Definition of Damping Factor

Definition: Technically, damping factor is an amplifier specification related to the

amplifier's output impedance. It is a good measure of the amplifier's ability to control the
motion of the woofer cone. A higher damping factor is best and usually means tighter
bass response. Damping factor is simply a number, usually between 50 and 200, although
many amplifiers have much higher damping factor specifications.
Pronunciation: damp•ing fac•tor

What is the Relationship between

Loudness and Amplifier Power?
Decibels and Watts
Decibels (a measure of loudness) and watts (a measure of amplifier power) are common
terms used to describe stereos. They can be confusing, so here is a simple explanation of
what they mean and how they relate.
What is a Decibel?
A decibel is actually two words, ‘deci’ (meaning one-tenth) and ‘bel’ (named after
Alexander Graham Bell, a scientist and inventor of the telephone). A ‘bel’ is a unit of
sound and a deci-bel is one-tenth of a bel. The human ear is sensitive to a wide range of
sound levels from 0 decibels (complete silence) to 140 decibels (pain). For clarity, a quiet
conversation is about 60 dB, a jet engine at close range is about 120 dB. The human ear is
capable of hearing and recognizing an increase or decrease in sound level of about 1 dB.
Anything less than +/-1 dB is hard to hear. An increase of +10 dB (easy to hear) is
perceived as being approximately twice as loud by most people.
What is a Watt?
A watt is a unit of energy, like horsepower or joules. In audio, a watt is used to describe
the energy output of a receiver or amplifier used to power a loudspeaker. The relationship
between power output and speaker loudness or volume is not linear or straight (+10 watts
does not equal +10 dB). For example, if you compare the maximum volume of a 50-watt
amplifier with a 100-watt amplifier the difference is only 3 dB, barely greater than the
ability of the human ear to hear the difference. It would take an amplifier with 10 times
more power (500 watts!) to be perceived as being twice as loud (a +10 dB increase).
Keep this in mind when purchasing an amplifier or receiver. 2X the power output = +3
dB increase, 10X the power output = +10 dB increase, or twice as loud. So, what is the
benefit of more power output? Read my article about amplifier power and loudness.

Amplifier Output Power: How Much Power is Enough?

Amplifier output power is one of the most important considerations in choosing a stereo
receiver. Power is measured in watts per channel and the decision about how much power
you need should be based on your selection of loudspeakers, the size and acoustic
characteristics of your listening room, and how loud you like to listen.
It is always best to match the power requirements of the speakers with the output power
of the receiver. Some speakers require more or less power, expressed as loudspeaker
sensitivity (in decibels , dB), which is a measure of how much sound output is produced
with a specified amount of amplifier power. A speaker with lower sensitivity of 88dB-
93dB (also known as speaker efficiency) will require more amplifier power than a
speaker with a higher sensitivity (94dB to 100dB or more) to play at the same volume
level.
Power output and speaker volume is not a linear relationship.For example, a receiver with
100 watts per channel will not play twice as loud as a receiver with 50 watts per channel
using the same speakers – the difference in maximum loudness would be barely
discernable, only 3 decibels (dB). Rather, more amplifier power will allow the system to
more easily handle musical peaks without straining, which results in better sound clarity.
When comparing the power output of different amplifiers, it is important to know how
the power is measured. The most accurate measure of power is RMS (Root Mean Square,
a mathematical formula), as opposed to peak amplifier power, a less accurate
specification. Some manufacturers inflate specifications by measuring power at a single
frequency, say 1kHz, instead of the frequency entire range, 20Hz-20kHz. When
comparing receivers with different power outputs, always make sure they are measured
the same way

Primary Consideration – Power Output

Power output is one of the most important considerations in choosing a receiver. Power is
expressed in watts per channel and the decision about how much power you need should
be based on your selection of loudspeakers, the size and acoustic characteristics of your
listening room, and how loud you like to listen. It is always best to match the power
requirements of the speakers with the output power of the receiver. Some speakers
require more or less power, expressed as loudspeaker sensitivity (in decibels, dB), which
is a measure of how much sound output is produced with a specified amount of amplifier
power. Power output and speaker volume is not a linear relationship. For example, a
receiver with 100 watts per channel will not play twice as loud as a receiver with 50 watts
per channel using the same speakers – the difference in maximum loudness would be
barely discernable, only 3 decibels (dB). Rather, more amplifier power will allow the
system to handle musical peaks without straining.
Generally, a speaker with lower sensitivity (88dB-93dB) will require more amplifier
power than a speaker with a higher sensitivity (94dB to 100dB or more) to play at the
same volume level. Most receivers are rated with a minimum power output of 75 watts
per channel and higher. When comparing power output, it is important to know how the
power is measured. The most accurate measure of power is RMS (Root Mean Square, a
mathematical formula), as opposed to peak output power. Some manufacturers inflate
specifications by measuring power at a single frequency, say 1kHz, instead of the entire
range, 20Hz-20kHz. When comparing receiver power outputs, always make sure they are
measured the same way.
5.1, 6.1, 7.1 Channels – How Many Do You Need?
A/V receivers are distinguished from two-channel or stereo receivers by having five or
more amplifier channels to power speakers that can reproduce movie theater sound or
multi-channel music in your home. Most DVD discs and other multi-channel sources are
encoded in Dolby Digital 5.1 and/or DTS 5.1 channel sound for playback on home
theater systems. A basic system consists of 5.1 channels of sound. The five channels are
left and right, like a stereo system, a center channel for movie dialog or music vocals and
on-screen sound, and left and right surround channels, for special effects and surround
sound. An additional subwoofer channel (the .1 LFE channel, Low Frequency Effects)
adds very low bass for music sources and special effects on DVD movie sound tracks.
The composite of the five main channels plus a subwoofer channel produces a
“soundfield” that envelops the listener. A 5.1 channel system is capable of playing
programs encoded in Dolby Digital and DTS.
6.1 channel a/v receivers have an additional rear-center channel output, and are becoming
increasingly popular, even at lower prices. Some DVDs are encoded with 6.1 channel
Dolby Digital EX and DTS-ES, and can be played back on this type of system. If
properly installed, 6.1-channel sound can create a more enveloping surround sound
effect.
7.1 channel receivers have three front channels, two surround and two surround-back
channels, plus a subwoofer channel. The additional rear channels produce a soundfield
with more precise placement of surround effects. Some 7.1 channel receivers offer
THX™ soundfield enhancement, which is a system developed by Lucas Film™ designed
to present film sound or multi-channel music with the most authentic quality. THX
processing is offered as Select or the more advanced Ultra 2 format, which is optimized
for movies and multi-channel music. Many other manufacturers have proprietary
soundfield programs, called DSP, which also provide enhancements for music and movie
sound. Sony, for example, has Digital Cinema Sound™ and Yamaha has Cinema DSP™.
Many 7.1 channel receivers also permit the two surround-back channels to be reassigned
to a second zone for a multi-room system, leaving the main system with 5.1 channels.

5.1 vs 7.1 Channel Home Theater Receivers - Which is Right

For You?

One question I get asked often is which is better, a 5.1 or 7.1 channel home theater
receiver.
In turns out that both options have advantages and disadvantages, depending on what
source components you are using and what your personal preferences are.
5.1 Channel Home Theater Receiver Overview
5.1 channel home theater receivers have been the standard for two decades. They provide
a perfectly good listening experience, especially in small to average-sized rooms. A 5.1
channel system consists of:
1. A Center Channel to provide an anchor stage for dialog or music vocal.
2. Left and Right Front channels to provide the main soundtrack information, or for
stereo music reproduction.
3. Left and Right surround channels for side and front to rear motion effects from movie
soundtracks and ambient sounds from music recordings.
4. The Subwoofer channel, which provides the extreme low frequency effects, such as
explosions or bass response in music performances.
7.1 Channel Home Theater Receiver Overview
However, when trying to decide if a 5.1 or 7.1 channel home theater receiver is right for
you, there are several practical features of a 7.1 channel receiver that could be of benefit
that you may not have considered.
1. A 7.1 channel system incorporates all the elements of a 5.1 channel system, but instead
of combining both surround and rear channel effects into two channels, a 7.1 system
splits the surround and rear channel information into four channels. In other words, side
sound effects and ambience are directed to left and right surround channels, and the rear
sound effects and ambience are directed to two rear or back channels. In this set-up the
surround speakers are set to the side of the listening position and the rear or back
channels are placed behind the listener.
For a visual look at the difference between a 5.1 channel speaker layout and 7.1 channel
speakers layout, check out an excellent diagram provided by Dolby Labs.
The 7.1 channel listening environment can add more depth the surround sound
experience, provide more a specific, directed, and immersive soundfield, especially for
larger rooms.
2. Although most Blu-ray soundtracks are 5.1, there are an increasing amount of Blu-ray
soundtracks that contain 7.1 channel information - whether it be 7.1 channel
uncompressed PCM, Dolby TrueHD, or DTS-HD Master Audio. If you have a 7.1
channel receiver with audio input and processing capability via HDMI connections (not
pass-through only connections), you can take advantage of some, or all, of these audio
capabilities. Check the specifications, or user manual, for each 7.1 channel receiver you
may be considering for more specifics on its HDMI capabilities.
3. Also, even with playback of standard DVDs, if your DVD soundtrack only contains
Dolby Digital or DTS 5.1 or, in some cases, DTS-ES 6.1 or Dolby Surround EX 6.1
soundtracks, by using the Dolby Pro Logic IIx extension or other available 7.1 DSP
surround modes that may be available on your receiver, you can still extract a 7.1 channel
surround field from both 2 or 5.1 channel source material.
4. Other surround sound extensions that can utilize 7.1 channels are Dolby Pro Logic IIz
and Audyssey DSX. However, instead of adding two surround back speakers, Dolby Pro
Logic IIz and Audyssey DSX allow the addition of two front height speakers. This
provides additional speaker setup flexibility.
5. In addition, if you prefer 5.1 channels for your main room, most 7.1 channel receivers
have the ability to use the extra two channels to provide a 2-channel speaker operation in
a second location. What this means is that, in many cases, while you are listening and
watching your DVDs in 5.1 channel surround sound in your main room, someone else
could be listening to a CD (provided you have a separate CD player connected to your
receiver) in another room, without having a separate CD player and receiver in the other
room - just the speakers.
6. Another option that is becoming more common on 7.1 channel receivers is the use of
Bi-amping. How this works is that if you have front channel Speakers that have separate
speaker connections for the midrange/tweeters and the woofers (I am not referring to the
subwoofer, but the woofers in your front speakers), you can reassign the amplifiers
running the 6th and 7th channels to your front channels. Then enables you to retain a full
5.1 channel setup, but still add two additional channels of amplification to your front left
and right speakers.
Using the separate speakers connections for the 6th and 7th channel on your bi-ampible
speakers, you can double the power delivered to your front left and right channels. Your
front mid-range/tweeters end up running off of the main L/R channels and your front
speakers woofers running off your 6th and 7th channel Bi-amp connections.
The procedure for this type of setup is explained and illustrated in the user manuals for
many 7.1 channel receivers. However, as I mentioned earlier, although this is becoming a
more common feature, but is not included in all 7.1 channel receivers.
9.1 Channels and Beyond
As more sophisticated surround sound processing options become available, such as DTS
Neo:X, that can expand the number of channels that can be reproduced or extracted from
source content, manufacturers are upping the ante on the number of channels they can
cram into a home theater receiver chassis. When moving into the high-end home theater
receiver arena, there are an increasing number of receivers that now offer 9.1/9.2 and a
small number that even offer 11.1/11/2 channel configuration options.
However, just as with 7.1 channel receivers, it depends on what you want to accomplish
in your home theater setup. Both 9 and 11 channel receivers can be used to set up 9 or 11
speakers (plus one or two subwoofers) in your home theater room. This allows you to
take advantage of surround sound processing systems, such as DTS Neo:X. However, a 9
or 11 channel receiver can also provide flexibility in terms of assigning two of the
channels to Bi-Amp the front speakers, or using 2 or 4 channels to create 2nd and/or 3rd
Zone two channel systems that can still be powered and controlled by the main receiver.
This can still leave you with 5.1 or 7.1 channels to use in your main home theater room.
Keep in mind that the vast majority of DVD, Blu-ray, and any surround sound audio that
you will receive from source content is mixed for 5.1 channel playback, with a smaller
number of source content mixed for 6.1 or 7.1 channel playback. This means that a 5.1 or
7.1 channel receiver with Dolby/DTS decoding and processing can easily fill the bill (A
5.1 channel receiver can place a 6.1 or 7.1 channel source within a 5.1 channel
environment). When moving up to a 9.1 or 11.1 channel receiver, the receiver is actually
post-processing the original 5.1, 6.1, or 7.1 channel encoded soundtracks and placing
them in a 9 or 11 channel environment. The results can be quite impressive, depending on
the quality of the source material, but it does not mean that it is required that you make
this leap. After all, many don't have the room for all those extra speakers!
Final Take
To put it all into perspective, a good 5.1 channel receiver is a perfectly fine option,
especially for a small or average room in most apartments and homes.
However, once you get into the $500 range and up, there is an increasing emphasis by
manufacturers with 7.1 channel equipped receivers. Additionally, when you get into the
$1,300 an up price range you start seeing some 9.1 channel receivers. These receivers can
provide very flexible setup options as you expand your system's needs, or have a large
home theater room.
On the other hand, even if you don't need to use the full 7.1 (or 9.1) channel capability in
your home theater setup, these receivers can easily be used in a 5.1 channel-only system.
This frees up the remaining two or four channels on some receivers for Bi-amping use, or
to run one or more two-channel stereo 2nd Zone systems.
Amplifier Power and Speaker Efficiency
Amplifier power, measured in watts, can be a confusing subject and is commonly
misunderstood. A common misconception is that wattage has a direct correlation to
loudness or volume. Some believe that doubling the power output will result in a
maximum volume that is twice as loud. In fact, power has little to do with loudness.
Power output is relevant to two main issues:
Speaker efficiency
The ability of the amplifier to handle musical peaks
Speaker Efficiency
Speaker efficiency, also known as speaker sensitivity, is a measure of the speaker's
output, measured in decibels, with a specified amount of amplifier power. For example,
speaker efficiency is often measured with a microphone (connected to a sound level
meter) placed one meter from the speaker . One watt of power is delivered to the speaker
and the level meter measures the volume in decibels. The output level results in a
measure of efficiency. Speakers range in efficiency or sensitivity from about 85dB (very
inefficient) up to 105dB (very efficient). As a comparison, a speaker with 85 dB
efficiency rating will take twice the amplifier power to reach to same volume as a speaker
with 88 dB efficiency. Similarly, a speaker with a 88 dB efficiency rating will require ten
times more power than a speaker with a 98 dB efficiency rating to play at the same level.
If you're starting with a 100 watt/channel receiver, you would need 1000 watts (!) of
power output to double the perceived volume level.
Dynamic Range
Music is dynamic in nature. It is constantly changing in volume level and frequency. The
best way to understand music's dynamic nature is to listen to live acoustic (un-amplified)
music. An orchestra, for example, has a wide range of volume levels, from very quiet
passages, to loud crescendos and some in-between quiet and loud. The range in volume
level is known as dynamic range, the difference between the softest and loudest passages.
When the same music is reproduced through an audio system, the system should
reproduce the same range in loudness. When played back at an average volume level, the
soft and medium passages in the music would require minimal power. If the receiver had
100 watts of power per channel, the soft and medium passages would require roughly 10-
15 watts of power. However, the crescendos in the music would require more
significantly more power for short periods of time, perhaps as much as 80 watts. A
cymbal crash is another good example. Although it is a short term event, the cymbal
crash demands lots of power for a short period of time. The ability of the receiver to
deliver bursts of power for a short time is important for accurate sound reproduction.
Although the receiver may only use a small portion of its maximum output most of the
time, it must have the 'headroom' to deliver large amounts of power for short periods of
time.

Distortion (THD)

Definition: Distortion specs in audio are expressed by the term THD (Total Harmonic
Distortion), which is measured in percentage terms, where the percentage represents the
amount of distortion present in the audio signal at full output level.
A receiver, or amplifier, that has a .01% - .08 distortion level at full output level would be
excellent at all listening levels. However, a receiver or amplifier that has a 5% - 10%
distortion measurement at full output would be unlistenable, even at a moderate level.

Dynamic Headroom

Definition: Dynamic Headroom refers to the ability of a receiver or amplifier to output

power at a significantly higher level than normal for short periods to accommodate
musical peaks or extreme sound effects in films. This specification is important in home
theater, where extreme changes in volume occur during the course of a film.
Dynamic Headroom is measured in Decibels. If a receiver/amplifier has the ability to
double is power output for a brief period to accommodate the conditions described above,
it would have a Dynamic Headroom of 3db.

