Difference between revisions of "Terminology & Concepts"

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== Backline Amplifiers ==
 
== Backline Amplifiers ==
{{Instruments/Backline Amplifiers}}
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{{:Instruments/Backline Amplifiers}}
  
 
== Comb Filtering ==
 
== Comb Filtering ==
 
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{{:Comb Filtering}}
Comb filtering occurs when two identical (or nearly identical) signals, one delayed in time relative to the other, are added. Depending on the delay time, the resulting summed signal can sound hollow or “boingy”, and is usually considered an undesirable sound.. Comb filtering occurs most commonly when signals are combined electronically, such as in a hard disc based recording system, but can also occur acoustically, such as a talker located slightly off axis of two identical microphones spaces inches apart.
 
 
 
(Thanks to Ken-at-Bose for this information).
 
 
 
See [[{{PAGENAMEE}}#Nulls|Nulls]] and [[{{PAGENAMEE}}#Phase_.28cancellation.2C_interference.29|Phase]] below.
 
  
 
== Crossover ==
 
== Crossover ==
See [[Crossover]]
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{{:Crossover}}
  
 
== DI (box) ==
 
== DI (box) ==

Revision as of 04:13, 23 August 2006

Terminology & Concepts

Backline Amplifiers

Backline Amps

The amplifiers that musicians typically have on stage (behind them) to amplify the sound of their instruments. Examples: Guitar amps, Bass amps, Keyboard amps.

Some musicians (Electric Guitarists especially) treat their backline Amplifiers as part of the instrument. For these folks there is just no substitute for that sound. Micing the speaker cabinet and amplifying it through the Bose Personalized Amplification System™ is a completely viable way to let them create the tone and share it with the room.


Comb Filtering

Comb filtering occurs when two identical (or nearly identical) signals, one delayed in time relative to the other, are added. Depending on the delay time, the resulting summed signal can sound hollow or “boingy”, and is usually considered an undesirable sound.. Comb filtering occurs most commonly when signals are combined electronically, such as in a hard disc based recording system, but can also occur acoustically, such as a talker located slightly off axis of two identical microphones spaces inches apart.

(Thanks to Ken-at-Bose for this information).



Crossover

The following is based on information that was available when the L1 Classic was in production. This information is equally applicable to the L1 Model I. The L1 Model II crossover is fixed at 200 Hz and does not vary depending on whether or not B1 Bass Modules are attached to the Power Stand.

Crossover —Definition

I'm familiar with the term 'crossover,' but not really with its meaning.

Audio crossovers are a class of electronic filters designed specifically for use in audio applications, especially hi-fi. A commonly used dynamic loudspeaker driver is incapable of covering the entire audio spectrum all by itself. Thus, crossovers serve the purpose of splitting the audio signal into separate frequency bands which can be handled by individual loudspeaker drivers optimized for those bands. A combination of multiple drivers each catering to a different frequency band constitutes most hi-fi speaker systems. An audio crossover may also be constructed mechanically and is commonly found in full-range speakers. -- more at Wikipedia


If you play Guitar then you should be able to relate to the frequencies I will mention to describe the crossover idea. You can use the picture of the Keyboard to help if that works better for you. (Lowest notes are at the top). (click the keyboard to see that image in its original context).

Your bottom E string has a fundamental frequency of about 82 Hz. (Cyles per second). That is just a reference for this discussion.

When there is nothing attached to Amp 3 output (where we normally connect the blue B1 cable) the Powerstand does this:

L1 Classic and L1 Model I

  • Frequencies above 110 Hz are sent to the L1 Cylindrical Radiator™ This is less of a "crossover" and more of a cutoff just because there's no point sending frequencies to the L1 that it can't reproduce.
  • It doesn't mean that if the L1 cutoff is set to 110 Hz, you won't hear anything from the low E string. Our perception of tones is based not only on the fundamental (in the case of the low E at 82 Hz it is lower than 110 Hz), but it is also based on the harmonics we will hear in multiples of the fundamentals (2 x 82, 3 x 82, 4 x 82).

