all ??? If so this points to an inbalance between the left and right speakers and more work will have to be done to get this better. After this has been done then listen to the relative positions

of the strikes. If the strikes are even and equally placed fine. If one side seems to be more dominant than the other then there is a balance problem within the system. If however you seem to struggle achieving a centre image then this can be crossover point induced or caused by the overall relative level of midrange.

 

Track 18 explantion of linearity evaluation (tracks 19-23)

Track 19 level setting instructions for track 20

Track 20 level setting track for linearity evaluation

 

To set this correctly the volume should be set so as to be equal to the spoken voice of the judge at a normal level. Once the volume has been set it will not be changed between the tracks until the evaluation is finished. The judges are looking for differnces between the three levels.

The low level linearity is generally used to test if the system still has the correct amount of sub-bass in the system on low.

The double bass and the kickdrum are difficult instruments to get right at low level even though this is not a bass heavy track at all some systems will be concieved as having a lack of subbass.

The mid-range should always stay open and clear without ever becoming harsh or overbearing especially at high level.

 

Track 24 noise evaluation instructions

 

Track 25 noise evaluation "A la valse" (OFFICIAL JUDGING TRACK)

This is a very clear and clean recording which starts loud and then towards the end of the track gets quiter. Good systems should still have music playing at 1.03 seconds into the track without being overcome by gain hiss or a noise gate being induced.

Edited January 2, 2007 by Boring

[Bobcatsysop]

What is Sound Quality and how is it judged?

This session of the Clinic will deal with sound quality in mobile audio systems. We will cover the gamut of SQ, starting with definitions, philosophy, how it is judged in competition, and then how to get proper SQ in a car taking all these into consideration.

We will then move to types of sound system set-ups, including different front speaker arrays, both with conventional drivers and horn-loaded compression drivers, and move into the design and building of such systems. While I cannot guarantee coverage of all possible aspects of SQ in one article, we will likely split this topic into three or four parts, and I will be as thorough as I can without getting carried away in hard-to-understand technical jargon. Bear with me guys; this is a VERY complicated topic.

So what is Sound Quality? SQ is that aspect of a sound system, which encompasses all of the performance factors which give the system the ability to re-produce an accurate and life-like rendition of the original recording as perfectly as possible. It includes factors such as tonality, ambience, subtle nuance, system gain structure, dynamics, transient response, and the list goes on and on. SQ is a combination of all these technical factors as well as proper speaker placement and proper system design. When a system is said to have perfect SQ, it generates the most accurate sound possible, with a sense of musical realism that gives the listener the impression that they "hear" a live performance right in front of them, as if the listener was in the audience watching the actual performers on an actual stage. Now, to get this realistic musical "sound stage" in a car, it takes careful system planning, speaker placement, and tuning.

This holds true in competition organizations as well. Sound systems are judged according to how well they re-create the "live" performance that was originally recorded. Judges are encouraged to attend as many live concerts as they can to learn how they should sound in such areas as tonality, dynamic impact, listening room ambience, and each musician's placement on the stage they are playing from. During SQ judge's training, a small "reference" home audio system is set up to train the judges on how the music was recorded and how it is supposed to sound in a properly-setup vehicle. The competition organizations produce their own CDs, which contain a variety of musical selections chosen to test the limits of sound systems in a wide range of aspects. These selections include orchestral works, a variety of instrument soloists, vocalists (both male and female), opera, choir, percussionists, as well as dynamic tracks. A good SQ system will reproduce ANY type of music as realistic as possible without any sense of distortion. The judges use these CDs in each car at a competition and score each system on how well it reproduces the recordings based on what the music "should" sound like live. The judges are intently familiar with the material on the CDs they use and look for problem areas during the car's evaluation.

So, in laymen's terms, a system with proper SQ in one where you can sit in the seat, close your eyes, and *feel* as if you are sitting in the audience at a live performance. Seems simple enough, right? We'll see.

Tonality:

Commonly referred to as tonal accuracy and spectral balance, tonality is that quality of a system that gives the musical instruments their natural sound. If a saxophone is played, for example, it sounds exactly like a real saxophone, and you can tell it is not a trombone, French horn, tuba, or any other brass instrument. Likewise, any instrument has it's own characteristic sound, and a system with good tonality will allow the listener to differentiate the instruments being played. According to the Official IASCA (International Auto Sound Challenge Association) rulebook of competition, Tonal Accuracy and Spectral Balance is a combination of six characteristics---loudness, pitch, timbre, modulation, duration, and attack and decay. I won't go in-depth on each of these, but will give a brief definition: loudness-the magnitude of the sound. pitch-the quality of a sound that determines it's position on a musical scale. timbre-harmonics that give a sound it's sonic signature. modulation-changes in amplitude, phase, or frequency that occur in a sound. duration-length of time a sound is heard. attack and decay-the time it takes a sound to build-up (attack) and die-down (decay). Here is a quote from the IASCA rulebook summing the Tonal Accuracy section: "Superior systems will sound effortless and natural with any judging track. Weaker systems will exhibit distortion, unnatural coloration, dynamic compression, and frequency response errors. This leads to listening fatigue and lends an unnatural sound to the music."