Continuous Power (RMS)

Definition: Continuous Power refers to the ability of a receiver or amplifier to output its
full power continuously. In other words, just because your receiver/amplifier may be
listed as being able to output 100WPC, doesn't mean it can do so for any significant
length of time. Always make sure that, when you check for Specifications, that the WPC
output is measured in RMS terms. This means that the listed power output is a sustained
output at a specific volume level.
Also Known As: RMS

Surround Sound - The Audio Side of

Home Theater
Part One - Beginnings: From Mono to Stereo to
Fantasound and Quad
Introduction
Ever since Stereophonic sound became popular in the 50's the race has been on to create
the ultimate home listening experience. Even as far back as the 1930's, experiments with
surround sound were conducted. In 1940, Walt Disney incorporated his innovative
Fantasound surround sound technology in order to totally immerse the audience in both
the visual and audio sensations of his animation achievement, Fantasia.
Although "Fantasound", and other early experiments in surround sound technology could
not really be duplicated in the home environment, that didn't limit the quest by recording
engineers for both music and film to develop processes that would eventually result in the
surround sound formats that are enjoyed in home theaters all around the world today.
Monophonic Sound
Monophonic sound is a single-channel, unidirectional type of sound reproduction. All
elements of the sound recording are directed using one amplifier and speaker
combination. No matter where you stand in a room you hear all the elements of the sound
equally (except for room acoustic variations). To the ear, all the elements of the sound,
voice, instruments, effects, etc... appear to originate from the same point in space. It is as
if everything is "funneled" to a single point. If you connect two speakers to a
Monophonic amplifier, the sound will appear to originate at a point equidistant between
the two speakers, creating a "phantom" channel.
Stereophonic Sound
Stereophonic Sound is a more open type of sound reproduction. Although not totally
realistic, stereophonic sound lets the listener experience the correct sound staging of the
performance.
The Stereophonic Process
The main aspect of Stereophonic sound is the division of sounds across two channels.
The recorded sounds are mixed in such a way that some elements are channeled to the
left part of the soundstage; others to the right.
One positive result of stereo sound is that listeners experience the correct soundstaging of
symphony orchestra recordings, where sounds from the various instruments more
naturally emanate from different parts of the stage. However, monophonic elements are
also included. By mixing the sound from a lead vocalist in a band, into both channels, the
vocalist appears to be singing from the "phantom" center channel, between the left and
right channels.
Limitations Of Stereo Sound
Stereophonic Sound was a breakthrough for consumers of the 50's and 60's, but does have
limitations. Some recordings resulted in a "ping-pong" effect in which the mixing
emphasized the difference in the left and right channels too much with not enough mixing
of elements in the "phantom" center channel. Also, even though the sound was more
realistic, the lack of ambience information, such as acoustics or other elements, left
Stereophonic sound with a "wall effect" in which everything hit you from front and
lacked the natural sound of back wall reflections or other acoustic elements.
Quadraphonic Sound
Two developments occurred in the late 60's and early 70's that attempted to address
limitatons of stereo. Four Channel Discrete and Quadraphonic Sound.
Problems With Four-Channel Discrete
The problem with Four Channel Discrete, in which four identical amplifiers (or two
stereo ones) were needed to reproduce sound, was that it was extremely expensive (these
were the days of Tubes and Transistors, not IC's and Chips).
Also, such sound reproduction was really only available on Broadcast (two FM stations
each broadcasting two channels of the program simultaneously; obviously you needed
two tuners to receive it all), and four channel Reel-to-Reel audio decks, which were also
expensive.
In addition, Vinyl LP's and Turntables could not handle playback of four channel discrete
recordings. Although several interesting musical performances were simulcast using this
technology (with a co-operating TV Station broadcasting the Video Portion), the whole
set-up was too cumbersome for the average consumer.
Quad - A More Realistic Surround Approach
Taking a more realistic and affordable approach to surround sound reproduction, than that
of Four Channel Discrete, the Quadraphonic format consisted of matrix encoding of four
channels of information within a two channel recording. The practical result is that
ambient or effects sounds could be imbedded in a two channel recording that could be
retrieved by a normal phono stylus and passed through to a receiver or amplifier with a
Quadraphonic decoder.
In essence, Quad was the forerunner of today's Dolby Surround (in fact, if you own any
old Quad equipment--they have the ability to decode most analog Dolby Surround
signals). Although Quad had the promise to bring affordable surround sound to the home
environment, the requirement to buy new amplifiers and receivers, additional speakers,
and ultimately lack of consensus amongst hardware and software makers on standards
and programming, Quad merely ran out of gas before it could truly arrive.

Surround Sound - Part Two: Dolby

Surround And Dolby ProLogic
Surround Sound Transitions from the Cinema to the
Home
The Emergence Of Dolby Surround
In the mid-70's Dolby Labs, with breakthrough film soundtracks such as Tommy, Star
Wars, and Close Encounters of the Third Kind, unveiled a new surround sound process
that was more easily adaptable for home use. Also, with the advent of the HiFi Stereo
VCR and Stereo TV Broadcasting in the 1980's, there was an additional avenue for which
to gain public acceptance of surround sound: Home Theater. Up to that point, listening to
the sound portion of a TV Broadcast or VCR tape was like listening to a tabletop AM
radio.
Dolby Surround Sound - Practical For The Home
With the ability encode the same surround information into a two channel signal that was
encoded in the original Movie or TV soundtrack, software and hardware manufacturers
had a new incentive to make affordable Surround sound components. Add-on Dolby
Surround processors became available for those that already owned Stereo-only
receivers. As the popularity of this experience reached into the more and more homes,
more affordable Dolby Surround sound receivers and amplifiers became available, finally
making Surround sound a permanent part of the Home Entertainment experience.
Dolby Surround Basics
The Dolby Surround process involves encoding four channels of information--Front Left,
Center, Front Right, and Rear Surround into a two channel signal. A decoding chip then
decodes the four channels and sends them to the appropriate destination, the Left, Right,
Rear, and Phantom Center (center channel is derived from the L/R front channels).
The result of Dolby Surround mixing is a more balanced listening environment in which
the main sounds derive from the left and right channels, the vocal or dialog emanates
from the center phantom channel, and the ambience or effects information comes in from
behind the listener.
In musical recordings encoded with this process the sound has a more natural feel, with
better acoustical cues. In movie soundtracks the sensation of sounds moving from front to
rear and left to right adds more realism to the viewing/listening experience by placing the
viewer in the action. Dolby Surround is easily useful in both musical and film sound
recording.
The Limitation Of Dolby Surround
Dolby Surround does have its limitations however, with the rear channel being basically
passive, it lacks precise directionality. Also, overall separation between channels is much
less than a typical Stereophonic recording.
Dolby Pro Logic
Dolby Pro Logic addresses the limitations of standard Dolby Surround by adding
firmware and hardware elements in the decoding chip that emphasize important
directional cues in a movie soundtrack. In other words, the decoding chip will add
emphasis to directional sounds by increasing the output of the directional sounds in their
respective channels.
This process, although not important in musical recordings, is very effective for film
soundtracks and adds more accuracy to effects such as explosions, planes flying
overhead, etc.. There is greater separation between channels. In addition, Dolby Pro
Logic extracts a dedicated Center Channel that more accurately centers the dialog (this
necessitates a center channel speaker for full effect) in a movie soundtrack.
The Limitation Of Dolby Pro-logic
Although Dolby Pro-Logic is an excellent refinement of Dolby Surround, its effects are
derived strictly in the reproduction process, and even though the rear surround channel
employs two speakers, they are still passing a monophonic signal, limiting rear-to-front
and side-to-front motion and sound placement cues.

Surround Sound - Part Three - Dolby

Digital and Dolby Digital EX
Dolby refines the home theater surround sound
experience
Dolby Digital
Dolby Digital is often referred to as a 5.1 channel system. However, it must be noted that
term "Dolby Digital" refers to the digital encoding of the audio signal, not how many
channels it has. In other words, Dolby Digital can be Monophonic, 2-channel, 4-channel,
5.1 channels, or 6.1 channels. However, in its most common applications, Dolby Digital
5.1 and 6.1 is often referred to as just Dolby Digital.
The Benefits Of Dolby Digital 5.1
Dolby Digital 5.1 adds both accuracy and flexibility by adding stereo rear surround
channels that enable sounds to emanate in more directions, as well as a dedicated
Subwoofer Channel to provide more emphasis for low frequency effects. The subwoofer
channel is where the .1 designation comes from. For more details, refer to my article:
What the .1 Means in Surround Sound.
Also, unlike Dolby Pro-logic which requires a rear channel of only minimal power and
limited frequency response, Dolby Digital encoding/decoding requires the same power
output and frequency range as the main channels.
Dolby Digital encoding began on Laserdiscs, and migrated to DVD and Satellite
programming, which has solidified this format in the marketplace. Since Dolby Digital
involves its own encoding process, you need to have a Dolby Digital receiver or amplifier
to accurately decode the signal, which is transferred from a component, such as a DVD
player, via either a digital optical connector or digital coaxial connector.
Dolby Digital EX
Dolby Digital EX is actually based on the technology already developed for Dolby
Digital 5.1. This process adds a third surround channel that is placed directly behind the
listener.
In other words, the listener has both a front center channel and, with Dolby Digital EX, a
rear center channel. If you are losing count, the channels are labeled: Left Front, Center,
Right Front, Surround Left, Surround Right, Subwoofer, with a Surround Back Center
(6.1) or Surround Back Left and Surround Back Right (which would actually be a single
channel - in terms of Dolby Digital EX decoding). This obviously requires another
amplifier and a special decoder in the A/V Surround Receiver.
The Benefits Of Dolby Digital EX
So, what is the benefit of the EX enhancement to Dolby Digital Surround Sound?
Essentially, it boils down to this: In Dolby Digital, much of the surround sound effects
move towards the listener from the front or sides. However, the sound loses some
directionality as it moves along the sides to the rear, making a precise directional sense of
sounds from moving objects moving or panning across the room difficult. By placing a
new channel directly behind the listener, panning and positioning of sounds emanating
from the sides to the rear are much more precise. Also, with the additional rear channel, it
is possible to originate sounds and effects from the rear more precisely as well. This
places the listener even more in the center of the action.
Dolby Digital EX Compatibility
Dolby Digital EX is completely compatible with Dolby Digital 5.1. Since the Surround
EX signals are matrixed within the Dolby Digital 5.1 signal, software titles encoded with
EX can still be played on existing DVD players with Dolby Digital outputs and decoded
in 5.1 on existing Dolby Digital Receivers.
Although you may end up buying new EX-encoded versions of films you may have
already in your collection when you finally get your EX setup running, you can still play
your current DVDs through a 6.1 Channel Receiver and you will be able to play your
new EX-encoded discs through a 5.1 channel receiver, which will just retain the
additional information with the current 5.1 surround scheme.

Surround Sound - Part Four - More

Options
Dolby Pro Logic II - Dolby Pro Logic IIx - Dolby Pro
Logic IIz - Audyssey DSX
Dolby Pro Logic II and Dolby Pro Logic IIx
Although the previously outlined Dolby surround sound formats are designed to decode
surround that is already encoded on DVDs or other material, there are thousands of music
CDs, VHS movies, Laserdiscs, and television broadcasts that contain only simple analog
two channel stereo or Dolby Surround encoding.
Surround Sound For Music
Also, with surround schemes such as Dolby Digital and Dolby Digital-EX primarily
designed for movie viewing, there is a lack of an effective surround process for music
listening. In fact, many discriminating audiophiles reject much of the surround sound
schemes, including the new SACD (Super Audio CD) and DVD-Audio multi-channel
audio formats, in favor of traditional two-channel stereo playback.
Manufacturers, such as Yamaha, have developed sound enhancement technologies
(referred to as DSP - Digital Soundfield Processing) that can can place the source
material in a virtual sound environment, such as a jazz club, concert hall, or stadium, but
cannot "convert" two or four channel material into a 5.1 format.
The Benefits Of The Dolby Pro Logic II Decoding Process
With this in mind, Dolby Labs has come to the rescue with an enhancement to its original
Dolby Pro-Logic technology that can create a "simulated" 5.1 channel surround
environment from a 4-Channel Dolby Surround signal (dubbed Pro-Logic II). Although
not a discrete format, such as Dolby Digital 5.1 or DTS, in which each channel goes
though its own encoding/decoding process, Pro Logic II makes an effective use of
matrixing to deliver an adequate 5.1 representation of a film or music soundtrack. With
advancements in technology since the original Pro-Logic scheme was developed over 10
years ago, channel separation is more distinct, giving Pro Logic II the character of a
discrete 5.1 channel scheme, such as Dolby Digital 5.1.
Extracting Surround Sound From Stereo Sources
Another benefit of Dolby Pro Logic II, is the ability to adequately create a surround
listening experience from two-channel stereo music recordings. I, for one, have been less
than satisfied trying to listen to two-channel music recordings in surround sound, using
standard Pro Logic. Vocal balance, instrument placement, and transient sounds always
seem to be somewhat unbalanced. There are, of course, many CD's that are Dolby
Surround or DTS encoded, which are mixed for surround listening, but the vast majority
are not and thus, can benefit from the application of Dolby Pro-Logic II enhancement.
Dolby Pro Logic II also has several settings that allow the listener to adjust the
soundstage to suit specific tastes. These settings are:
Dimension control, which allows users to adjust the soundstage either towards the front
or towards the rear.
Center Width Control, which Allows variable adjustment of the center image so it may
be heard only from the Center speaker, only from the Left/Right speakers as a "phantom"
center image, or various combinations of all three front speakers.
Panorama Mode which extends the front stereo image to include the Surround speakers
for a wraparound effect.
A final advantage of a Pro-Logic II decoder is that it can also perform as a "regular" 4-
channel Pro-Logic decoder, so, in essence, receivers that include Pro-Logic decoders can,
instead, include Pro Logic II decoders, giving the consumer more flexibility, without
having to having the expense of requiring two different Pro-Logic decoders in the same
unit.
Dolby Pro Logic IIx
Lastly, a more recent variant of Dolby Pro Logic II is Dolby Pro Logic IIx, which
expands the extracting capabilities of Dolby Pro Logic II, including its preference
settings, to 6.1 or 7.1 channels on Dolby Pro Logic IIx-equipped receivers and preamps.
Dolby Pro Logic IIx serves to deliver the listening experience to a greater number of
channels without having to remix and reissue the original source material. This makes
your record and CD collection easily adaptable to the latest surround sound listening
environments.
Dolby Prologic IIz
Dolby Prologic IIz processing is an enhancement that extends surround sound vertically.
Dolby Prologic IIz offers the option of adding two more front speakers that are placed
above the left and right main speakers. This feature adds a "vertical" or overhead
component to the surround sound field (great for rain, helicopter, plane flyover effects).
Dolby Prologic IIz can be added to either a 5.1 channel or 7.1 channel setup. For more
details, check out the official Dolby Pro Logic IIz page.
NOTE: Yamaha offers a similar technology on some of its home theater receivers called
Presence.
Dolby Virtual Speaker
Although the trend towards surround sound relies on adding additional channels and
speakers, the requirement of multiple speakers around an entire room is not always
practical. With that in mind, Dolby Labs has developed a way to create a fairly accurate
surround experience that gives the illusion that you are listening to a complete surround
speaker system, but is utilizing just two speakers and a subwoofer.
Dolby Virtual Speaker, when used with standard stereo sources, such as CD, creates a
wider sound stage. However, when stereo sources are combined with Dolby Prologic II,
or Dolby Digital encoded DVDs are played, Dolby Virtual speaker creates a 5.1 channel
sound image using technology that takes into account sound reflection and how humans
hear sound in a natural environment, enabling the surround sound signal to be reproduced
without needing five or six speakers.
Audyssey DSX (or DSX 2)
Audyssey, a company that develops and markets automatic speaker room equalization
and correction software, has developed its own immersive surround sound technology:
DSX (Dynamic Surround Expansion).
DSX adds front vertical-height speakers, similar to Prologic 11z, but also incorporates the
addition of left/right wide speakers positioned between the front left and right and
surround left and right speakers. For a more detailed explanation and speaker setup
illustrations, check out the Official Audyssey DSX Page