BUT

Add the B1 (with all four conductors working) and the PS1 does this:

  • Frequencies above 180 Hz are sent to the L1 (the crossover is moved up).
  • Frequencies from 40-180 Hz are sent to the B1 (and some processing (EQ) is applied to the 40-180 Hz range) to optimize things with the design of the B1.

For reference, 40 Hz gets us into the range of the low E string on an Electric Bass (an octive below our low E on an Acoustic Guitar).

Here's a bit more from Hilmar-at-Bose about the really low notes:

Bass Frequencies

24db / ocatve

The crossover in the power stand is a pretty steep one, 24dB/octave. So an octave above crossover, the B1's are 24dB down and below crossover the L1's are 24dB down. They get out of each other's way pretty quickly. — Cliff-at-Bose [1]

More Bass Talk

Hilmar-at-Bose explained in More Bass Talk

Crossover

If there is no B1 and nothing connected to the Bass Line Out. The L1 sees frequencies from 110Hz up. Feeding it anything lower, doesn’t make sense, since it couldn’t produce any acoustic output and if would rip the drivers to shreds.

In any other case the L1 sees signals only from 180Hz up. There is no other variation in frequency or gain for the L1 no matter what else happens

Bass Line Out and B1 behavior

This is based on the design goal that “You should always sound the same; no matter how much Bass stuff is attached” I can try to explain my view of why this is a good design goal (of which you may disagree) but let’s look at the actual behavior first.

Without Bass Line out

1xB1: 40Hz-180Hz, B1 specific EQ, some nominal gain that we call 0dB 2xB1: 40Hz-180Hz, B1 specific EQ, -6dB as compared to nominal

With Bass Line Out

0xB1: 40-180 Hz, flat, roughly the same gain as 2 B1 1xB1: 40Hz-180Hz, B1 specific EQ, -6dB as compared to nominal 2xB1: 40Hz-180Hz, B1 specific EQ, -12dB as compared to nominal

What this complex behavior does is the following. No matter if you attach 1, 2, or 4 B1s, you will get pretty much the same balance between all combined B1s and the L1s. It’s a little off for 3, 5, 6, 7 & 8 B1s, but still reasonably close.

Frequency content of an acoustic guitar

Oldghm, you did some really interesting experiments there. However, you have to be really careful when using an RTA. You can feed these things a pure sine wave at 80 Hz and by turning it up make the 63 Hz and even the 40Hz LED light up. They will be lower than the 80 Hz LED, but still come on. That does NOT mean, that the sine wave contains any other frequency than 80 Hz (it certainly doesn’t). It only means that the RTA has a pretty limited frequency resolution. The 63 Hz LED will respond best to 63 Hz signal but it’s in no way “blind” to 80 Hz signal. Thus being said, the actual frequency content is not easy to determine. All sounds that have a pitch are certainly constraint to 80 Hz and up (in standard tuning) and there isn’t actually too much energy at the fundamental. However, the “non-pitched” sounds like a hard string attack or whacking the top with your hand can very well have lower frequencies. Unfortunately, I don’t have any hard data on that, but we will measure that at some point.

Equal loudness curves

Here is the bunch http://hyperphysics.phy-astr.gsu.edu/hbase/sound/eqloud.html These curves tell us two things: First, the same physical sound energy produces different perceived loudness depending on frequency. You can turn that around into “The physical sound energy required to produce the same perceived loudness varies with frequency”. Second, this frequency dependency is a function of overall level.

The first statement is not particularly bothersome. Your auditory system is well calibrated to that. A voice sounds normal because it sounds like what you are used to, not because it has “constant sound energy” or “constant perceived loudness” with frequency.

The second statement is much more trouble. It basically says that if you amplify an acoustic source (even if you do it perfectly), the perceived spectral balance will change. This is a well-known effect, and most of our home entertainment systems have actually and “automatic loudness compensation” that changes the system voicing with overall level. We actually contemplated adding this to the Personalized Amplification System™ but after some soul searching we thought it would be too intrusive on the musician. The main corrections are at very low levels, and in most practical live music settings, the effect is pretty minor. As a rule-of-thumb guideline, turn the bass up a notch as you turn the volume down.


