Terminology and Concepts

If you are a performer new to managing your own sound, some of the terminology you encounter here in the forum may be unfamiliar.

Here are some terms that come up fairly often when discussing the Bose Personalized Amplification System™ family of products.  I've added some quick explanations and references. Most of these are terms that sent me searching for definitions within the first few months of owning my System. I hope you find this helpful.

If you have suggestions, additions, or corrections please send me a private message

Original Post

Terminology and Concepts

If you are a performer new to managing your own sound, some of the terminology you encounter here in the forum may be unfamiliar.

Here are some terms that come up fairly often when discussing the Bose Personalized Amplification System™ family of products.  I've added some quick explanations and references. Most of these are terms that sent me searching for definitions within the first few months of owning my System. I hope you find this helpful.

If you have suggestions, additions, or corrections please send me a private message

Backline Amplifiers

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

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).

Please see: Nulls and Phase cancellation below.


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.

Stereo vs. Mono in a Live Setting
Dual mono is usually not a good idea ...
Can I use it as a PA? (Cliff-at-Bose talks about Dual Mono)


Equal Loudness Curves

Click image to see it in context
(also known as Fletcher Munson 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


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

See the web page where this image appears in context

Doubling the distance drops the intensity by about 6 dB and that 10 times the distance drops the intensity by 20 dB. Click the image to the right to see the page in which it originally appeared.

Localization, Spaciousness and Reverberation

Originally posted by Hilmar-at-Bose:

There may be some confusion about “localization” and “spaciousness”. Both are perceptual attributes, which have been intensively studied especially in the context of concert hall acoustics.

“Localization” refers to your ability to detect the direction of a specific sound source. Let’s say you have your eyes closed but you can immediately tell where the guitar is (as opposed to the bass or the violin).

“Spaciousness” refers to the perceived size of the sound source and how much you feel enveloped with sound. Most researchers believe that a “good” concert hall provides both: accurate and easy localization and a sufficient amount of spaciousness. The main contributing factors for spaciousness are “early lateral reflections”. These are reflections from the sidewalls. Reflections from the side help quite a bit with creating a sense of space and envelopment. Reflection from the floor and ceiling are less desirable since they don’t enhance spaciousness but can negatively impact clarity and spectral balance. There are very interesting physiological reasons of why lateral reflections are perceived so much differently than vertical ones. I could drone on for hours on this topic, but I’ll save that for later. Anyway, it’s not too much of a surprise that most renowned concert halls (e.g. Musikvereinssaal in Vienna, Concertgebow in Amsterdam, or Boston Symphony Hall) have a “shoebox” shape that is long, narrow and tall. This shape provides good lateral reflections pretty much everywhere in the audience. Ironically, it’s not very popular with architects since the sight lines are terrible. Visually oriented architects prefer the “shell” shape, which provides great viewing but (unfortunately) no lateral reflections whatsoever. But I’m digressing again… The L1 does (in my humble opinion) the best thing. It radiates very wide horizontally but very little much towards the ceiling. Thus it provides enough energy for lateral reflections but keeps interfering ceiling reflections at bay. Since it’s a single source, it provides very easy and accurate localization, but in most venues the image will also be pleasantly spacious. That’s at least my own experience. - read it in context -

Originally posted by Ken-at-Bose (later in the same discussion):

What we perceive as "reverberance" is caused physically almost entirely by the persistence in time of an acoustic event.

Spaciousness -- the feeling of envelopment and room size is different.

Localization -- where we perceive the sound to be coming from -- is different again. Localization is almost entirely determined by the direction of the first arrival of sound (almost always the direct sound from the speaker in this case) and thus is almost entirely immune from the effects of reflections. Spaciousness is heavily determined by lateral reflections, which are produced in relative abundance by the L1 because it's so wide in its pattern.

High reverberance is caused by lots of reflections in rooms with little absorption. The L1 is particularly good at NOT producing as much reverberance because it sends little sound to the upper walls and ceiling where many detrimental reflections originate and are perpetuated. - read it in context -



Dead spots or areas of a room where there is a significant drop in volume.

When your sound is hollow, diffuse, and thin, you may be experiencing nulls or comb filtering. This is usually the result of a single sound source be amplified from several loudspeakers. Phase Interference

Out of Phase (for Drum mics)

Originally posted by Hilmar-at-Bose: Here's the nerds view :) Wiring the the two mics out of phase creates essentially a "dipole". Everything that is in the middle (i.e. equal distance) between the two microphones will get equally but out-of-phase so it cancels when the two microphone signals are summed together. In essence it creates a "blind spot" for the microphones for whatever is right in the middle plane. For the drums, that's mainly the kick (as Larry pointed out) and also the drummer (when he/she is hemming and hawing, squeaking with the chair, yelling about or in general having a grand old time). Another nice trick is to place the L1 that gets the mic signal somewhere the middle plane of the microphones. This drastically reduces potential for feedback and unwanted regeneration. Sound sources that are significantly closer to any one of the microphones are not much affected by the whole procedure.