Surround Sound - Part Five - Other

Players in the Game
DTS - SRS Labs - Headphone Surround - Dolby
TrueHD - DTS-HD Master Audio
DTS
DTS is also a well-known player in surround sound and has adapted its surround sound
process for home use. Basic DTS is a 5.1 system just like Dolby Digital 5.1, but since
DTS uses less compression in encoding process, many feel that DTS has a better result on
the listening end. Also, while Dolby Digital is mainly intended for the Movie Soundtrack
experience, DTS is used in the mixing and reproduction of Musical performances.
DTS-ES
DTS has come up with its own 6.1 channel systems, in competition with Dolby Digital
EX, referred to as DTS-ES Matrix and DTS-ES 6.1 Discrete. Basically, DTS-ES Matrix
can create a center rear channel from existing DTS 5.1 encoded material, while DTS-ES
Discrete requires that the software being played already has a DTS-ES Discrete
soundtrack. As with Dolby Digital EX, DTS-ES and DTS-ES 6.1 Discrete formats are
backwards compatible with 5.1 channel DTS Receivers and DTS encoded DVDs.
DTS Neo:6
In addition to DTS 5.1 and DTS-ES Matrix and Discrete 6.1 channel formats, DTS also
offers DTS Neo:6. DTS Neo:6, functions in a similar fashion to Dolby Prologic II and
IIx, in that, with receivers and preamps that have DTS Neo:6 decoders, it will extract a
6.1 channel surround field from existing analog two-channel material.
DTS Neo:X
The next step that DTS has taken is to introduce its 11.1 channel Neo:X format. DTS
Neo:X takes cues already present in either 5.1 or 7.1 channel soundtracks and creates
height and wide channels, enabling a more enveloping "3D" sound. To experience the
maximum benefit of DTS Neo:X processing, it is best to have 11 speakers, with 11
channels of amplification, and a subwoofer. However, DTS Neo:X can be modified to
work with a 9.1 or 9.2 channel configuration.
DTS Surround Sensation
Surround Sensation creates phantom center, left, right, and surround channels within a
two-speaker or stereo headphone setup. It is able to take any 5.1 channel input source and
recreate a surround sound experience with just two speakers. In addition, surround
sensation can also expand two-channel compressed audio signals (such as MP3) for a
more surround-like listening experience.
SRS: Tru-Surround and Tru-Surround XT
SRS Labs is another company that also offers innovative technologies that can enhance
the home theater experience (Note: As of July 23th, 2012, SRS Labs is now officially a
part of DTS).
Tru-Surround has the ability to take multi-channel encoded sources, such as Dolby
Digital, and reproduce the surround effect by just using two-speakers. The result is not as
impressive as true Dolby Digital 5.1 (the front and side surround effects are impressive,
but the rear surround effects fall a little short, with the sense they are coming from just to
rear of your head rather than from the back of the room). However, with many consumers
reluctant to fill their room with six or seven loudspeakers, Tru-Surround and Tru-
SurroundXT do give the ability to enjoy 5.1 channel sound within a normally-limited two
channel listening environment.
SRS Circle Surround and Circle Surround II
Circle Surround, on the other hand, approaches surround sound in a unique way. While
Dolby Digital and DTS approach surround sound for a precise directional standpoint
(specific sounds emanating from specific speakers), Circle Surround emphasizes sound
immersion. To accomplish this, a normal 5.1 audio source is encoded down to two
channels, then re-decoded back into 5.1 channels and redistributed back to the 5 speakers
(plus subwoofer) in such a way as to create a more immersible sound without loosing the
directionality of the original 5.1 channel source material.
The results are more impressive than that of Tru-Surround or Tru-Surround XT.
First, panning sounds such as flying planes, speeding cars, or trains, sound even as they
cross the sound stage; often in DD and DTS, panning sounds will "dip" in intensity as
they move from one speaker to the next.
Also, rear-to-front and front-to-rear sounds flow smoother as well. Second,
environmental sounds, such as thunder, rain, wind, or waves full the sound field much
better than in DD or DTS. For example, instead of hearing rain coming from several
directions, the points in the soundfield between those directions are filled, thus placing
you within the rain storm, not just listening to it.
Circle Surround provides an enhancement of Dolby Digital and similar surround sound
source material without degrading the original intent of the surround sound mix.
Circle Surround II takes this concept further by adding an additional rear center channel,
thus providing an anchor for sounds emanating from directly behind the listener.
Headphone Surround: Dolby Headphone, CS Headphone, Yamaha Silent Cinema,
Smyth Research, and DTS Headphone:X.
Surround Sound is not limited to the large-multi channel system, but can also be applied
to headphone listening. SRS Labs, Dolby Labs, and Yamaha all have incorporated
surround sound technology with the headphone listening environment.
Normally, when listening to audio (either music or movies) the sound seems to originate
from within your head, which is unnatural. Dolby Headphone SRS Headphone, Yamaha
Silent Cinema, and Smyth Research employ technology that not only gives the listener an
enveloping sound, but removes it from within listener's head and places the sound field in
the front and side space around the head, which is more like listening to a regular
speaker-based surround sound system.
In another development, DTS has developed DTS Headphone:X that can provide up to an
11.1 channel surround sound listening experience using any pair of headphones plugged
into a listening device, such as a smartphone, portable media player, or home theater
receiver that is equipped with DTS Headphone:X processing.
Higher Definition Surround Sound Technologies: Dolby Digital Plus, Dolby
TrueHD, and DTS-HD Master Audio
With the introduction of Blu-ray Disc and HD-DVD, in conjunction with the HDMI
interface connection, the development of high definition surround sound formats in both
DTS (in the form of both DTS-HD and DTS-HD Master Audio) and Dolby Digital (in the
form of Dolby Digital Plus and Dolby TrueHD) provides extended accuracy and realism.
The increased storage capacity of Blu-ray and HD-DVD, and wider bandwidth transfer
capabilities of HDMI, which is required for accessing Dolby Digital Plus, Dolby
TrueHD, and DTS-HD, have allowed for true, discreet, audio reproduction for up to 7.1
Channels of surround sound, while still being backwards compatible with older 5.1
channel surround sound formats and audio/video components.
Note: HD-DVD has been discontinued but is referenced in this article for historical
purposes.
Looking Into The Future
Dolby Atmos - Are You Ready for 64-Channel Surround Sound?
Multi-Dimensional Audio - Rethinking Surround Sound
Conclusion - For Now...
Today's surround sound experience is the result of decades of evolution. The surround
sound experience is now easily accessible, practical, and affordable for the consumer,
with more to come in the future. Go get surrounded!

Subwoofers - What You Need To Know

How do subwoofers affect your Home Theater experience? What is a subwoofer,
anyway?
Subwoofers are becoming more and more crucial to the home theater experience. When
you go to the movie theater, you marvel not only at the images projected on the screen,
but the sounds emanating around you. What really grabs you, though, is the sound you
actually feel; the deep bass that shakes you up and gets you right in the gut.
A specialized speaker, known as a subwoofer, is responsible for this experience. The
subwoofer is designed only to reproduce the lowest of audible frequencies. In home
theater, this is often referred to as LFE (Low Frequency Effects.
With the popularity of home theater sound systems resulting in specialized speakers for
center channel dialog, main soundtracks, and surround effects, the need for a speaker to
reproduce just the deep bass portion of a movie soundtrack is all the more important.
Although these subwoofers are not quite as "thunderous" as the subwoofers employed at
the local movie theater, these unique loudspeakers can still shake the house down or
annoy the downstairs neighbors in your apartment or condo complex.
Subwoofers come in two basic types, Passive and Powered.
Passive Subwoofers
Passive subwoofers are powered by an external amplifier, in the same fashion as other
speakers in your system. The important consideration here is that since extreme bass
needs more power to reproduce low frequency sounds, your amplifier or receiver needs to
be able to output enough power to sustain bass effects in the subwoofer without draining
the amp. How much power depends on the requirements of the speaker and the size of the
room (and how much bass you can stomach!).
Powered Subwoofers
To solve the problem of inadequate power or other characteristics that may be lacking in
a receiver or amplifier, powered subwoofers are self-contained speaker/amplifier
configurations in which the characteristics of the amplifier and sub woofer are optimally
matched.
As a side benefit, all a powered subwoofer needs is a line output from an amplifier. This
arrangement takes a lot of the power load away from the amp/receiver and allows the
amp/receiver to power the mid-range and tweeters more easily.
For more on the differences and how to hook-up Passive and Powered Subwoofers, read
my supplementary article: Passive Subwoofers vs Powered Subwoofers.
Additional Subwoofer Characteristics
Additional subwoofer design variations include Front-firing , and Down-firing, and the
use of Ports or Passive Radiators.
Front-firing subwoofers employ a speaker mounted so that it radiates the sound from the
side or front of the subwoofer enclosure. Down-firing subwoofers employ a speaker that
is mounted so that it radiates downward, towards the floor. In addition, some enclosures
employ an additional port, which forces out more air, increasing bass response in a more
efficient manner than sealed enclosures. Another type of enclosure utilizes a Passive
Radiator in addition to the speaker, instead of a port, to increase efficiency and
preciseness. Passive radiators can either be speakers with the voice coil removed, or a flat
diaphragm.
Crossovers
Typically, a good subwoofer has a "crossover" frequency of about 100hz. The crossover
is an electronic circuit that routes all frequencies below that point to the subwoofer; all
frequencies above that point are reproduced the main, center, and surround speakers.
Gone is the need for those large 3-Way speaker systems with 12" or 15" woofers. Smaller
satellite speakers, optimized for mid-and-high frequencies, take up much less space and
are now common in many home theater systems.
Directionality
In addition, since the deep-bass frequencies reproduced by the subwoofers are non-
directional (as frequencies that are at or below the threshold of hearing). It is very
difficult for our ears to actually pin-point the direction in which the sound is coming.
That is why we can only sense that an earthquake seems to be all around us, rather from
coming from a particular direction. As a result, the subwoofer can be placed anywhere in
the room, however, optimum results depend on room size, floor type, furnishings, and
wall construction. Typically, best placement for a subwoofer is in the front of the room,
just to the left or right of the main speakers. There are more installation tips in the
conclusion of this article.

Subwoofers: What You Need To Know -

Page 2
Subwoofer Alternatives - Shopping and Installation
Tips
Since the subwoofer experience entails more of what we can feel than what we can hear,
using a loudspeaker-based design is not the only approach that can be used to reproduce
low frequency information. For some interesting alternatives to the traditional subwoofer,
consider the following:
The Buttkicker
More than just a subwoofer, the Buttkicker is a low frequency transducer that not only
puts more feeling in your bass, but....Kicks Butt! Using a unique "suspended magnetic
system" to reproduce sound waves that are not air dependent, the Buttkicker can
reproduce frequencies down to 5HZ. This is well below human hearing, but not below
human feeling! Variations of the Buttkicker are found in professional settings, such as
movie theaters, and concert halls, but have been adapted for use in your own home
theater.
Clark Synthesis Tactile Sound Transducer
Don't just hear sound, touch it! With a very compact transducer design, the Clark
Synthesis Tactile Sound Transducer can be placed inside (or on the bottom of) chairs,
couches, etc... to produce deep bass response that is both intimate and effective (others in
the room will wonder what is getting you so excited!).
Crowson Technology Tactile Transducers
The key technology employed in Crowson Tactile Transducers is Linear Direct-Drive.
Instead of vibrating air, like a subwoofer, or employing a piston that vibrates inside
housing that indirectly transfers the shaking sensation to a chair, such as a bass shaker
(both of which take of energy), Linear Direct Drive transfers sonic vibrations directly
through the chair itself via its feet, which is similar to techniques used in direct hearing
via human bone conduction. Thus, if someone is sitting in the chair, they will feel the
direct effect of the linear drive process on their body.
This method requires much less energy to produce vibration effects than other methods,
thus enabling a more dynamic effect with faster response times. In other words, the
Crowson Tactile Transducer can capture the subtle vibrations of a car driving on a
country road to the big boom of an atomic bomb explosion.
Bass Shakers:
Bass Shakers are another type of transducer device designed to reproduce inaudible low
frequencies, designed to give an extra "punch" to your sound system. The Shaker is
usually attached directly to the object to be "shaken", such as a chair (similar to the Clark
Tactile Transducer) in order to realize its effect. Bass Shakers can be used not only by
themselves, but in conjunction with a traditional subwoofer setup.
One final note on these subwoofer alternatives, however. Although very effective in home
theater setups for effects that contain a lot of inaudible low frequency information, such
as explosions, earthquakes, gun blasts, rocket and jet motor effects, Shakers and Tactile
Transducers are not very effective in the typical home music listening environment. A
good, traditional, subwoofer is more than adequate for the lowest musical effects, such as
bass drums.
Subwoofers - Shopping and Installation Tips
Despite all of the technical specifications and design factors of subwoofers, the type of
subwoofer you choose for your system depends on the characteristics of the room and
your own preferences. When you go to a dealer, take a favorite DVD and/or CD that has
a lot of bass information and listen to how the bass sounds through various subwoofers.
In addition, make sure you find out the return policy of your dealer, just in case the
subwoofer doesn't perform well in your listening environment. Place the subwoofer in
various parts of the room, using the owner's manual as a guide, to find out what sounds
pleasing to you.
The subwoofer should not sound "boomy", but deep and tight. This is especially
important if you intend to use your subwoofer for music listening. Many subwoofers are
great for DVD movies, but may not perform well with the subtle deep bass in music
performances. When installing your subwoofer, experiment with the crossover settings.
In addition, most AV receivers have internal crossover settings for your subwoofer which
depends on whether your other speakers are large or small. In this way your subwoofer
can either take the entire bass load or split the bass load with large main speakers, with
the subwoofer only producing the very lowest bass frequencies.
Also, if you live in an upstairs apartment, a down-firing subwoofer may disturb your
downstairs neighbors more easily that a front-firing design. Lastly, in some cases,
integrating two subwoofers into your system may provide a better option, especially in
very large room.
For some additional subwoofer installation tips, read the following articles from
About.com Stereos: How to Get the Best Subwoofer Performance, How to Connect a
Subwoofer to a Receiver, Processor, or Amplifier, and The Benefits of Using Multiple
Subwoofers.
To get you started in finding a subwoofer that may be right for your system, check out my
list of top subwoofers and subwoofer brands.

Guide to Stereo Component Features and Specifications

How Much Amplifier Power Do You Need?

After deciding on a receiver, integrated amplifier or separate components, power output
is the next consideration. Power output requirements are determined by the speakers, the
size of the listening room and how loud you like to listen. Power output specifications are
commonly misunderstood. An amplifier with 200-watts per channel will not play twice as
loud as an amplifier with 100-watts per channel. In fact, the difference in maximum
volume will be hardly audible, about 3 decibels. A typical amplifier playing at a moderate
level will only output about 15-watts of power to the speakers. When the music reaches a
peak or crescendo the amplifier will output much more power, but only during the period
of highest demand. Read more about amplifier power and how much power is really
required.

How Many Source Components Do You Want to Connect?

Some stereo systems include a CD Player, DVD Player, Tape Deck, Turntable, Hard Disk
Recorder, Game Console and video components, while other systems may have only a
CD or DVD player. Consider the number and type of components you have when
selecting a receiver, amplifier or separates. This Guide to Audio and Video Connections
describes the different types of components and connections available.

Power output is one of the most important considerations in choosing a receiver. Power is
expressed in watts per channel and the decision about how much power you need should
be based on your selection of loudspeakers, the size and acoustic characteristics of your
listening room, and how loud you like to listen. It is always best to match the power
requirements of the speakers with the output power of the receiver. Some speakers
require more or less power, expressed as loudspeaker sensitivity (in decibels, dB), which
is a measure of how much sound output is produced with a specified amount of amplifier
power. Power output and speaker volume is not a linear relationship. For example, a
receiver with 100 watts per channel will not play twice as loud as a receiver with 50 watts
per channel using the same speakers – the difference in maximum loudness would be
barely discernable, only 3 decibels (dB). Rather, more amplifier power will allow the
system to handle musical peaks without straining.
Generally, a speaker with lower sensitivity (88dB-93dB) will require more amplifier
power than a speaker with a higher sensitivity (94dB to 100dB or more) to play at the
same volume level. Most receivers are rated with a minimum power output of 75 watts
per channel and higher. When comparing power output, it is important to know how the
power is measured. The most accurate measure of power is RMS (Root Mean Square, a
mathematical formula), as opposed to peak output power. Some manufacturers inflate
specifications by measuring power at a single frequency, say 1kHz, instead of the entire
range, 20Hz-20kHz. When comparing receiver power outputs, always make sure they are
measured the same way.
5.1, 6.1, 7.1 Channels – How Many Do You Need?
A/V receivers are distinguished from two-channel or stereo receivers by having five or
more amplifier channels to power speakers that can reproduce movie theater sound or
multi-channel music in your home. Most DVD discs and other multi-channel sources are
encoded in Dolby Digital 5.1 and/or DTS 5.1 channel sound for playback on home
theater systems. A basic system consists of 5.1 channels of sound. The five channels are
left and right, like a stereo system, a center channel for movie dialog or music vocals and
on-screen sound, and left and right surround channels, for special effects and surround
sound. An additional subwoofer channel (the .1 LFE channel, Low Frequency Effects)
adds very low bass for music sources and special effects on DVD movie sound tracks.
The composite of the five main channels plus a subwoofer channel produces a
“soundfield” that envelops the listener. A 5.1 channel system is capable of playing
programs encoded in Dolby Digital and DTS.
6.1 channel a/v receivers have an additional rear-center channel output, and are becoming
increasingly popular, even at lower prices. Some DVDs are encoded with 6.1 channel
Dolby Digital EX and DTS-ES, and can be played back on this type of system. If
properly installed, 6.1-channel sound can create a more enveloping surround sound
effect.
7.1 channel receivers have three front channels, two surround and two surround-back
channels, plus a subwoofer channel. The additional rear channels produce a soundfield
with more precise placement of surround effects. Some 7.1 channel receivers offer
THX™ soundfield enhancement, which is a system developed by Lucas Film™ designed
to present film sound or multi-channel music with the most authentic quality. THX
processing is offered as Select or the more advanced Ultra 2 format, which is optimized
for movies and multi-channel music. Many other manufacturers have proprietary
soundfield programs, called DSP, which also provide enhancements for music and movie
sound. Sony, for example, has Digital Cinema Sound™ and Yamaha has Cinema DSP™.
Many 7.1 channel receivers also permit the two surround-back channels to be reassigned
to a second zone for a multi-room system, leaving the main system with 5.1 channels.