Crossover with Line Out

B2 Bass ModuleThis information is applicable to the B2 Bass Module
B1 Bass ModuleThis information is applicable to the B1 Bass Module
A1 PackLite AmpThis information is applicable to the PackLite® A1 Power Amplifier
L1 Compact
S1 Pro  This information is applicable to the S1 Pro system

Can I add a Bose bass module like a B1 Bass Module or B2 Bass Module to an L1 Compact or S1 Pro System
The Line Out from the L1 Compact and S1 Pro System is full range. The B1 Bass Module and B2 Bass Module are designed to work with a signal from 40 Hz to 200 Hz, and preferably with an EQ curve set in the L1 Model 1S or L1 Model II Power Stand.

If you add a PackLite® power amplifier model A1 and B1 Bass Module or B2 Bass Module to an L1 Compact or S1 Pro System you will need a crossover between the Line Out and the PackLite® power amplifier model A1.

Check out these discussions in the community.

Compact with Packlite + B1

The Compact and the B2, via the A1 and the Rolls SX21




DI (box)

DIbox.gif

Use a DI when you want to connect two devices and you have any of these issues:

  • impedance mismatch
  • line level mismatch
  • differences in wiring or connectors (e.g. Balanced XLR to Unbalanced 1/4" Tip-Sleeve)
  • noise - especially "hum" (ground loop)

A DI unit or DI box is an electronic device designed for connecting a piece of equipment with an electronic audio output to a standard microphone or line level input. It performs both level and impedance matching to minimise both noise and distortion. DI is variously claimed to stand for direct input, direct injection or direct interface. DI units are extensively used with professional and semi-professional PA systems and in sound recording studios. -- Wikipedia

You will also see the term DI used to refer to devices used to modify the tone as well as other properties of a signal. This is often in the context of Acoustic Guitar and Electric Bass. In the picture above the first two are passive DIs used for solving problems. The others are all sold as DIs that also shape the sound.

Dual Mono

This is amplifying the same sound source through two separate loudspeakers.

Equal Loudness Curves

(also known as Fletcher Munson curves)

Equal Loudness Curves

"You will see lots of references to equal loudness curves or equal loudness contours- these are based on the work of Fletcher and Munson at Bell labs in the 30s, or perhaps refinements made more recently by Robinson and Dadson. These were made by asking people to judge when pure tones of two different frequencies were the same loudness. This is a very difficult judgment to make, and the curves are the average results from many subjects, so they should be considered general indicators rather than a prescription as to what a single individual might hear" - Numbers and Initials of Acoustics

See also:

http://www.sfu.ca/sonic-studio/handbook/Equal_Loudness_Contours.html

http://hyperphysics.phy-astr.gsu.edu/hbase/sound/eqloud.html

Fletcher Munson Curves

See Equal Loudness Curve above

Gain Before Feedback

Gain before feedback refers to the maximum sound pressure level that can be attained before the sound from a speaker enters the microphone and is amplified a second time, creating a loop that only builds on itself: feedback.

An often not very scientific measure of how loud a sound reinforcement system can be turned up before any open microphone(s) will feed back. The point at which feedback occurs is effected by numerous variables, including atmospheric conditions (temperature, humidity, etc.) so it's not something that anyone considers an objective measure of performance. Instead the phrase is used to state relative differences: "By adjusting the EQ I was able to get 'more' gain before feedback." -- Sweetwater

Inverse Square Law

Localization, Spaciousness and Reverberation

Nulls

Out of Phase (for Drum mics)

See Drums

Phase (cancellation, interference)

See Phase

Phase cancellation occurs when two signals of the same frequency are out of phase with each other resulting in either a boost or cut in the overall level of the combined signal. -- Phase at the Zen Audio Project

If you are suffering from some or all of these, you could be experiencing Phase Interference

  1. "Hot" and "cold" spots in the audience area
  2. Tonal coloration
  3. Poor speech intelligibility
  4. Lack of music clarity
  5. Poor gain-before-feedback
  6. Poor imaging

See: Practical Realities of Phase Interference


Presets

Processors

Proximity Effect