Listening Position relative to the perceived sound stage:

Basically, in a concert, the musicians are on a stage in front of you, and likely they are well in front of you as you'll be sitting in the audience. Good SQ systems will give the illusion of this same perceived stage being well in front of you. Systems with poor "listening positions" will make you feel like the musicians are in your face, around you rather than in front of you, or even worse, behind you. Remember the goal is to simulate watching and listening to a live performance, and as such there are no musicians beside or behind you. The best systems will give you the impression of the stage being a considerable distance in front of you, as if you were sitting a few rows back in the audience.

Stage Width:

How wide is the sound stage? Car interiors are horrible for good sound reproduction, but we will cover this later on. However, a wide sound stage is an important factor. Bad systems will have almost no width, sounding as if only a center channel speaker is playing. Better systems have well defined left and right stage boundaries, but these boundaries will stay well within the interior of the vehicle.

The best systems will have stage boundaries, which extend BEYOND the physical area of the vehicle, with noticeable sounds that seem to emanate from a location outside the car (like from the side mirrors). The key to getting good width is to do so WITHOUT affecting center imaging, creating a "sonic hole" in the middle of the stage. We shall cover this later.

Stage Height:

Simply, how high is the sound stage? Additionally, is the stage height stable, meaning even height from right to left? When we sit in an audience, we IDEALLY see the musicians slightly above us. Thus, when we try to reproduce the "live performance" feeling in a car, the stage should seem to be in front of us, at eye level OR slightly higher. Too low, and it doesn't seem real, as our focus is skewed downward and we end up looking at the floor to envision the performance. Too high, and we feel like we are "star-gazing" and again, end up focusing on the sun visors or sunroof, again, it is unnatural. Getting the stage height to be stable means you get the picture of, for example, a guitarist at far left, a drummer in the center, and a harmonica far right. The key to a "stable" stage is to get these locations to project at exactly the same height. Often times, a car with an unstable height will portray the center image nicely, but the left and right images will be very noticeably lower, giving us the "rainbow" effect, and thus a mediocre stage height score. Likewise, some cars have frequency-dependant stages, where the high frequencies from the tweeters might be at eye level, the midrange frequencies seem slightly lower, and the midbass and sub bass appear to be coming from the floor. This is also unacceptable, and many factors affect this phenomenon. Some of them are speaker location, resonation of speaker enclosures, coloration caused by standing waves inside the enclosures, and poor equalization and sonic balance. We shall cover each of these in depth shortly. So, we want to see a "stable" sound stage projected at eye level in a proper SQ set-up.

Stage Depth:

Depth and Position to Soundstage ore often mistaken for one another. As discussed, Position to Sound stage pertains to how far in front of the listener the sound stage actually is. Stage Depth pertains to the placement of musicians On the sound stage either in front of or behind one another. Often times, for instance, the drum set will be located "behind" the guitarists, and the vocalist will be in front of these instruments, front and center. Systems with no stage depth will portray a flat sounding, one-dimensional stage where every musician appears to be side by side. Systems with excellent stage depth will give the listener a sense of space between the performers, and you should be able to tell that instruments are being played at different distances from you.

Ambience:

Ambience is that phenomenon that gives you a sense of space around an instrument. Many people confuse ambience with artificial "surround-sound" echoes meant to add the illusion of music all around you. Every recording contains ambience of some sort, be it a sense of "air" around the instrument or the "sound" of the room the track was recorded in. Using reference recordings where the actual recording room characteristics are known ahead of time, SQ judges can score ambience based on whether the performance "seems" to be played from inside this actual room. For example, there is a track on the '97 IASCA SQ competition CD that was recorded by a single microphone at a level of approximately 20' above the stage, INSIDE a large auditorium. With proper ambience, it should seem like you are in that large auditorium when you listen to this track. THAT is proper ambience, not artificial ambience. Tuning and speaker quality have the most impact on ambience, followed closely by controlling sonic reflections and resonations in the vehicle and speaker placement with proper x/o selection. Some people use ambiently tuned rear fill speakers to try to accomplish this aspect, but in a properly designed and tuned system, rear fill is NOT necessary to acquire excellent ambience.