Audio and Video Connections

Audio and video inputs are important because they determine the number of components
you can connect to your system. Consider the components you have now and those you
might want to add in the future. At a minimum you will want to connect a DVD and/or
CD player and a television, but you may want to add a DVR (Digital Video Recorder) or
a cable or satellite dish receiver in the future. Digital (optical and coaxial) and analog
audio connections are equally important.
Most receivers have six-channel analog inputs, which are very important for playback of
the latest, high-resolution audio discs such as DVD-Audio or SACD. These discs offer
significantly better fidelity than CDs but require a capable disc player and corresponding
receiver inputs. The six-channel analog input may also be necessary for any future audio
formats that come along. Lately, many higher priced receivers also offer an IEEE 1394
FireWire™ digital connection (also called I-Link) between the player and the receiver for
playback of DVD-A and SACD discs.
Video connections come in three types: composite, S-video and component video, with
component offering the best performance including better contrast and sharper images.
Most new DVD players and television monitors have component video connections.
Many receivers feature a circuit that “up-convert” all incoming video signals to
component video with a single video cable connecting the a/v receiver with the TV,
which simplifies connections and improves video quality. The bandwidth or frequency
range of the component video circuit in the receiver is important for the highest video
quality. Bandwidth, measured in megahertz (MHz), is a specification that indicates how
much information can go from the source through the video circuits in the receiver to the
television or monitor. More bandwidth generally means a better picture. The minimum
bandwidth is about 30MHz, but 60-100MHz is better. Some higher priced receivers, such
as the Yamaha RX-Z9 include video enhancement circuits to further improve picture
quality.
Front panel A/V inputs on the receiver are important if you want to connect a video game
console or a camcorder to the receiver without removing it from the cabinet or shelf to
use the rear-panel connections. Typically front panel inputs are analog audio (sometimes
digital audio, too) and composite or S-video inputs.
Multi-Room / Multi-Source Capability
If you think you might enjoy sound in more than one room, this is an important feature to
consider. Many receivers can route signals to second or third zones (rooms), enabling you
to enjoy audio and video throughout your home using the same receiver. Some receivers
send analog line level signals (un-amplified), speaker level signals or both to the remote
zone(s). In either case, you will have to run audio and video cables to the remote zones
plus you will need to add an amplifier and speakers. Consider whether you have the skills
and tools to do this, or if you need to hire a professional contractor or electrician to install
the necessary wiring. Sherwood has introduced new receivers and separate components
with Advanced Room 2 that outputs digital audio signals to a second zone for higher
quality multi-room sound. More about multi-room audio distribution systems.
Remote Controls
In addition to an a/v receiver, you will probably have a CD player, DVD player and a
television, and maybe a tape player, cable box or satellite receiver and a DVR. Soon your
coffee table will be covered with remote controls. You need a Universal Remote Control,
which can be programmed with codes to control all of your components, or can learn
codes from other remote controls. Programming and learning functions are easy and will
simplify the operation of your system and reduce clutter. Many remotes have Macro
Command capability, which can execute several system functions with the touch of one
button. Higher priced receivers often include an LCD touch-panel remote control that can
operate the entire system by pressing icons on the remote’s screen.

System Control and Upgrade Capability

Receivers with RS-232 serial ports are often compatible with third-party control systems
such as those made by Crestron, AMX and others. These touch-panel controls allow
customization of your system and are generally installed by a professional since they
involve complex programming. They are often more expensive than the audio
components, but offer simplicity in system control and operation.
If you’ve waited to purchase an a/v receiver because you’re concerned about
obsolescence, wait no more. Many mid-priced receivers now include microprocessor
upgrade capability, via an RS-232 serial computer connection or a USB port. Upgrades
could include new DSP soundfield modes, new decoding formats such as Dolby Pro-
Logic II, or other system improvements. Some of the upgrades may be available on the
manufacturer’s website as a download or through your dealer, and may involve a cost.
System Set-Up
Some manufacturers offer special computer applications that allow you to set-up or
configure your a/v receiver using a PC. Yamaha, for example, offers a free application
from their website called Receiver Editor that allows you to configure most receiver
settings on a PC that is connected to the RS-232 port on the receiver. B & K has
developed BKcSuite on CD-ROM or as an Internet download for the same purpose. The
settings can also be saved as a file for future use. Manufacturers that include RS-232
ports are as follows, but not limited to B & K, Denon, Marantz, Pioneer, Sherwood and
Yamaha.
Manufacturers have also developed systems that optimize the receiver for the room
acoustics, making set-up easier and more effective. Pioneer offers MCACC (Multi-
Channel Acoustic Calibration System) and Yamaha has introduced YPAO (Yamaha
Parametric Room Acoustic Optimizer), which checks for and adjusts speaker size,
distance, frequency response, levels and other settings. Both of these do a good job of
calibrating the system and take some of the guesswork out of system set-up.

Best Stereo Systems

Top Picks Stereo Component Systems
Best Stereo Systems: These three top pick stereo systems have a stereo
receiver, a disc player and two speakers with suggested retail prices
ranging from $700 to $1,997. Click on the 'Compare Prices' links to find
the best prices.
System One: Yamaha RX-397 Receiver, Yamaha DVD-S661 CD/DVD
Player, Polk Audio TSi100 Speakers - MSRP: $707
The Yamaha RX-397 Stereo Receiver has 50-watts/channel, with dual zone audio output
and a remote control. Read a profile of the Yamaha RX-397. The Yamaha DVD-S661
DVD Video player has built-in Dolby Digital and DTS decoders, HDMI out and DIVX
disc playback. The Polk Audio TSi100 compact bookshelf speakers are two-way with a
5.25" woofer and 1" tweeter and are available in black or cherry finish. They can be
placed on a shelf or wall-mounted and can also be used as surround speakers in a home
theater system.
Yamaha RX-397
Yamaha DVD-S661
Polk Audio TSi100
System Highlights:
← Most affordable stereo component system.
← Zone 2 output can be used for music in a separate room (wiring and amplifier
 required).
← Low Impedance capability means it can power speakers from 4 to 8 ohms.
← High damping factor (150) means tighter bass.

System Two: Denon DRA-397 Receiver, Denon DVM-1845 5-Disc

CD/DVD Changer, Monitor Audio BR-2 Speakers - MSRP: $1,067
The Denon DRA-397 Receiver has 80 watts per channel and features dual zone audio
outputs, XM Satellite Radio Ready and has an optional iPod docking station. The Denon
DVM-1845 is a 5-disc progressive scan DVD Changer with 192kHz/24-bit audio D/A
converters. The Monitor Audio BR-2 speakers are two-way bass reflex with great sound
quality and are available in black, cherry or finishes. Read my full review of the Monitor
Audio BR-2 speakers.
Denon DRA-397
Denon DVM-1845
System Highlights:
← DRA-397 is XM Satellite Radio Ready
← Optional iPod Docking Station
← 5-disc CD/DVD Changer

System Three: Pioneer SX-A9 Receiver, Pioneer PD-D6 CD/SACD

Player and Monitor Audio BR-5 Tower Speakers MSRP: $1,997
The SX-A9 and PD-D6 are from Pioneer's Elite series of audio components. The receiver
has 70-watts per channel and dual-mono construction with two toroidal power supplies,
which is like having two separate amplifiers that allow the receiver to respond to the
power needs of each channel independently improving channel separation and soundstage
performance. The SX-A9 incorporates Pioneer’s Wide-Range Linear Circuit for extended
frequency response from 5Hz to 100kHz, especially important for SACD sources. Read
my full review of the SX-A9 Receiver. The companion Pioneer PD-D6-J CD/SACD
player is equipped with Burr-Brown 192 kHz / 24-bit DACs (digital to analog
converters), which offer better dynamic range, greater detail and improved low-level
signal resolution and detail. It also employs Pioneer’s proprietary Hi-Bit Legato Link
Conversion that extends high-frequency response out to 40kHz. Read my full review of
the PD-D6-J. The Monitor Audio BR-5 speakers are one of the best values I've heard.
Read my profile of the BR-5 speakers. They have excellent tonal balance, outstanding
midrange clarity and warm, yet articulate bass response, a nice combination of warmth
and definition.

Before You Buy, Read this Guide to Stereos and Stereo Systems
A complete stereo system has several elements including speakers, components, sources
and the listening room. Whether you're a stereo novice or an experienced listener, this
overview covers the essential parts of a good stereo and how to get the best sound from
your system.
The Listening Room
The acoustic quality of your listening room is the foundation of a good stereo system and
plays an important role in the way your system ultimately sounds. Your listening room is
as at least as important as choosing the right speakers and components. Optimizing
speaker placement, listening position and purchasing room acoustic treatments is the best
way to get the most performance from your system. Click on the links below for more
information and guidelines about speaker placement, room acoustic treatments and
listening position.
← Correct Speaker Placement Guidelines
← Room Acoustic Treatments
← The 'Sweet Spot'
Stereo Speakers
Stereo speakers determine the overall sound quality of your stereo system more than any
other component. Speakers come in all sizes, types, shapes and prices so you have a lot of
choices. There is no 'best' speaker, only the one that is right for you and your needs.
Sound is a very personal decision and you should listen to several models before
purchasing speakers. Learn more about selecting speakers in the following articles.
← Speaker Basics
← How to Choose Speakers
← How to Properly Connect Speakers
Stereo Components & Product Reviews
Stereo components are available in a wide variety of types and prices from separate
components, stereo receivers, integrated amplifiers, or as a pre-packaged system. The
stereo components best for you depends on your budget, listening preferences and how
often you listen to music. You get a lot for your money with stereo components and even
a modest stereo system can provide years of music enjoyment. The following articles and
product reviews will help you make the best buying decisions.
← Introduction to Stereo Components
← Buying a Pre-Packaged Stereo System
← Product Reviews
Stereo Source Components
A source component is first in the audio reproduction chain and is just as important as a
receiver or speakers. Source components can be analog or digital. As an example, a
digital source component can be a CD or DVD player, and an analog source component
could be a tape player or phonograph. Learn more about different source components in
this section.
← Guide to DVD Player Audio and Video Features
← Guide to SACD Discs and Players
Multiroom Audio Systems - Music in Every Room
Multiroom audio systems make it possible to listen to music in any room in your home,
even outdoors. A multiroom system can be as simple as using the Speaker B switch on
your receiver to more sophisticated systems that allow you to listen to different sources in
every room and operate the system with a remote control. There are many types of
multiroom audio systems and new technologies are coming to market. Learn more about
multiroom audio systems.
← Types of Multiroom Audio Systems
← HomePlug Technology
 ← Using the Speaker B Switch on Your Receiver
Stereo System Accessories
Accessories help you get the most from your stereo system. Read more to learn about
stereo accessories, such as premium speaker wires that can improve performance and
make your listening experience more enjoyable. Speaker stands are useful for getting the
best sound from bookshelf speakers and high-quality headphones can be a good substitute
for a speaker system in an apartment, condominium or dorm room.
← Stereo Accessories to Consider
← High-quality Stereo Headphones
← Top Ten Stereos and Accessories
Advanced Stereo Topics
Beyond the basics includes advanced audio topics such as new technologies that make it
easier to have music throughout your home, automatic room equalization systems that
compensate for typical room acoustic problems, the best ways to maximize the
performance of an audio system and how to choose the best type of surround sound
speakers.

Place Speakers Correctly

Correct stereo speaker placement is the best place to start to get great audio performance
from your system. It's free and all it takes is some time and patience.
In general, don’t place stereo speakers too near the front wall (the wall behind the
speakers). Placement closer to the wall amplifies bass response and probably makes the
bass sound too loud or boomy. Every room is different, but here are two speaker
placement methods that should make your system sound better.
Difficulty: Easy
Time Required: One hour
Here's How:
Apply the Golden Rectangle Rule
If your room permits, try placing the speakers about 3’ from the front wall. This
reduces bass reflections from the front wall and helps tame boomy bass.
The distance from the side wall(s) is equally important. The Golden Rectangle
Rule states that the speaker’s distance from the side wall should be 1.6 times the
distance from the front wall. If the distance from the front wall is 3’, the distance
from the side wall should be 4.8’ from the side wall (or vice versa if your room is
wider than longer). Finally, angle the speakers towards the listening spot, called
speaker toe-in.
Apply the 1/3 - 1/5 Rule
Position the speakers so that the distance between the front wall is 1/3 or 1/5 the
length of the room. Both of these methods prevent the speaker from exciting room
resonances. Angle the speakers towards the listening position, as above. Your
listening position is as important as speaker position to achieve the best sound
quality. More on finding the 'sweet spot' soon.
Good Listening!
Tips:
Don't be afraid to experiment with speaker placement. Every room is different and the
methods presented above are guidelines.
Use masking tape on the floor to mark the speaker position as you experiment with
placement options.

Your Listening Room: The Most Important Component in

Your Stereo System

Here’s a short quiz. You have $1,000 to spend on upgrading your stereo or home theater
system, what do you buy to get the most bang for your buck in sound quality?
Premium speaker cables
A new receiver
Room acoustic treatments
Hi-definition DVD player.
If you answered anything other than ‘room acoustic treatments’, you might achieve only
an incremental improvement in sound quality. If you answered ‘room acoustic treatments’
you would be making a significant upgrade. The reason is simple: The listening room is a
critical component in the sound reproduction chain, at least as important as speakers,
electronics, sources and cables, yet the listening room is often the most neglected
component. When sound waves leave a speaker they interact with the walls, ceiling,
floors, furnishings and other surfaces in the room causing room resonances and
reflections that color the sound you ultimately hear.
Room Resonances
Room resonances are sound waves generated by the speakers from 20Hz to about 300Hz.
The frequency of the resonances are based on the dimensions (length, width and height)
of the listening room. A room resonance either reinforces or attenuates bass frequencies
and the most common symptom is heavy or muddy bass, or conversely, thin, weak bass.
A typical room will have boomy bass somewhere between 50Hz and 70Hz. There is an
easy way to identify the resonances in your room using a room acoustics calculator.
Clicking on the link will download the calculator (Excel file). Enter the dimensions of
your room (height, width and length) and the calculator will determine the problem
frequencies.
The first step in compensating for room resonances is correct speaker placement, which
places the speakers in a position where they do not excite room resonances. It's the first
step towards improving bass response, but if the bass still sounds heavy, the next step is
room acoustic treatments, primarily bass traps. A bass trap absorbs bass a specific
frequencies, thus overcoming the heavy bass caused by room resonances.
Room Reflections
Room reflections are caused by sound, mostly high frequencies reflecting off of adjacent
walls that combine with the direct sounds you hear from the speakers. In most cases, you
hear more reflected than direct sounds. The reflected sounds reach your ears milliseconds
later than the direct sounds because they travel a longer distance. In general, sound
reflections degrade imaging, soundstaging and the overall tonal quality, important
characteristics of a good sound system. A simple way to locate the reflection points in
your room is to have a friend hold a small mirror against the wall while your seated in
your primary listening position. Have the friend move the mirror around the wall until
you can see the speaker in the mirror. The location of the mirror is a reflection point.
The solution for room reflections is acoustic absorbers and diffusors, that when placed
correctly allow you to hear more of the speakers and less of the room. In other words,
more direct sound and less reflected sound. From personal experience I can say that room
acoustic treatments have improved the sound quality of my system more than any
upgrade I have ever made. Any upgrade! When bass improves, tonal balance is restored
and the rest of the system sounds better. When room reflections are controlled (not
eliminated) it is possible to resolve much more detail.