See the entire discussion in context: Wiring two SM 75s with a Y cord

Phase (cancellation, interference)

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


On the PS1 power stand, there are selectable presets for Channels 1 and 2. Bose has created optimized settings for specific microphones, instruments and applications. For more about how to work with them see the Presets section in the Unofficial Users' Guide.


These devices are often placed between an instrument and an amplification system to modify the tone and other aspects of the sound of the instrument. Some musicians are using these in place of backline amplifiers and running their processors directly into their Bose System. Typically these can be run into Channels 3 or 4 of the PS1 powerstand. They can also be run to Channels 1 or 2 with preset #00.

Proximity Effect

The bass response of all directional microphones is increased as the signal source - your voice - comes closer to the mic capsule. This is called Proximity Effect and it becomes apparent at a range of one foot and increases as the distance of mouth to mic decreases.

For speakers or singers with high or thin voices, proximity effect can boost the bass, filling out the sound.

- source Shure Notes






Q: What is a Pad?

A "pad" is short for an "Attenuation Pad".

This is a device used to lower the signal level between two other devices.

"attenuator or attenuator pad Electronics. A passive network that reduces the voltage (or power; see usage note under gain) level of a signal with negligible distortion, but with insertion loss. Often a purely resistive network, although any combination of inductors, resistors and capacitors are possible, a pad may also provide impedance matching."

We use them with our Bose Systems when running a pro level +4 dBu signal to the XLR inputs on Channels 1 or 2 on the Powerstand.

We want to do this because the input sensitivity for these inputs is set for microphones. To get a better match for levels between the source and our inputs, we can use a pad.

Another application (although relatively rare) is if you want to daisy-chain the Line-Out of one PS1 Powerstand to Channel 1 or 2 of another. In that case you probably want to have a -20 dB pad inline, between the Line-Out of the first System and the Channel 1 or 2 XLR input of the second.

You can find separate attenuation pads like this:


(click image above to see it in context)

This an example, and you will want to look into the details to determine if you need a pad, and the kinds of connections that are appropriate for your input device.

Rane Professional Audio Reference

All About Pads

edit: fixed links
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:

  • 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).


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:

Originally posted by Hilmar-at-Bose:

before I answer the technical stuff, please let me try to clarify a few things. Many people think that a 5-string bass need a speaker with a frequency response down to 30 Hz or you wouldn’t be able to hear the low B “properly”.

That’s really not the case. The sound of the bass does not only consist of the actual “fundamental” tone (i.e. 31 Hz in case of the low B) but also many multiples of that frequency (i.e. 62 Hz, 93 Hz, 123 Hz, etc.), which are called the “harmonics”. In fact the amount of energy in the fundamental is relatively small as compared to the rest, so you really don’t loose much by keeping the response low at 30 Hz.

Furthermore, very low frequencies (say below 50 Hz) are not perceptually associated with “bass” or “thump” or “punch” but are much more like an indistinct rumble. We actually have analyzed the frequency content of a lot of recorded music and there is little energy below 50Hz and virtually none below 40Hz. The only type of content that uses very low frequencies regularly is movies. But there it’s not used for musical purposes but as a sound effect, e.g. explosions, starting rockets, a T-Rex stomping along, etc.

In live sound situations very low frequencies tend to make the sound very muddy unless it’s carefully designed fixed installation (as it is in theaters), so most live sound amplification gear will not go particularly low. For example an Ampeg SVT cabinet is only specified down to 55 Hz and most people would claim it does a very adequate job on a 5 string bass.

In fact our B1 goes actually deeper than most other bass cabinets. But it does stop at 40 Hz and does so very abruptly. You will not be able to get any usable energy out of the B1 or the Bass Line Out at 30 Hz. The reason for this is mainly protection. Most bass box designs cannot handle anywhere near the fully rated power below their port tuning frequency and there is no point in trying. They can’t radiate any appreciable amount of sound energy and it’s very possible to damage the driver.

Anyway, if you have a chance, why don’t you just plug in your bass into a double B1 system and let your ears decide whether there is anything missing or not?

Hope that helps


See the original discussion.

edit: added definition from wikipedia
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