Imaging:

The sound stage should be looked at in 5 separate sections: Left, left center, center, right center, and right. These are the most common locations on the stage used to evaluate the imaging characteristics of the car. By using reference material, the judges know exactly where the different instruments should be "located" on the stage. A system that images well will have well-defined and focused instruments located exactly where they should be on the sound stage. These images will not wander from their locations, and you could easily close your eyes and point to where each instrument actually is. Many systems exhibit frequency-dependant imaging where as the frequency changes in pitch or scale, the image will move accordingly. This is not good imaging. Likewise, some systems have good left and right imaging, but fail to get a center-stage focus at all. To many, the center image is the most important location on the stage, and it is easy to tell which cars have good centers and which don't. Speaker placement, path length differences, and proper equalization vastly affect imaging, and to a lesser degree, resonations, reflections, standing waves, and uneven interior surfaces play a role in the imaging characteristics of a system. Again, we will get to these aspects shortly.

Sound Linearity:

How well balanced does the system sound at low, moderate, and high volume? A system with good linearity will sound equally balanced at all three loudness levels, remaining accurate tonally and free of distortion of any kind. This depends mainly on proper gain settings and equalizer tuning, though other factors can possibly affect it.

Absence of noise:

A good SQ system will be a symbience of many factors coming together to provide a performance free of unwanted noises such as speaker pops, alternator whine, ground loop noise, additional noise floor in the form of extra hiss, on/off thumps or pops, and any other form of unwanted noises. All recordings exhibit a noise floor in the background, It is a byproduct of the recording process that cannot be overlooked and for this reason and only the noise level present in the original recording will be acceptable. Again, several factors affect system noise, and an "avoiding noise" section will deal with them all.

Dynamic Impact:

A superior system should be capable of reproducing the proper "feel" of the music. We need to consider the fact that music has Two dimensions---That which we hear, and that which we feel. If we were to go to a rock concert, sitting front and center, and the drummer decides to go-off on an improve solo while the rest of the band grabs a cold one and some ho-hoes, we are treated to a barrage of dynamic percussion sounds. When facing the stage, we feel the sound waves both in our chest and abdomen as well as on our skin. The bass drum obviously will be the most prominent; however the toms, snare, and even hard cymbal strikes can be felt, and felt easily if the guy is REALLY going off. We can even plug our ears to make this effect more pronounced. Sound is emitted in waves. These waves possess energy levels that are dependant on amplitude (loudness) and the proximity of the listener to the source, thus, we can feel the sound waves as they interact with the touch-receptors in our bodies. Ever flinch or blink upon hearing an abrupt, loud sound? Well, I am not certain of precise figures, but I'll bet it is about 50% due to what we hear, and 50% due to a subconscious physical reaction to "feeling" the sound wave, causing sudden stimulus in our nerve endings. The best car systems are capable of re-creating this sense of dynamics, as it should be "felt" live. And they should do so without feeling percussive waves emanating from rear mounted subs (Yet another topic we will discuss), making us hear the bass player up front, but he feels like he is behind us at the concession stand getting a sausage dog or something. NOT good.

This ends part one. We should be on track as far as the definition of Sound Quality and the factors we need to consider to achieve a truly amazing sound stage in a car. Of course, this is only the beginning (a Preface, if you will), and I will be going into understandable but great detail of every aspect of achieving our goal. So, hang in there folks. The fun stuff is right around the corner.

Article extracted from this teamaudionutz website

[Bobcatsysop]

2nd article extracted from this teamaudionutz website:

SQ system set-ups and the theories behind them

This article cannot possibly cover all the products out there that have the ability to afford great sound, so we shall jump into the "proper installation" category. First up is a discussion of a typical sound quality system and the theories behind its design. I can field any Q's you have as far as system component quality in a one-on-one basis in the Q&A section. Typically, a system will contain a sub-woofer set-up with a mid/tweet combo and a possible extra set of tweeters somewhere. More complex systems will go one step farther by adding either dedicated midbass drivers or front-mounted subwoofers. In either case, the number of subs varies, but the midbass and midrange drivers are restricted to one pair. This is done due to the phenomenon known as "multiple point-sourcing", where having more than one pair of drivers playing the critical imaging-cue frequencies can cause double-imaging, phase errors, and loss of image focus because the sound emitted seems to emanate from more than a single location. There are instances where multiple midbass and midrange driver

[Coolbreeze90]

Hi all,

[/size]

Happen to stumble across this website that provides a good glossary of audio terms used. Want to share with everyone this knowledge.