The Benefits of Using Multiple

Subwoofers
It's Not About More Bass, it's About Better Bass
Getting the best bass response is one of the most important performance aspects of a
good audio system and one of the most difficult to achieve. Bass that sounds boomy,
over-resonant and flabby can ruin the listening experience. On the other hand, tight, well-
defined and evenly distributed bass vastly enhances stereo and home theater listening.
Subwoofer Placement & Listening Position
Bass quality is mostly determined by subwoofer placement and listener position. In a
typical home listening room, bass frequencies can sound over-bearing in some locations
and very lean in others depending on where the subwoofer is placed and where you sit.
The reason is room resonances, areas where the some of the subwoofer's sound waves
build up to make the bass too loud, or where some of the waves cancel each other to
make the bass sound weak. Experimenting with subwoofer placement to find the best
location helps eliminate a lot of the peaks and dips.
More Can be Better
There is a third factor that can greatly influence bass quality: the number of subwoofers.
While one subwoofer may produce enough bass for an average sized room, adding
additional subwoofers can reduce the occurrence of room resonances and improve the
overall quality of bass throughout the room. It's not about adding more bass, it's about
improving bass quality and distributing it more evenly throughout the room.
Two, three or even four properly positioned subwoofers can effectively cancel some room
resonances and distribute bass more uniformly in the room, improving bass in multiple
listening positions.
Typically I use two subwoofers in my listening room positioned in opposite corners of the
room, which is a significant improvement over one. I recently reviewed a subwoofer
system with four separate subwoofers powered by the same amp and experienced even
better bass reproduction almost everywhere in the room. While four subs may seem like
overkill, even two subwoofers provide much better bass than a single sub.
Subwoofers are available in a wide range of prices from a few hundred to many
thousands of dollars. The improvement in bass response with multiple subwoofers is so
apparent I would advocate purchasing multiple subs, even if it means spending less on
each one.
Where to Place Two Subwoofers
If you're using two subwoofers, try experimenting with placement as follows:
← In opposite corners of the room
← On the center of the front and rear walls
← On the center of the side walls
Where to Place Four Subwoofers
Using a similar strategy, try placing four subwoofers as follows:
← One in each corner of the room
← One on the center of each wall, front, rear, left, right

Before You Buy A Subwoofer

Subwoofers are becoming more and more crucial to the home theater experience. When
you go to the movie theater, you marvel not only at the images projected on the screen,
but the sounds emanating around you. What really grabs you, though, is the sound you
actually feel; the deep bass that shakes you up and gets you right in the gut.
A specialized speaker, known as a subwoofer, is responsible for this experience. The
subwoofer is designed only to reproduce the lowest of audible frequencies.
Passive Subwoofers
Passive subwoofers are powered by an external amplifier, in the same fashion as other
speakers in your system. The important consideration here is that since extreme bass
needs more power to reproduce low frequency sounds, your amplifier or receiver needs to
be able to output enough power to sustain bass effects in the subwoofer without draining
the amp. How much power depends on the requirements of the speaker and the size of the
room (and how much bass you can stomach!).
Powered Subwoofers
To solve the problem of inadequate power or other characteristics that may be lacking in
a receiver or amplifier, powered subwoofers are self-contained speaker/amplifier units in
which the characteristics of the amplifier and sub woofer are optimally matched.
As a side benefit, all a powered subwoofer needs is a line output from a receiver. This
arrangement takes a lot of the power load away from the amp/receiver and allows the
amp/receiver to power the mid-range and tweeters more easily.
Front-Firing and Down-Firing Subwoofers
Front-firing subwoofers employ a speaker mounted so that it radiates the sound from the
side or front of the subwoofer enclosure.
Down-firing subwoofers employ a speaker that is mounted so that it radiates downward,
towards the floor.
Ports and Passive Radiators
Some subwoofer enclosures also employ an additional port, which forces out more air,
increasing bass response in a more efficient manner than sealed enclosures.
Another type of enclosure utilizes a Passive Radiator in addition to the speaker, instead of
a port, to increase efficiency and preciseness. Passive radiators can either be speakers
with the voice coil removed, or a flat diaphragm.
Crossovers
The crossover is an electronic circuit that routes all frequencies below a specific point to
the subwoofer; all frequencies above that point are reproduced the main, center, and
surround speakers. Typically, a good subwoofer has a "crossover" frequency of about
100hz.
Gone is the need for those large 3-Way speaker systems with 12" or 15" woofers. Smaller
satellite speakers, optimized for mid-and-high frequencies, take up much less space and
are now common in many home theater systems.
Deep Bass is Non-Directional
In addition, since the deep-bass frequencies reproduced by the subwoofers are non-
directional (as frequencies that are at or below the threshold of hearing). It is very
difficult for our ears to actually pin-point the direction in which the sound is coming.
That is why we can only sense that an earthquake seems to be all around us, rather from
coming from a particular direction.
Subwoofer Placement
As a result of the non-directional sound that is reproduced by the subwoofer, it can be
placed anywhere in the room. However, optimum results depend on room size, floor type,
furnishings, and wall construction. Typically, best placement for a subwoofer is in the
front of the room, just to the left or right of the main speakers, or in a front corner of the
room.
The Bottom LIne
Despite all of the technical specifications and design factors of subwoofers, the type of
subwoofer you choose for your system depends on the characteristics of the room and
your own preferences. When you go to a dealer, take a favorite DVD and/or CD that has
a lot of bass information and listen to how the bass sounds through various subwoofers.

Passive Subwoofers vs Powered

Subwoofers
The difference between a Passive and Powered
Subwoofer
Subwoofers are a common part of home theater systems, from that inexpensive home
theater-in-a-box to that high-end custom installed system. However, how you connect a
subwoofer depends on whether is Passive or Powered.
Passive Subwoofers
Passive subwoofers are called "passive" because they need to be powered by an external
amplifier, in the same fashion as traditional loudspeakers. The important consideration
here is that since subwoofers need more power to reproduce low frequency sounds, your
amplifier or receiver needs to be able to output enough power to sustain bass effects
reproduced by the subwoofer without draining the power supply in your receiver or
amplifier. How much power depends on the requirements of the subwoofer speaker and
the size of the room (and how much bass you can stomach!).
Powered Subwoofers
To solve the problem of inadequate power or other related characteristics that may be
lacking in a receiver or amplifier, Powered Subwoofers (also referred to as Active
Subwoofers) are utilized. This type of subwoofer is a self-contained speaker/amplifier
configuration in which the characteristics of the amplifier and subwoofer are optimally
matched and are both encased in the same enclosure.
As a side benefit, all a powered subwoofer needs is a single cable connection from a
home theater receiver or surround sound preamp/processor line output (also referred to as
a subwoofer preamp output). This arrangement takes a lot of the power load away from a
receiver and allows the receiver's own amplifiers to power the mid-range and tweeter
speakers more easily.
Which is Better - Passive or Powered?
All other things being equal, whether a subwoofer is passive or powered isn't the
determining factor on how good the subwoofer is. However, Powered subwoofers are by
far the most commonly used as they have their own built-in amplifiers and are not
dependent on any amplifier limitations of another receiver or amplifier. This makes them
very easy to use with today's home theater receivers. All home theater receivers come
equipped with either one or two subwoofer pre-amp line outputs that are specifically
designed to connect to a powered subwoofer.
On the other hand, what you need to do in order to use a passive subwoofer is purchase a
dedicated subwoofer amplifier, which, in many cases could be more expensive than the
passive subwoofer you have. In other words, in most cases it would be more cost
effective to just buy a powered subwoofer in place of a Passive Subwoofer. If you choose
this option the subwoofer pre-out from a Home Theater Receiver would connect to the
external subwoofer amplifier's line-in connection, with the external amplifier's subwoofer
speaker connection(s) going to the speaker connections on the passive subwoofer.
The only other connection option is that is available for a passive subwoofer is that if the
passive subwoofer has in and out standard speaker connections, you could connect the
left and right speaker connections on a receiver or amplifier to the passive subwoofer and
then connect the left and right speaker output connections on the passive subwoofer to
your main left and right front speakers (see photo).
What happens in this setup is that the subwoofer will "strip off" the low frequencies
utilizing an internal crossover, which sends the midrange and high frequencies to the
additional speakers connected to the subwoofer's speaker outputs.
This type of setup would eliminate the need for an extra external amplifier just for the
passive subwoofer, but could put more strain on your receiver or amplifier because of the
demands for low frequency sound output.
The Exception to the Subwoofer Connection Rules
It is also important to note that many subwoofers have both line input AND speaker
connections (see example). If this is the case, the subwoofer is a Powered Subwoofer.
However, in this example, it is a subwoofer that can accept signals from either an
amplifier's speaker connections or an amplifier/home theater receiver subwoofer preamp
output connection.
This means that if you have an older home theater receiver or amplifier that does not have
a dedicated subwoofer pre-amp output connection, you can still use a powered subwoofer
that does have standard speaker connections available.
For more tips connecting this type of subwoofer, check out the article: How To Connect a
Subwoofer to a Receiver, Processor, or Amplifier by Gary Altunian, About.com Guide for
Stereos.
Final Take
When purchasing a subwoofer to use with your home theater, check to see if your home
theater, AV, or surround sound receiver has a subwoofer preamp output (often times
labeled Sub Pre-Out or Sub Out). If so, then you should use a powered subwoofer.
Also, if you just purchased a new home theater receiver, and have a left-over subwoofer
that originally came with a home theater-in-a-box system, check to see if that subwoofer
is actually a passive subwoofer. The giveaway is that it does not have a subwoofer line
input and only has speaker connections. If this is the case, you will need to purchase
either an additional amplifier to power the subwoofer, or, if the subwoofer has both
speaker input and speaker output connections, you may be able to connect the subwoofer
to the Left/Right main speaker outputs of the receiver and then connect your main left
and right speakers to the speaker connection outputs of the passive subwoofer.

Powered Subwoofers
The subwoofer has emerged as a critical component of the home theater system. Today's
DVDs and Blu-ray Discs contain low frequency information that add more impact to
those explosions and other special effects, as well as revealing those lower musical tones.
For the best results, you need a subwoofer. If you are considering adding a little punch to
your system, consider my moderately-priced powered subwoofer picks.
Paradigm DSP-3400 14-inch Powered Subwoofer

Photo © Robert Silva - Licensed to

About.com
The Paradigm DSP-3400 Powered Subwoofer delivers powerful, clear, tight, and
undistorted deep bass response from the combination of its large front facing 14-inch
driver, dual front ports, and 300 watts of continuous power supplied by its built-in
amplifier. In addition, the DSP-3400 offers an array of controls that allow the user to
tweak the performance parameters to specific room environments and listening tastes.
Short Review - Full Review - Photo Gallery - Official Product Page
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Martin Logan Dynamo 700w

Photo © Robert Silva - Licensed to

About.com
The Martin Logan Dynamo 700w is a compact 10-inch subwoofer that not only provides
strong power output (300 watts RMS - 600 watts Peak) and great low frequency
performance from a small package, but also provdes some innovative installation options.
With its removable feet, the Dynamo 700w can be repositioned as either a down-firing or
front-firing subwoofer. In addition, this sub features both wired and wireless connectivity
options with a home theater receiver or compatible sound bar.
Review - Photo Profile - Compare Prices
Also available, the Martin Logan Dynamo 1000w, which incorporates a 12-inch driver
and a 500 watt RMS/1,000 watt peak power output amplifier - Compare Prices
Definitive Technology Powered Subwoofers
Definitive Technology well known for their loudspeaker line, but they also make great
subwoofers well-suited for both music and home theater applications. Check out a great
list of choices from their impressive line-up that also includes some very compact
models, as well as tower speakers with powered subwoofers built-in.
Compare Prices
Velodyne Powered Subwoofers
Velodyne is well-known for their fine line of subwoofers. Check out a selection of great
subs for both small and large sized-rooms.
Compare Prices
Velodyne WiConnect WIC-10 Wireless Powered Subwoofer

Photo © Robert Silva - Licensed to

About.com
The Velodyne WiConnect WIC-10 is an affordable powered subwoofer that features a 10-
inch downfiring driver, rear port for extra bass extension, and a 125 watt amplifier. The
WIC-10 also offers very flexible installation options by providing either wired or wireless
connectivity with your home theater receiver. The WiConnect WIC-10 makes a good
addition to your home theater system if you need a replacement subwoofer or trying to
match a subwoofer with an existing set of speakers.
Review - Photo Profile - Compare Prices
Klipsch Powered Subwoofers
If you are looking for a great subwoofer to round out your home theater system without
breaking the bank, check out Klipsch powered subwoofers, including their innovative
triangular powered subs that are perfect for corner installation, as well as their digital
powered subwoofer line that can be customized for various types of musical or movie
response requirements.
Official Klipsch Subwoofer Page
SVS Powered Subwoofers

Photo © Robert Silva - Licensed to

About.com
One of leading independent subwoofer makers, SVS has a consistent quality product
across the board. No matter which unit you choose from their lineup, whether box-type or
cylindrical, you will experience deep, tight, bass deserving of any home theater system.
For an example, check out my SVS SB-1000 Front Firing 12-inch Ultra Compact
Powered Subwoofer Review - Official SVS Subwoofer Page
HSU Research Powered Subwoofers
A great subwoofer doesn't have to cost a fortune. Dr Poh Ser Hsu designs and makes
world class subwoofers for home theater and audio applications at affordable prices. Be
sure to check out the VTF or STF Series. Selecting a "best" from this group is impossible
-- they are all great values.
Official HSU Research Subwoofer Page

What the .1 Means in 5.1, 6.1, or 7.1

Channel Surround Sound
Surround Sound and .1
One of the concepts that confuses consumers is what the terms 5.1, 6.1, and 7.1 mean
with regards to surround sound and home theater receiver specifications.
The first part of the number refers to the number of channels that are present in a
soundtrack or the number of channels that a Home Theater Receiver can provide. These
channels reproduce a full range of audio frequencies, from high frequencies to normal
bass response. This number is usually 5, 6, or 7.
In addition, another channel is also present, which only reproduces the extreme low
frequencies, referred to as Low Frequency Effects (LFE). This channel is designated
as .1, due to the fact that only a portion of the audio frequency spectrum is reproduced. In
addition, this LFE channel requires the use of specialized speaker, called a Subwoofer,
that designed only to reproduce extreme low frequencies, and cuts-off all other
frequencies above a certain point, usually 100HZ to 200HZ or above.
So, next time you see the terms Dolby Digital 5.1, Dolby Digital EX (6.1), DTS 5.1,
DTS-ES (6.1), you will know what the terms are referring to.
NOTE: You will also run into some home theater receivers that are labeled as having 7.2,
10.2, or even 11.2 channels. All the .2 designation means is that these receivers have two
subwoofer outputs. You don't have to use both, but it may come in handy if you have a
very large room, or are using a subwoofer with lower power output that you desire.
For a more detailed explanation of Surround Sound and what the terms Dolby Digital,
Dolby Digital EX, and DTS mean, check out my resource article: The History and Basics
of Surround Sound.
 Secrets of Amplifier and Speaker Power
Requirements Revealed
As audio/video hobbyists, most of us grew up thinking that if we have an amplifier with
50 watts of rated output power into 8-ohm speakers, and that combination produces
reasonably clean and loud music, then by doubling the amplifier power to 100 watts per
channel, the system would then play twice as loud. Many readers likely still believe that.
Not so.
Although it's not the easiest thing to comprehend, doubling the amplifier power does not
double the loudness. In the above example, the sound from the speakers would not be
"twice as loud"; it would only be "a little louder," an increase of 3 decibels. How loud is
that? Hearing tests with large groups of people have revealed that a one-decibel (1 dB)
change in loudness is approximately the smallest audible step that the average listener
can detect, so an increase of 3 dB most listeners term "slightly louder."
So why doesn't that 100-watt amplifier always sound twice as loud? Because the
acoustic decibel--the decibel (dB) being the unit of measurement used worldwide to
quantify the acoustic loudness of sound--has a peculiar relationship to amplifier power
output measured in electrical watts. That relationship is called "logarithmic." If that word
gives you an instant headache (nightmares of high-school math), then here's a simpler
explanation:
If a sound gets louder by 3 decibels or "slightly louder," it takes twice as much
electrical power from your receiver or amp to produce that modest increase.
Therefore, a 100-watt amplifier will produce sound only slightly louder than a 50-watt amplifier.
Incidentally, if you'd like a
kind of immortality, be
terribly clever and work
out a system of
measurement. It may be
named after you. The
"decibel," one tenth of a
bel and named for