 

Hope it is helpful

 

- A -[/url]

Absorption [/url]- Short for the term Acoustical Absorption (quality of a surface or substance to take in, not reflect, a sound wave).

AC [/url]- An abbreviation of the term Alternating Current (electric current which flows back and forth in a circuit; all studio signals running through audio lines are AC).

Acoustic[/url] / Acoustical[/url] - Having to do with sound that can be heard by the ears.

Acoustic Amplifier[/url] - The portion of the instrument which makes the vibrating source move more air or move air more efficiently; this makes the sound of the instrument louder. Examples of acoustic amplifiers include: 1) The body of an acoustic guitar, 2) The sounding board of a piano, 3) The bell of a horn and 4) The shell of a drum.

Acoustic Echo Chamber[/url] - A room designed with very hard, non-parallel surfaces and equipped with a speaker and microphone; dry signals from the console are fed to the speaker and the microphone will have a reverberation of these signals that can be mixed in with the dry signals at the console.

Acoustical Absorption[/url] - The quality of a surface or substance to take in the sound wave and not reflect it or pass it through, or an instance of this.

A/D [/url]- An abbreviation of Analog to Digital Conversion (the conversion of a quantity that has continuous changes into numbers that approximate those changes), or Analog to Digital Converter.

ADAT [/url]- A trademark of Alesis Corporation designating its modular digital multitrack recording system released in early 1993.

ADSR[/url] - The letters A, D, S &R are the first letters of: Attack, Decay, Sustain and Release. These are the various elements of volume changes in the sounding of a keyboard instrument.

AES[/url] - An abbreviation of Audio Engineering Society.

AES/EBU Professional Interface[/url] - A standard for sending and receiving digital audio adopted by the Audio Engineering Society and the European Broadcast Union.

Aliasing[/url] - A sampler mis-recognizing a signal sent to it that is at a frequency higher than the Nyquist Frequency. Upon playback, the system will provide a signal at an incorrect frequency (called an alias frequency). Aliasing is a kind of distortion.

Alternating Current[/url] - Electric current which flows back and forth in a circuit.

Ambience [/url]- The portion of the sound that comes from the surrounding environment rather than directly from the sound source.

Ambient Field [/url]- A term with the same meaning as the term Reverberant Field (the area away from the sound source where the reverberation is louder than the direct sound).

Ambient Micing [/url]- Placing a microphone in the reverberant field (where the reverberation is louder than the direct sound) so as to do a separate recording of the ambience or to allow the recording engineer to change the mix of direct to reverberant sound in recording.

Amp [/url]- 1) An abbreviation of the term Amplifier (A device which increases the level of an electrical signal. 2) An abbreviation of Ampere (the unit of current). 3) An abbreviation of amplitude (the height of a waveform above or below the zero line).

Ampere[/url] - The unit of current, abbreviated Amp.

Amplification[/url] - An increasing of signal strength.

Amplifier[/url] - A device which increases the amplitude (level) of an electrical signal (making it louder).

Amplitude [/url]- The height of a waveform above or below the zero line.

Analog (Analogue)[/url] - Representative, continuous changes that relate to another quantity that has a continuous change.

Analog Recording[/url] - A recording of the continuous changes of an audio waveform.

Analog To Digital Converter[/url] - The device which does the conversion of a quantity that has continuous changes (usually of voltage) into numbers that approximate those changes.

Assign[/url] - To choose to which place an output is going to be sent.

Assistant Engineer [/url]- A less elevated version of the term Second Engineer. Experienced seconds often place microphones, operate tape machines, break down equipment at the session end and keep the paperwork for the session.

Atom[/url] - The smallest particle which makes up a specific substance. It's composed of a center around which electrons revolve.

Attack[/url] - The rate the sound begins and increases in volume.

Attenuation[/url] - A making smaller: reduction of electrical or acoustic signal strength.

Audio[/url] - Most often referring to electrical signals resulting from the sound pressure wave being converted into electrical energy.

Automatic Gain Control[/url] (Automatic Volume Control) - A compressor with a very long release time used to keep the volume of the audio very constant.

Automation[/url] - In consoles, a feature that lets the engineer program control changes (such as fader level) so that upon playback of the multitrack recording these changes happen automatically.

Aux Send [/url]- Short for the term Auxiliary Send (a control to adjust the level of the signal sent from the console input channel to the auxiliary equipment through the aux buss.

Auxiliary Equipment[/url] - Effects devices separate from but working with the recording console.

Axis [/url]- A line around which a device operates. Example: In a microphone, this would be an imaginary line coming out from the front of the microphone in the direction of motion of the diaphragm.

Edited February 6, 2007 by Coolbreeze90

[Bobcatsysop]

http://www.recordingeq.com/glossary/glosae.htm Audio Recording Terms Glossary Index A-E

http://www.recordingeq.com/glossary/glosfj.htm Audio Recording Terms Glossary Index F-J

http://www.recordingeq.com/glossary/glosko.htm Audio Recording Terms Glossary Index K-O

http://www.recordingeq.com/glossary/glospt.htm Audio Recording Terms Glossary Index P-T

http://www.recordingeq.com/GlosPubUZ.htm Audio Recording Terms Glossary Index U-Z

[Coolbreeze90]

Yeah thats the one

[Sarong1]

Good stuff but make sure its well digested and not wasted...

[Boring]

Take from another web-site

 

Many people say, "Well I want to use this speaker, but I'm put off by the fact that it's 8 ohms."

 

Let me explain why it's ok to use an 8 ohm speaker, and why it could actually be better than a 4 or 2 ohm speaker.

 

First of all, using a higher impedance than what your amp is nominally rated for is always ok. In fact, if your amp is rated for 4 and 2 ohm impedances, typically you will get less power into 8 ohms meaning that your amp will run cooler and more efficiently at higher impedances. It won't hurt your amp, and in fact it's actually much better for your amp's longevity.

 

Now, you're probably saying how is getting less power out of my amp a good thing? Think of your amp's power reserves as your bank account. Just because you have 100 dollars in your bank account doesn't mean you have to spend it all. Quite the opposite. Wouldn't you rather spend LESS and get MORE? That's exactly what you're doing when you use less power from your amp, and get the same amount of output from a high efficiency speaker.

 

Remember, most speakers are rated at 2.83V. A 4 ohm speaker rated for 90db spl at 2.83v is really being rated at 2 watts! Whereas an 8 ohm speaker rated for 90db spl at 2.83v is only being rated at 1 watt. You can do the math for yourself, Power = Voltage^2 / resistance. At 2 watts, we can assume that same 8 ohm speaker is actually rated at 93db spl (remember, every doubling of power gives you a theoretical 3db gain in spl). So using that 8 ohm speaker will give you the same amount of output, at half the power required as a 4 ohm speaker. Your amps run cooler and draw less power from your vehicle's charging system, your speakers run cooler, and everyone is happy!

 

Ok, now let's look at another example of an 8 ohm versus 4 ohm voice coil. Typically, the efficiency of a speaker is given by:

 

Efficiency = ( B^2 * L^2 ) / ( R * Sd^2 * Mms^2 )

 

B = magnetic field strength

L = length of wire

R = resistance

Sd = surface area

Mms = mass

 

So for your 8 ohm voice coil, using the same wire as a 4 ohm voice coil, you would need twice the L or length to get an 8 ohm impedance. That makes sense doesn't it? A longer wire will have more resistance. Now, looking at the formula above, doubling L actually causes your efficiency to rise, even though the impedance also rises. So in this very oversimplified example, raising the impedance actually causes efficiency to go up and lowering the impedance actually causes a loss of efficiency.

 

What's important to remember is that it's the overall output and efficiency of the speaker that's important, not the impedance. A high impedance, high efficiency driver can get just as loud off a small amount of power as a low impedance, low efficiency driver that sucks a ton of power! Just because you have a 100 watt amp doesn't mean you have to use all 100 watts.... it's all about being efficient.

 

I also found this excellent post by Dan Wiggins over at carstereos.org:

 

"I think one thing to consider is that going to a higher impedance voice coil will result in better packing of the voice coil, meaning a higher cross-sectional-area of copper in the flux, for a given mass.

 

If you take a given driver, and simply swap out voice coils, you end up with more efficiency as you increase the impedance. Take a voice coil, say 2" diameter, 1" winding length, 24AWG 4 layer, and swap it with a 2" diameter, 1" winding length, 27AWG 4 layer, and you double the impedance, but the efficiency also goes up - less mass and better packing density."

 

It's because the moving mass has dropped, and if desired - because of the thinner wire diameter which packs in tighter - you can put more layers in the voice coil and potentially raise the BL."

[Sarong1]

Er..Hmm..(clearing my throat)...you are trying to explain why people like your tweeter aren't you???