Alexander
Graham Bell, recognizes
his contributions to the
understanding of sound.
Likewise, we have to
thank James Watt, Georg
Simon Ohm, and Heinrich
Hertz for their
contributions to the
industry. And then there's
the Lofft, a measurement
of neighbors' tolerance to
testing new speaker
systems . . .
So far, so good. But what if it's party time, and you're listening to music "very loud," a
level defined as about 90 dB Sound Pressure Level (SPL), and your speakers are
gobbling up swings of 15 to 20 watts per channel on those musical peaks.
Drink in hand, you advance to the volume control on your receiver thinking, "I'll just
crank this up to make the music twice as loud," and you turn up the volume control until
there's a 10 dB increase in the sound level. Now your party-time goal of "twice as loud"
will make huge electrical demands on your nice little multi-channel receiver or power
amp. The receiver must deliver ten times as much power to double the subjective
loudness. Between 6 dB and 10 dB is double the volume level, where 6 dB is four times
the power and 10 dB is 10 times the power. In the aforementioned example, the amp must
produce 150 to 200 watts per channel for those peaks in loudness. Therefore, every 10-
dB increase in acoustic loudness--from 80 dB to 90 dB, or 90 dB to 100 dB--requires
ten times as much electrical power in watts.
That's all very well if you have a monster amplifier or multi-channel A/V receiver with
huge reserves of power output (most of us don't). If not, watch out. Your receiver or
amp may "clip" or distort (or both), which will put a clamp on the output of the amp.
When you push your amplifier into overload or "clipping," several things may happen.
First, the top and bottom of the waveforms (representing the audio signals) are clipped
off, generating distortion. Next, the amplifier's protection circuits are activated, removing
those portions of the signal that are causing the overload, generating distortion. And
finally, the amplifier's power supply may fluctuate according to the demands of the music
signals.
Not everyone is affected by this scenario, of course. Some people (increasingly few, it
seems) don't listen to loud music. They like background levels, and with average
speakers, background levels demand 1 watt or less of amplifier power. Or they may have
very efficient speakers (Klipsch, Cerwin-Vega, Tannoy, and the like) that will play
extremely loud using modest amplifiers, the trade-off being a very large degradation in
tonal accuracy, a definite harshness, and a complete loss of off-axis performance that
accompanies horn-loaded designs. But in many situations, speakers will be damaged and
distorted sound will offend many ears.
No discussion of decibels, acoustic loudness, and electrical watts is complete without an
explanation of loudspeaker "sensitivity." (Another way to define a speaker's sensitivity is
to look at how efficiently the speaker converts electrical power, in watts, to acoustic
sound output in decibels.) Let it be said in a general way that speakers are not very
efficient or sensitive devices. They need a lot of electrical power input to produce
relatively little acoustic output. Nevertheless, speakers do vary quite a bit in sensitivity.
To determine a speaker's sensitivity, we feed the speaker with 1 watt of amplifier power,
using a test signal of pink noise, and measure in decibels how loud the sound is at a
distance of 1 meter (about 3 feet). A lot of domestic hi-fi speakers measure in at about 89
or 90 dB SPL at 1 meter. Larger speakers, with bigger woofers and more drivers,
typically produce greater acoustic output; smaller bookshelf models have to work harder,
and their output is typically less, often between 86 and 88 dB SPL at 1 meter.
Placing the speaker in a room helps (the walls, ceiling, and floor reflect and reinforce
the speaker's sound), adding about 4 dB to its output. For example, a speaker like
Axiom's M80ti has a measured sensitivity in an anechoic chamber of 91 dB SPL at 1 watt
at 1 meter. But putting the M80ti in a room raises its sensitivity rating to 95 dB SPL at 1
watt, 1 meter. A 95-dB sound level happens to be "very loud," as most of us would
subjectively describe it. And it is--from 3 feet (1 meter) in front of the speaker. But let's
move our listening seat back twice as far, to 6 feet. Guess what happens? We instinctively
know that sound gets weaker as the distance from the source is increased, but by how
much? A formula called the "inverse square law" tells us that when the distance
from the source is doubled, the sound pressure weakens by 6 dB. Among sound
engineers, there's a common saying: "6 dB per distance double." So at a 6-ft. distance, the
M80ti is now producing 89 dB. Now let's double that distance again to 12 feet, a fairly
common listening distance. The speaker now produces 83 dB, which isn't all that loud at
all. And if you sat 24 feet away, a not uncommon distance in big rooms, the speaker
would produce 77 dB SPL.
But what about stereo, I hear you shout. Here's another oddity of loudness and the
decibel. When one speaker is producing a level of 90 dB, adding a second speaker
playing at the same level only increases the overall loudness by 3 dB! (The loudness does
not double!). So the two speakers in stereo produce a loudness level of 93 dB.
So adding a second M80ti will raise the loudness at 12 feet from 83 dB to 86 dB. And
don't forget we're still using 1 watt of amplifier power output into Axiom's most sensitive
speaker. But how loud are real-life instruments, orchestras and rock bands? Now,
while 86 dB SPL is "fairly loud," it's not nearly as loud as what you might hear from a
good seat at an actual rock concert or from an orchestra or pianist in a concert hall. A solo
grand piano can reach peak levels of 109 dB SPL, a full orchestra and chorus in a concert
hall will measure 106 dB, and a rock group, 120 dB SPL. Now let's try and get our peak
speaker sound levels to 96 dB, "twice as loud" as our 86-dB listening level. That isn't that
difficult because right now we're only using 1 watt per channel to drive the M80ti's to 86
dB. So we'll need ten times as much power, or 10 watts, to reach 96 dB. Big deal. We've
got lots more.
But things begin to change, and rather dramatically. Let's push the M80ti's to what we
might experience from a solo grand piano, 109 dB. We're at 96 dB with 10 watts per
channel. Let's go to 106 dB. So that requires 10 x 10, or 100 watts. Close, but not quite
there yet. Just 3 dB more. Remember, we have to double the power for a 3-dB increase in
sound level. So 100 watts becomes 200 watts. Yikes! Our receiver has only 110 watts
maximum output! We've run out of amplifier power! And what about the rock concert?
Let's lower our expectations and aim for 119 dB. Going from 109 dB SPL, which needs
200 watts per channel, to 119 dB SPL (get out your ear plugs) is another 10-dB jump
and--you do the math--that requires 10 x 200, or 2,000 watts per channel!
From all this you can see the huge power requirements inherent in reproducing real-
life acoustic sound levels in average or big rooms. The M80ti's are tested to levels of
1,200 watts of input power so they come very close. But the truth is that if we are seeking
real-life acoustic sound levels in our listening rooms, there's a very persuasive argument
for very large, powerful amplifiers. And if your speakers are less sensitive (and many
are), then the power demands rise even more dramatically. Sizeable rooms and greater
listening distances will also increase power demands tremendously.
And what many of us don't realize until we hear it, is that clean undistorted loud sound
often does not sound that "loud." The key here is that in most or our home listening, there
are small amounts of distortion caused by a lack of dynamic headroom (but more on that
next month). It's the distortion that makes it sound "loud" in a domestic setting. To
remove those distortions and increase dynamic headroom relates to even more power.
We've become accustomed to accepting some distortion with our reproduced music,
because all amplifier's distortion ratings gradually increase as they approach their output
limits or slightly clip the audio signals. When that happens, we turn down the volume,
because distortion starts to intrude on our listening pleasure, and it sounds "too loud."
The lesson in all this is that you can never have too much power, and that big amplifiers
rarely damage speakers. Little amplifiers driven into clipping burn out speakers. In the
scheme of high fidelity, that last barrier to realism is having enough power and being able
to approximate real-life loudness levels.

What watt amplifier will power a 1200

watt subwoofer?
Answer:

1200W is the either the MAXIMUM power that a passive subwoofer (driver only) can
HANDLE or the power rating of the amplifier on an active subwoofer (amp built-in).

If passive, you can use pretty much any amplifier as long as it's 1200W or under. You can
even use an amp that is > 1200W -- you just can't push it past 1200W or you risk
damaging your sub. The relative loudness at a given dial on the volume knob is
dependent on the sensitivity of the sub (measured in dB). Also bear in mind that power
varies with resistance so an amp that pushes 1200W @ 4 ohm will not be able to deliver
1200W to an 8 ohm speaker -- often times it can only deliver 1/2 the power (i.e. 600W @
8 ohm in this case).

If active, well the sub already has an amp so you just need to feed it a line-level or
speaker-level signal.

Either way if you're talking about your car or home I guarantee you won't be pushing
ANYWHERE near 1200W -- that's window-shattering, deafening loud. Just to put things
into perspective, I use an amp that's rated at 10 W into 8 ohms and never turn the volume
dial past 1 or 2 o'clock, which means the amp is never delivering more than maybe 6 or 7
watts. If you can get remotely close to 1200W that means the amp is so inefficient that
it'll probably be melting and/or burning down your house/car at that point.
Audio Power Amplifier Power Rating Mysteries Explained

(Once and For All!)

Rev (F) 05 April 2007
(c) 2000-2007 By Joe Roberts
Why this article?
I receive quite a number of questions via e-mail regarding power ratings on amplifiers as well as many
questions on what clipping is and why it can be bad for speakers. Other questions include things like "What
is instantaneous power, peak power, RMS power?" "What is a crest factor and how does it relate to
headroom?" I also sense that there is a fair amount of mis-information circulating within the consumer (and
sometimes the pro audio) audio community. When I was a teen just learning all of this stuff, there was no
one I knew that really knew this material... there was no one I could ask questions of. It took me years to
finally figure it all out. This article will attempt to make it easy for the reader to clear up the mysteries of
power ratings dealing with audio power amplifiers. This article is quite extensive and will cover many
topics in a fair amount of detail. Plots and graphs will be used to illustrate various concepts. There is
necessarily some math in this article, but please do not be too frightened by this. I will try to keep things as
readable as possible!
Assumptions: For most of this article some assumptions must be made. For the power amplifier we will
consider, the power supply will have "rails" that do not sag with load, and we will treat speaker loads as
purely resistive (otherwise things will get too complicated too early). Later sections will cover "real world"
amplifiers and what performance can suffer as a result.
Amplifier Basics: Before we explore what power ratings are all about we need to cover some basics about
amplifiers. The first thing to understand is that an amplifier is a device that takes a weak electrical signal
and makes it stronger. In the case of an audio amplifier we wish to take the weak signal from the source
(CD player, tuner, tape deck, etc.) and make it strong enough to drive one or more loudspeakers. The ideal
amplifier will do nothing to the input signal other than make it stronger. In a real world amplifier, the signal
does get stronger (amplified) but because no amplifier is perfect there are other undesired characteristics
that appear in the output signal (noise and distortion). In the very best amplifiers these undesired
characteristics are quite small (but they are not zero). Also, all amplifiers have limitations as to how much
power they can put out. If you try to get more power out of an amplifier than it was designed to provide, the
output will rapidly become very distorted because the amplifier will go into a condition commonly referred
to as "clipping".
For this article to make sense, the reader must have at least a minimal knowledge of the basic topology of
an audio amplifier. There are a number of classes of amplifiers used for audio (such as A, B, AB, D, H and
G to name some of the more common ones). The class of an amplifier refers to the method in which the
components within operate (you may sometimes hear this referred to the "bias" type of the amplifier). We
will discuss the most common form used for audio amplifiers; that is, class AB. The various classes used
for amplifier design determine the level of distortion, efficiency, heat dissipation, etc. You do not need an
understanding of the details of the class of an amplifier to gain an understanding of power ratings. Consider
the diagram below:

Figure 1. High Level Block Diagram of Class AB Amplifier

Figure 1 shows a very high level diagram of the internal workings of a typical class AB audio power
amplifier. Notice that there is a power supply; this component converts the 120 VAC voltage from a wall
socket into a DC voltage. The power supply creates two DC voltages, equal in magnitude but opposite in
polarity. These two voltages are often referred to as the rails of the power supply. We will refer to this term
often within this article! Note that the power supply rails are fed directly to the output stage of the
amplifier. The amplifier also has an input stage; this is the portion of the amp that receives and conditions
the input signal (it is where the level controls would be located). It too receives its operating power from
the power supply, but input stages of amplifiers often use a lower voltage that is more tightly regulated.
Low level circuits at the input stage are often of class A design in order to maintain the highest possible
fidelity. The output stage of the amplifier is what drives the speaker(s).
Now let us consider the power supply once again. Specifically, we will consider a hypothetical amplifier
that has voltage rails of +40 VDC and -40 VDC. These voltages represent the largest possible voltage that
the amplifier may apply to a speaker (either +40 volts or -40 volts for amplifiers that are not running in
what is known as "bridged" mode). In most real world amplifiers, the actual maximum output voltage to the
speakers is slightly less than the value of voltage present on the rails. For this example we assume that the
maximum output can reach the rail voltage. For purposes of example, let us assume this amplifier has a
voltage gain of 20 (with the level control turned all the way up). This means that any signal applied to the
input of this amplifier will be amplified by a factor of 20; if we apply one volt to the input terminals we
should get 20 volts out. Similarly, if we apply 0.1 volts at the input we should get 2 volts out. But what
happens if we apply 5 volts to the input? In theory the amplifier would try to generate 100 volts out...
however we already stated that the absolute maximum voltage that the amplifier can generate is 40 volts.
So, for this amplifier, any voltage at the input that exceeds 2 volts will result in a 40 volt output. Anything
more at the input will still result in a 40 volt output. If we exceed 2 volts at the input (for the amplifier of
this example) we will "clip" the output of the amplifier (more on this topic shortly). Note that in this
paragraph we are talking about voltage applied to a speaker (not power); the voltage and the speaker
impedance determines how much power (wattage) will be delivered to the speaker (by use of ohm's law).
So, here are the important things to understand from this section: (1) Know what "rails" are pertaining to
amplifier power supplies and (2)Understand that the voltage output of an amplifier (the voltage that gets
applied to the speaker) cannot exceed the voltage of its power supply rails. Read number 2 again to drive
the point home! Note that we are talking about the amplifier's output voltage here, not its output power!
Since the rail voltages are known, we can calculate the maximum theoretical power that the amplifier can
put out (more on this later, it's more complicated than just the rail voltage).
Clipping Illustrated: In this section we will illustrate what happens when an output signal from an
amplifier is clipped. For this section we will assume that the input signal to the amplifier is a sine wave (a
signal that contains only one frequency). Let's assume we are using the amplifier described above (voltage
gain of 20 and power supply rails of +/- 40 VDC). Recall that the voltage output of this amplifier cannot
exceed +/- 40 volts under any condition. If we apply an input signal of 1 volt (peak) we get a 20 volt (peak)
signal at the output of the amplifier. Figure 1 below shows the output signal:

Figure 2.
Figure 2 above shows what the output signal of the amplifier would look like if viewed using an
oscilloscope. The blue line represents the output signal, note that it is clean and undistorted. This is because
the output signal is well below the limits of the amplifier (+/- 40 VDC in this example).
Now let's see what things look like if we increase the input signal level to 2 volts (peak). See figure 2
below:

Figure 3.
Figure 3 shows that the output signal is at its maximum undistorted value. The peak of the signal is just
touching the +/- 40 volt level, the values of the power supply rails. This signal is still clean and undistorted,
however it is the maximum clean signal that is possible for this amplifier. Any increase in the input signal
amplitude will result in clipping of the output signal. Now consider what happens if we increase the input
signal amplitude to 3 volts. The output of the amplifier would try to be +/- 60 volts peak, but we know that
the amplifier has a maximum output of +/- 40 volts. Figure 3 below shows the resulting output:
 Figure 4.
Figure 4 clearly shows that the output signal is no longer a clean sine wave because the peaks of the
waveform have been chopped off or "clipped". Had we used a more powerful amplifier (one with larger
voltage rails), the signal peaks we would be able to achieve before clipping would be larger.
If we were to apply the signal illustrated in Figure 4 to a speaker, the sound would take on a "harsh" or
"raspy" sound. The sound quality will deteriorate further the more the signal is clipped. In extreme cases, a
sine wave will approach the shape of a square wave when clipped. Clipping introduces a large number of
what are known as harmonic components to a signal (and it also increases what is known as the RMS level
of the signal, something that can lead to blown speakers). It is the harmonics that cause the sound to harsh.
For those with a deeper knowledge, the frequency content of any non sinusoidal waveform consists of a
larger number of individual sine waves of varying magnitudes and phases. A square wave contains
fundamental frequency plus an infinite number of harmonically related sine waves, all lined up with a
particular phase relationship. To the human ear, waveforms that are rich in harmonics have a harsher tone.
Anyone who has cranked up a car radio or boom box to the maximum level has certainly heard the results
of severe clipping (terrible sound that may be barely intelligible).
For now we are just showing what clipping actually looks like. Clipping not only has a detrimental effect
on the sound quality, it can also be damaging to speakers. We will cover the reasons for this a bit later on.
Now that we have a basic understanding of what various output waveforms look like (clean and clipped),
we can move into a discussion of amplifier power ratings.
Power: Power is a measure of how much energy is dissipated per unit time. Audio power amplifiers are
rated as to how many watts they can put out. For example, an amplifier that is used in a home stereo may
have a rating of 100 watts per channel. The mathematical equation for power is as follows:
P=VI = I2R=V2/R
Where
 P = power in watts,
 I = current in amps,
 R= resistance in ohms,
 V = voltage in volts.
Let's look at a simple circuit to illustrate power (this is not an amplifier circuit but it is where we will start
to keep things as simple as possible):

In this case we have a 12 volt battery with a 100 ohm resistor as a load. A current (I) flows in the load as
shown. From Ohm's Law (V=IR), we can easily calculate that the current flowing through the load resistor
is equal to 12/100 = 0.12 amps. We can then determine the power dissipated by the load resistor by
dropping the values above into the equations for power (any of the three of them will work). If we do this
we find that the power is equal to 1.44 watts.

Amplifier Power Ratings: There are many terms used to describe the output capability of an amplifier.
Some of the terms you may have seen: peak power, RMS power, music power and instantaneous power (to
name a few of the more common ones). What the heck does all of this mean? We will cover most of the
major terms and hopefully take the mystery out of all of this. The many ways in which power can be stated
gives manufacturers a great opportunity to mislead unsuspecting people. Most reputable manufacturers will
not intentionally try to mislead their (potential) customers. However, if you have an understanding of the
various terms and how they relate you will be much better able to sort out the truth from the fiction when it
comes to power ratings.
 Shop New Items on SALE at Guitar Center!
Instantaneous Power: As you are probably aware, audio amplifiers put out a voltage (to a speaker) that is
constantly changing. A music signal can be called AC (alternating current), but it is not the same as a sine
wave (unless you happen to be playing test tones through your system). In the section above where we
covered the basics of power, we used a 12 volt (DC) battery for keeping the explanation as simple as
possible. Calculating the power output for an AC signal is somewhat more complicated because of the
nature of the AC signal. An AC signal is constantly changing in amplitude with time, and therefore (per
Ohm's law) the power delivered to the load (a loudspeaker in the case of an amplifier) is also constantly
changing. At any given moment an amplifier will have a particular output voltage, and the power delivered
to the load will be a function of that voltage. Let's consider the case where a sine wave signal is applied to
the input of the amplifier (resulting in a sine wave of larger amplitude at the amplifier output). We use a
sine wave because a sine wave signal is the simplest AC signal possible and it is used widely in testing and
rating of amplifiers. In this case the load will be an 8 ohm speaker (speaker assumed to be equivalent to an
8 ohm resistor) and the sine wave output of the amplifier will be at maximum output before clipping.
Consider Figure 1 below. This figure shows a plot of the output waveform applied to the speaker (as would
be visible on an oscilloscope):

Figure 4.
Ohm's law states that power is equal to the voltage squared divided by the resistance of the load. As should
be obvious in Figure 4 above, the voltage of a sine wave signal varies with time. For any given point (A, B,
C, or D) on the waveform, the output signal has a specific voltage value (and therefore a particular level of
power that will be delivered to a load). The power at any given point on the waveform is known as the
instantaneous power. Do note that the voltage level for audio signals is typically changing at a very rapid
rate (from tens of times per second to many thousands of times per second). As such, the power being
delivered to the load is also changing very rapidly! However, due to the "thermal mass" associated with
most speaker voice coils, the rise and fall of the temperature is not noticeable unless an extremely low
frequency is used. Regardless, when the waveform signal is equal to zero volts the power being delivered to
the load at that instant will also be zero! This is true despite music being played VERY loudly. The
speaker's voice coil could still be very hot however as it takes time for it to cool (the same way your electric
oven does not go "cold" the second you turn it off). At the moment of zero power the speakers voice coil is
not getting any hotter, that is certain. Another notable point: the power delivered to the load at the positive
peak of the waveform is the maximum possible for the amplifier, and it is the same value as the power
delivered to the load at the negative peak of the waveform! This second fact can be deduced by looking at
the equation P=V2/R . The squaring of the voltage component will make the product positive regardless of
the polarity (plus or minus) of the signal. The table below shows the power level (into an 8 ohm load) for
the points on the waveform above is as follows:
Point Voltage Power (into 8 Ohm load)

A 40 200 watts

B 20 50 watts

C 0 0 watts

D -30 112.5 watts

As can be seen from the table, a doubling of the voltage results in the power going up by a factor of four!
So, what is the power rating of this amplifier? 200 watts? If you guessed 200 watts it is unfortunately not
the correct answer (although it is not absolutely wrong either). This amplifier would be rated as a 100 watt
amplifier if it were sold in today's market (more on how we arrive at this later). The main point to be gained
about instantaneous power: understand that the power output of an amplifier is always changing (in
accordance with the output signal amplitude). This should be obvious... when a loud drum beat is played,
the power meters on an amplifier show a large burst of power for those beats (and low power for the period
in between)... when a violin solo is playing, the power level is quite low compared to when the entire
orchestra is playing!
Peak Power: The peak power output of an amplifier is just that: the maximum amount of power that can
be delivered to a load (this is usually for a very short instant of time). For the amplifier we have been using
in our example, the peak power (for an 8 ohm load) is 200 watts. This amount of power is delivered to the 8
ohm load at the instant when the output voltage of the amplifier is at +40 volts (or also at -40 volts). In real
world spec sheets, peak power (if listed) is basically the maximum power output that the amp can deliver
(into a particular load, usually 8 ohms) for a very short period of time. Basically, anything above this level
would result in clipping of the output signal.
RMS Power: The sections above showed that the power output of an amplifier changes all the time
depending on the signal level being applied to the load (speaker). So how does one determine the rating of
a power amplifier? Why do most amplifier spec sheets today rate their output in "watts RMS"?
Let's go back to the case where a simple DC supply (a battery for example) was used with a 100 ohm
resistor load. In this case the voltage is always the same (12 volts) so the power to the load is always the
same (in the example we used the power is 1.44 watts). As a result of dissipating 1.44 watts, the load
resistor will heat up to a certain temperature. Now let's say that we have a 40 volt battery and we connect
an 8 ohm load across it. The resulting power delivered to the load will be 200 watts (and the resistor will
get quite hot)! But what happens if we apply a sine wave with peak value of +/- 40 volts to the same load
instead of the DC signal? The resistor will dissipate power, however the power delivered to the resistor will
be less than the amount of power delivered for the case where the 40 volt DC signal is used. This is where
we bring the term "RMS" into play. RMS stands for "Root Mean Squared". When a sine wave signal is
used to supply power to a load, the voltage of the sine wave necessary to result in the delivery of the same
amount of power as the DC voltage must have a higher peak amplitude compared to the DC source capable
of delivering the same power. For a sine wave, we must apply what is called a 40 volt RMS signal to the
load in order to deliver the same amount of power to the load as the 40 volt DC supply. For a sine wave
signal, the peak voltage necessary to accomplish this is equal to the DC voltage times the square root of 2,
or about 1.414 times the DC voltage value. In our case we have a 40 volt DC signal, so the sine wave signal
necessary to deliver the same power in the load must have a peak value of 40 x 1.414 = 56.5 volts!
Another way to think of it is like this: The 40 volt DC signal resulted in the delivery of a constant power
level to the load at all times. However, the power dissipated in the load when driven by a 40 volt peak AC
waveform (sine wave) is only equal to that of the DC waveform for a short (most of the time the voltage
from the sine wave signal is less). Therefore, most of the time the power being delivered to the load is also
less, and as a result the load will not get as hot. Therefore, in order to get the same amount of power
delivered to the load as the 40 volt DC supply, the sine wave signal amplitude must be increased (by the
1.414 factor mentioned). Note that the 1.414 factor is valid only for sine waves, the reasons for this are
beyond the scope of this article at this time. Here's one more way to look at it. In figure 4 above, the sine
wave signal shown is peaking at 40 volts; this signal, if applied to an 8 ohm speaker, would deliver 100
watts to the speaker. The equivalent DC voltage necessary to deliver 100 watts to an 8 ohm load would be
28.3 volts!
For those who have more mathematical background: power delivered to a particular load (we've been using
8 ohms) is proportional to the voltage squared. For non DC voltages, we must integrate the area under the
voltage squared curve. For DC this is easy because the voltage curve is constant (not changing). For a sine
wave, we square the voltage (which among other things makes it all positive) and then we have to integrate
under the curve. For a DC voltage of 40 volts, and a sine wave that peaks at 40 volts, it will be very clear
(upon plotting the waveform of the sine voltage squared) that the area under the curve is definitely less than
that for the DC voltage. Someday when I have time I'll generate some plots to illustrate this...
So... where does this bring us? As you have probably seen in amplifier spec sheets, manufacturers rate the
output of their amplifiers in watts RMS. For example, let's consider the vintage amplifier Kenwood Model
KA-9100. This amplifier is rated to put out 90 watts RMS per channel into an 8 ohm load. Technically
speaking, the term "RMS" is not defined when referring to power (watts)! RMS is a valid term when
referred to voltage or current, but not power! Watts are watts, period! Despite the term "watts RMS" being
an incorrect term, it stuck with the community and has become the accepted way to rate an amplifier's
output. The
reason that audio amplifiers are rated in "watts RMS" is
because they are rated to deliver that amount of power using a
sine wave signal. Because amplifiers are rated this way, their peak power output will be twice
the RMS rating. So, for the Kenwood KA-9100 (which is rated to deliver 90 watts RMS into 8 ohms), the
peak power is 180 watts. Most amplifiers cannot sustain output at their peak capability for too long (and the
characteristic of most music signals is such that this is not necessary anyway). So, despite "watts RMS"
being a technically invalid term, it is used with audio amplifiers because of the sine wave signals that are
used to determine their power output specifications.
Question: What are (at minimum) the rail voltages necessary for an amplifier to deliver 90 watts into an 8
ohm load? It is not too hard to figure this out. We simply look at the equations for power:
P=VI = I2R=V2/R
In this case, there are two known items: power (90 watts) and resistance (of the speaker, 8 ohms). Plugging
these numbers into the equation (P=V2/R) yields a value for V of 26.83 volts. Is this the answer to the rail
voltage question? No... REMEMBER, if we applied 26.83 volts of DC (note: DC) across an 8 ohm speaker
the power to the speaker would in fact be 90 watts. However, amplifiers are rated using sine wave input
signals, and (as described above) we need to apply more voltage to a load (for a sine wave) in order to get
the same amount of power that would be delivered by a DC voltage. For sine waves, the multiplication
factor is 1.414. So, if we take the voltage of 26.83 and multiply it by 1.414 (the square root of 2), we get a
value of 37.94 volts. This value is the absolute minimum rail voltage needed for an amplifier to deliver 90
watts (with a sine wave signal) to a load! Had we mistakenly determined that 26.83 was the correct rail
voltage, the amplifier would begin to clip as the output tried to exceed 45 watts of output. Note that 45
watts is exactly one half of the 90 watt value. This shows that by increasing the rail voltage by a factor of
1.414 results in the amplifier having twice as much output capability! This can also be deduced by looking
at the equations for power.
To summarize, amplifiers use the technically incorrect term "watts RMS" in their output ratings because
amplifiers (those used for audio anyway) are rated using sine wave signals. Music is not a simple sine
wave, and it is not DC... however a sine wave is more representative of a music signal than DC is. So, the
standard practice in use today is for manufacturers to rate amplifiers in "watts RMS". Be aware that watts
are watts, there is technically no such thing as watts RMS.
Music Power: This is a term from the "old days". Back in the 1960's and early 1970's, there were no
standardized conventions for stating power ratings of amplifiers. As such, manufacturers would (naturally)
use a method that resulted in the largest numerical power rating possible (a marketing ploy). The result was
a power rating that stretched the limits of truth. Basically what was done was this: the voltage from one
amplifier rail to the other (under load) was measured. This voltage was then used to calculate a power
rating using a resistor of the same value used for the load (often 8 ohms). The resulting number loosely
equated to "peak to peak" maximum power (a term that for all practical purposes is meaningless). Such
ratings are no longer in use today as they are basically very misleading and suggest that an amplifier is
MUCH stronger than it really is. As an example, take the rail voltage necessary for the KA-9100. We
calculated it to be an absolute minimum of 37.94 volts (we'll round that to 38 to make things easier). 38+38
= 76; if we apply 76 volts across an 8 ohm speaker (assuming it is 8 ohms resistive) the power delivered to
the speaker will be 722 watts! In other words, it exaggerates the true power of the amplifier by a factor of 8.
Some amplifiers were "more conservatively" rated back then; they only used a single rail voltage (and for
the case of the KA-9100 it would be rated at 180 watts by that method: double its real output). This rating is
really the "instantaneous peak" power rating. While not meaningless, it is misleading at minimum. Music
power is no longer used today, however if you work with vintage amplifiers (and can get the spec sheets for
them) you may see some very optimistic power ratings!
AC Power Consumption: On the back of many amplifiers you may see a label that shows (for example)
"1000 watts". This is NOT the output power capability of the amplifier. This is (in almost all cases) the
amount of AC power that the amplifier requires from the 120 VAC wall socket for nominal operation at full
output using normal music signals. The amplifier's output power must always be less than this AC power
requirement specification, because the amplifier cannot deliver more (or even as much ) power that it takes
from the wall socket. All amplifiers generate heat to some degree, this is basically wasted electrical energy.
Typically, if an amplifier takes 1000 watts from the wall, it might have a power rating of approximately 300
watts per channel (it really depends on the class of the amplifier's circuit class). I see vintage amplifiers on
Ebay all the time... and those who are offering them for sale (who admit they know little or nothing about
amplifiers) often make the mistake of listing the output power of the amplifier by stating the power
requirement that is listed on the back of the amp.

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Why is clipping bad for speakers? There are several major reasons why speakers driven by an amplifier
operating in clipping can be bad: signal compression and more energy being generated in the high
frequency range (due to increased harmonic distortion).
Anyone who has ever looked at a music signal on an oscilloscope knows that the signal is rapidly changing
(meaning that it has periods of large amplitude, then low, etc). This signal, when applied to a speaker, of
course generates sound, but it also generates heat in the speaker's voice coil. The voice coil gets hot during
the loud passages of music, but has time to cool down somewhat during the softer passages. The louder the
music is played, the hotter the speaker's voice coil will become. If it becomes too hot, the speaker can fail
(due to the voice coil melting or burning). Basically what we are saying here is that if the music signal is
such that the speaker's voice coil cannot maintain a safe temperature, the speaker is at risk of failure. So
why does clipping exacerbate this situation?
When an amplifier is driven into clipping, the average level of the music signal increases. During clipping,
the loud parts of the music have already exceeded what the amplifier can output cleanly, but the softer parts
have not. However, the softer parts have become louder, and the key point here is that by operating the
amplifier in a clipping condition we have made the difference between the loud and soft passages of the
music less than it was when the amp was running clean. Basically, we have "compressed" the signal (and
such signals put more of a strain on a speaker). The result is that the speaker's voice coil cannot cool off (as
much) between passages of loud and soft music.
When an amplifier is driven into extreme clipping, the amplifier will put out significantly more power than
its RMS wattage rating (assuming the amp has a decent power supply). How can this be? Recall (from
discussions above) that for an amplifier to put out (for example) 100 watts with a sine wave signal, its
power supply voltage rails must be higher than that necessary to generate 100 watts if a pure DC voltage
was used. When an amplifier is driven into extreme clipping, the signal will look approximately like a
"square wave". The power output resulting from a square wave signal is twice that for a sine wave
(assuming that the peak value of the sine wave is the same as the value of the rail voltage when the square
wave is being generated, as would be the case for a high quality amplifier). In most real world amplifiers
the power supply will not be able to handle putting out twice the rated power, and in actuality the power
delivered during extreme clipping will not be twice the rated power of the amplifier. However, the output
will likely be substantially more than the rated output (perhaps by 50% or more)! So, even with real world
amplifiers, extreme clipping can ultimately send a lot more power to a speaker than you might otherwise
expect. Adding to the problem, this kind of output is continuous power (it gives the speaker's voice coil no
chance to cool down because there are no soft passages in music that is extremely clipped).
Let's say you have a high quality amplifier that is rated for 200 watts per channel (RMS) and you have
speakers that are rated to handle 200 watts (each). Can you blow the speakers? Possibly! If you operate the
amp with only occasional clipping there should be no problems. If you run the amp into extreme clipping,
the speakers will actually be attempting to handle more like 300 - 400 watts each. This could wreck them
unless they are very tough and conservatively rated! Now supposed you have speakers that are rated to take
600 watts each. Can this amp wreck them? Probably not. You could run the amp in a condition of extreme
clipping and the speaker would sit there and take it with no issues. However, the sound quality would be so
absolutely hideous that no reasonable person would consider hanging around! It should be noted that only
the very best amps will be capable of putting out twice their rated power when driven to extreme clipping
(be aware that some do exist!). This is because (usually) the power supply cannot supply the necessary
power to provide twice the rated output. Nevertheless, most amps will put out considerably more power
(compared to their rated output) when driven into extreme clipping. So, signal compression resulting from
clipping can not only be very detrimental to sound quality, it can be very bad for speakers. However, signal
compression is not the only factor to deal with when operating an amplifier into clipping (read on)!
There is another consequence of operating an amplifier into clipping: high frequency harmonics will be
generated. Any time a signal is clipped, the waveform's spectrum (frequency components) will be altered.
The result is that more high frequency energy is generated (as compared to what was present in the signal
to begin with). The crossover in the speaker system will direct the higher frequency energy to the midrange
and tweeter speakers, and these (especially tweeters) will be more susceptible to burnout. The risk of
damage to the speakers depends on the characteristic of the music (does it have lots of high frequency
energy to begin with?), to what degree of clipping is occurring and how conservatively the speakers are
rated. It is not uncommon to blow tweeters when operating an amp into clipping. With average music
material (and typical crossover frequencies for a 3-way speaker), about 70 percent of the amplifier's output
energy is directed to the woofer, maybe 20-25 percent to the midrange, and 5 or 10 percent to the tweeter. If
clipping occurs, the power to all speaker components increases, however the ratios given above change
(such that the midrange and tweeter end up trying to deal with more than their share of the power). As
should be fairly obvious, this can lead to premature failure of the midrange and/or tweeter.
Detailed Analysis of what happens when music is clipped: This section of the article will illustrate with
numbers and plots what happens to the output signal of an amplifier when the signal is clipped. We will
consider what happens when a 100 watt amp is overdriven, and we will use an actual music signal for the
analysis. To do this analysis, I recorded a portion of a song (Billy Preston’s “Outa-Space”) into the
computer (being certain not to exceed the audio card’s input range). I use an extremely high quality audio
card that maintains very low distortion and very low noise (much better performance than the typical
“SoundBlaster” card). I then used MATLAB software to do signal conditioning and analysis on this music
signal to generate the plots and numbers.
We are going to assume we are using the 100W amp that we have been using in this article. I took the
music signal I recorded and (using MATLAB) I adjusted the signal such that the output voltage just touched
the peak voltage output for a 100 W amp (40 volts for 8 ohms). I then took that same signal, and (again
using MATLAB), I modified it so that it would represent the full output of a 150W amp, a 200W amp, a
300W amp and a 400W amp. I then took these signals and “clipped” them (using MATLAB) so that they
would each represent the output of a 100W amp trying to act like those other amps. I then took the
resulting waveforms and determined a number of factors for each of them, including RMS voltage, power,
crest factors and signal compression. In this section we are delving into more detail, hopefully it will all
make sense. Plots were generated to help illustrate what things look like at various power levels.
First, let’s just show a plot of the music signal. The signal was sampled at 44100 samples per second (the
rate that music CDs work at). Below is a plot (Figure 5) that shows what the signal looks like (this is one
channel only). The vertical scale is in volts, and the horizontal scale is in samples. This “chunk” of music
is just under 6 seconds long (262144 samples):

Figure 5.
The signal represented in this plot is that of real music, and it is just at the threshold of clipping for this 100
W amp. One thing that can be seen is that most of the time the signal level is rather low… it is only for
very short bursts that the signal reaches the threshold of clipping. These short bursts of high level signal
are from bass drums! For the signal above, the amplifier would be delivering 100 watts of power for the
largest peaks (those that just reach the + or – 40 volt rail). The RMS voltage (note voltage not power) of
the signal in the plot above has a value of 6.4 volts RMS. The “RMS” power (RMS is in quotes because
RMS is not really a technically correct term) for this signal is about 5.13 watts! Is this correct? Yes… we
have a 100 W amp at the threshold of clipping, and the “average” power being delivered to the speaker is
only a little above 5 watts! This is not uncommon for high fidelity music recorded on CD. In this case, the
difference between the peak power (100 watts) and the RMS power is about 19.5 times, or about 12.9 dB!
This delta is known as crest factor (in this case for power). I am bringing attention to this value as we will
show how crest factor diminishes as clipping increases.
Next, we show a plot that shows what things look like when we play the 100 W amp as if it were a 150 W
amp. Because we are exceeding the 100 W amp’s capability, the music will be clipped. It will now look
like that in Figure 6 below:

Figure 6.
The signal in Figure 6 does not look terribly different than in Figure 5 above. However, the signal has
indeed changed. The RMS voltage of the signal above is now 7.83 volts and the RMS power is 7.66 watts.
The crest factor (power) is now approximately 11.2 dB (note that we have compressed the signal by about
1.7 dB by overdriving this amplifier). This makes sense, as a 150 W amp should have about 1.76 dB more
power than a 100 W amp.
Next, we show a plot that shows what things look like when we play the 100 W amp as if it were a 400 W
amp. Because we are exceeding the 100 W amp’s capability considerably, the music will be clipped pretty
badly. It will now look like that in Figure 7 below:

Figure 7.

In this plot it should be pretty obvious that the overall level of the signal is up considerably compared to the
first plot where there is no clipping. The RMS voltage of the signal in Figure 7 is now 12.14 volts and the
RMS power is 18.4 watts. The crest factor (power) is now approximately 7.35 dB (note that we have
compressed the signal by about 5.5 dB by overdriving this amplifier).
Rather than show lots of plots that look similar (for the other amplifiers), the table below shows the
numbers that result when a 100W amp is overdriven to the point of trying to act like other amplifiers of
various power capabilities:

100W Amplifier Power Supply RMS voltage RMS power Power Crest Power
trying to act like: Rails with music with music Factor
 signal (1) signal (2) Compression

100W +/- 40V 6.40 Vrms 5.13 W 12.90 dB 0 dB

150W +/- 49 V 7.83 Vrms 7.66 W 11.16 dB 1.74 dB

200W +/- 56.6 V 8.97 Vrms 10.06 W 9.98 dB 2.92 dB

300W +/- 69.3 V 10.75 Vrms 14.44 W 8.40 dB 4.50 dB

400W +/- 80 V 12.14 Vrms 18.41 W 7.35 dB 5.55 dB

NOTES: (1) The value listed in this column (for amps other than the 100 W amp) is the RMS value of the
voltage that would result if a 100 W amp was trying to amplify the input signal to the wattage listed in that
row. (2) The value listed in this column (for amps other than the 100 W amp) is the RMS value of the
power that would result if a 100 W amp was trying to amplify the input signal to the wattage listed in that
row.
What does this table tell us? When an amplifier is overdriven into clipping, the peak output power of the
amp does not increase, however the average (RMS) power to the speaker does go up. Also, compression of
the signal occurs (the amount of compression is a function of how bad the clipping is). Audiophiles strive to
maintain the best possible fidelity in music, so clipping an amp is highly intolerable. However, for live
music applications (or house parties), amplifiers being driven into clipping is the norm. Although it is not
shown in the table, in extreme clipping conditions the RMS power will approach the value of the peak
power the amp can deliver. For an amp rated at 100 W RMS, the power output during extreme clipping
will approach 200 W (assuming the power supply can keep up).
Some readers might now be asking “What’s the big deal?” The table above shows that I can run my 100 W
amp at a level that tries to mimic a 400 W amp and I only increase the average power to the speaker (with
this particular music CD) from about 5 watts to about 18 watts! The key thing is this: the music off the CD
I used for this analysis has a LOT of dynamic range. Not all music is the same… some types of music have
a lot less dynamic range (less crest factor), and when signals like this are overdriven the situation can be a
lot more dramatic. Music from an FM radio station is quite compressed compared to a CD, and this would
be the type of signal that could really cause problems if driven into clipping. The best advice I can give:
avoid situations where more than occasional clipping occurs.
Is clipping bad for an amplifier?: This question is a little off topic for this article, but I'll speak to it
briefly. It really depends a lot on how well designed the amplifier is as to whether or not clipping is bad for
it. Overall, some minor or occasional clipping is no big deal (in fact most bands and DJs operate this way),
but excessive clipping can stress things out. In cases of extreme clipping, operating an amplifier in this
fashion might be more likely to stress out (or damage) the power supply as compared to the output
semiconductors. Why? When extreme clipping occurs, the output transistors may actually have an easier
time (as compared to running at full output with clean music signals) because when extreme clipping
occurs they are operating more like a switch, either being fully ON or fully OFF. This means that the output
transistors have to deal with less heat dissipation (basically by being fully ON they are sending the vast
majority of the power supply's juice to the speakers). Relatively less heat will be dissipated in the transistor
in cases of extreme clipping. However, the power supply (consisting of transformer, rectifiers, etc) has to
dish out a lot more power than it was likely designed to do (continuously), and if the electronic components
used with it were marginally designed or rated to begin with, they could fail. Better amps use components
that are more conservatively rated and these amps will be better able to handle the added stress. Amps used
for professional touring outfits are often designed to be very rugged in this regard (and of course they cost
quite a bit more than amps with similar power ratings that are targeted to the consumer electronic market).
Low end amps that use electronic components with bare minimum ratings will be the first to quit when
operated in a clipped mode (especially if it is severe clipping). How can you tell if your amp is being
stressed out? If it has lighted meters, keep an eye on the light bulbs that light the meters. If they are
flickering (going dim when the music is really loud) you are probably at or past the limit of what the amp
can safely do. The lights are most likely dimming because the power supply within the amp (which likely
powers everything in it) is being "sucked down" by the excessive power being sent to the speakers.
In general, my personal recommendation is to avoid running an amplifier into more than occasional
clipping as it can only increase the chances of something bad happening. You could blow speakers, stress
out the amp, and if nothing else the sound quality will suffer!

Interpreting Amplifier Specifications: This section explains what various power amplifier specifications
mean. We will consider the specifications of the (vintage) Kenwood KA-9100 amplifier:
 Power Output: 90 watts per channel, minimum RMS, at 8 ohms, from 20 Hz to 20,000 Hz
with no more than 0.03% total harmonic distortion. This basically means that this amplifier
can deliver 90 watts per channel into an 8 ohm load, and it can do it at any frequency between 20
Hz and 20,000 Hz, and the total harmonic distortion will be no more than 0.03%. Although it is not
stated, this power is capable of being delivered with both channels operating at 90 watts at the
same time. Again, keep in mind that technically there is no such thing as RMS watts; this amplifier
basically delivers 90 watts, however it does it using a sine wave signal (not a DC signal)! Note: in
reality, almost all amps will put out more than their rated power output. When an amp is rated for
power output, the design is such that all production amplifiers will meet this rating. To guarantee
that this occurs, the ratings are on the conservative side (otherwise a manufacturer could be
accused of "lying" and it would be bad for their reputation). So, most amps will put out at least
10% more power than their actual ratings.
 Both channels driven: 95 +95 watts 8 ohms at 1000 Hz, 110 + 110 watts at 4 ohms at 1000 Hz.
This shows that the amp can do a bit better at 1000 Hz (in terms of maximum power output). Most
amps do better at mid frequencies with regard to maximum power output. Low frequencies and
high frequencies are more demanding and most amps will put out a little less power at these
extremes. This spec for the Kenwood also shows the power at 4 ohms.
 Dynamic Power Output: 470 watts at 4 ohms. This is basically showing the limits of what this
amp can do (instantaneous power) at 4 ohms. This is total output (from both channels). This amp
can do 110 watts per channel continuous RMS at 4 ohms; recall (from above explanations) that the
peak power for an amp is twice its RMS rating; so, 220 + 220 = 440. This is not the same as the
stated 470 watts; however this basically means that the 110 W RMS rating at 4 ohms had a little
bit of conservatism in it, and/or it is for very short bursts of sound that come and go before the
power supply rails of the amp can get pulled down by the demand.
 Total Harmonic Distortion: 0.03% at rated power into 8 ohms, 0.01% at 1 watt into 8 ohms.
This spec is basically saying that with a pure sine wave as an input signal, the amp will generate
no more than 0.03% of distortion as harmonics. So, if the amp is putting out 90 watts, 0.027 watts
will be distortion and 89.973 watts will be faithful reproduction. This amount of distortion is
vanishingly low in terms of what the human ear can detect. This basically represents a signal that
is about 35 dB down from the main signal. Being that this distortion signal is "harmonic
distortion" it is a lot more tolerable than (for example) intermodulation distortion. Note that the
distortion at 1 watt is a bit lower. This is normal for class AB amps (the Kenwood KA-9100 is one
such amp). Class AB amps generally have higher levels of distortion at very low output levels (due
to crossover distortion).
 Intermodulation Distortion (60 Hz : 7kHz = 4:1): 0.03% at rated power into 8 ohms, 0.01%
at 1 watt into 8 ohms . This is a bit more complicated. Basically, to arrive at this spec two sine
wave signals are applied to the amplifier input. In this case, one sine wave was 60 Hz and one was
7000 Hz. The 60 Hz sine wave amplitude was 4 times the amplitude of the 7000 Hz sine wave.
This composite signal (which is no longer a simple sine wave) is adjusted in amplitude (keeping
the relative ratio of 4:1) until the amp was putting out 90 watts. Then, a frequency analyzer is used
to look for frequency lines that are not harmonically related to the two input signals. For example,
the frequencies of 120 Hz, 180 Hz, 240 Hz, 300 Hz, etc are ignored. Also ignored are the
harmonics of the 7000 Hz signal (these would b 14,000 Hz, 21,0000 Hz, etc). What they do look
for in this spec is sum and difference frequencies. for example, frequencies such as 6940 Hz and
7060 Hz would be looked for (for starters), along with any other frequencies that are not
harmonics of the main tones. Intermodulation distortion results when two sine waves are applied
to a non linear system. Good amplifiers are quite linear (when operated within their design limits),
but they are not perfect. All amplifiers will generate some amount of intermodulation distortion.
The smaller the number the better, as this kind of distortion is easier for the human ear to detect as
compared to harmonic distortion.
 Power bandwidth: 5 Hz to 60,000 Hz. This is the frequency range over which the amplifier can
put out substantial amounts of power. Although not mentioned, it is likely that the frequency limits
given represent the "half power" points. In other words, the amp can put out (at least) 45 watts at 5
Hz and also at 60,000 Hz. At points in between it will put out more power (the full 90 watts per
channel is valid for frequencies of 20 Hz to 20,000 Hz). Basically, the wider the power bandwidth
the better, although in some cases an extra wide band can lead to problems.
 Frequency Response: DC to 100,000 Hz, +0 dB, - 1dB. This shows the frequency response of
the amplifier, but it does not mention the power at which it was measured (so we have to assume it
is probably 1 watt or some other small value). It is definitely not at full output of this amp. Why
DC? This amp uses DC coupled stages within it. Not all amps do this, and in general it is not
needed to have great sound.
 Signal to Noise Ratio: 115 dB (short circuited). This is a pretty important spec. It is measured by
putting a short circuit at the input (certainly not the output!) of the amplifier, and then measuring
the noise at the output. Although it is not mentioned, this is almost certainly what is called an "A"
weighted noise measurement (as compared to a "C" weighted measurement). A "C" weighting
measurement is basically no weighting (adjustment curve) at all, it is a "full band" noise
measurement (from 20 Hz - 20,000 Hz). "A" weighting is basically a measurement that is used to
make the measurement more akin to what the human ear hears. Human hearing is not as sensitive
at frequency extremes, so noise (for example) that is very low in frequency or very high in
frequency does not sound as loud as noise at mid band frequencies. The "A" weighting curve
(measurement) tries to take this into account. "C" weighting measurements would have lower
numbers (for example instead of 115 dB it might be 109 dB). "C" weighting numbers do not look
as good. So far as I know most amps (those used for amplifying music) are rated for signal to noise
ration using A weighting. Make sure you compared amps based on the same measurement scale or
one will seem to be a lot better than another when in fact they might be the same! What does a
rating of -115 dB mean? Basically that the sound level of the output noise of the amplifier would
sound (to the human ear) to be 115 dB quieter than the signal the amp can generate at full power.
You'd NEVER hear such noise with many kinds of music (when the music is being played at or
near the limits of the amplifier's power capability. The exception is possibly classical music, a type
of music that has very quiet passages mixed n with very loud passages. Amplifiers can only be so
quiet, there is no (practical) way to make a noiseless amplifier.
 Damping factor: 50 at 8 ohms. Damping factor is basically a measure of the output impedance of
the amplifier. High numbers are better.
 Input Sensitivity/Impedance: 1.0 V / 50 k ohms. This specification shows home much voltage
must be applied to the input of the amplifier (into a load impedance of 50,000 ohms) in order to
get the full rated output. If more signal than this is applied, the amp will clip with the volume
control all the way up. If less than this amount of signal is applied, the amp will not reach full
output capability. This number is important when considering what the previous stage can deliver
(a preamp for example). The prior stage should be able to deliver at least this amount of voltage
(preferably more).
  Speaker Impedance: Accept 4 to 16 ohms. Nothing too hard here, it basically states that speakers
with impedance anywhere from4 to 16 ohms will work

Audio crossover

From Wikipedia, the free encyclopedia

Jump to: navigation, search

A passive 2-way crossover designed to operate at loudspeaker voltages

Audio crossovers are a class of electronic filter used in audio applications. Most individual loudspeaker
drivers are incapable of covering the entire audio spectrum from low frequencies to high frequencies with
acceptable relative volume and lack of distortion so most hi-fi speaker systems use a combination of
multiple loudspeakers drivers, each catering to a different frequency band. Crossovers split the audio signal
into separate frequency bands that can be separately routed to loudspeakers optimized for those bands.
Active crossovers allow drivers covering different frequency ranges to be powered by separate amplifiers, a
configuration known as bi-amping.
Signal crossovers allow the audio signal to be split into bands that are adjusted (equalized, compressed,
echoed, etc.) separately before they are mixed together again. Some examples are: multiband dynamics
(compression, limiting, de-essing), multiband distortion, bass enhancement, high frequency exciters, and
noise reduction (for example: Dolby A noise reduction).

Overview

Comparison of the magnitude response of 2 pole Butterworth and Linkwitz-Riley crossover filters. The
summed output of the Butterworth filters has a +3dB peak at the crossover frequency.
The definition of an ideal audio crossover changes relative to the task at hand. If the separate bands are to
be mixed back together again (as in multiband processing), then the ideal audio crossover would split the
incoming audio signal into separate bands that do not overlap or interact and which result in an output
signal unchanged in frequency, relative levels, and phase response. This ideal performance can only be
approximated. How to implement the best approximation is a matter of lively debate. On the other hand, if
the audio crossover separates the audio bands in a loudspeaker, there is no requirement for mathematically
ideal characteristics within the crossover itself, as the frequency and phase response of the loudspeaker
drivers within their mountings will eclipse the results. Satisfactory output of the complete system
comprising the audio crossover and the loudspeaker drivers in their enclosure(s) is the design goal. Such a
goal is often achieved using non-ideal, asymmetric crossover filter characteristics.[1]
Many different crossover types are used in audio, but they generally belong to one of the following classes.

[edit] Classification
[edit] Classification based on the number of filter sections
In loudspeaker specifications, one often sees a speaker classified as an "N-way" speaker. N is a positive
whole number greater than 1, and it indicates the number of filter sections. A 2-way crossover consists of a
low-pass and a high-pass filter. A 3-way crossover is constructed as a combination of low-pass, band-pass
and high-pass filters (LPF, BPF and HPF respectively). The BPF section is in turn a combination of HPF
and LPF sections. 4 (or more) way crossovers are not very common in speaker design, primarily due to the
complexity involved, which is not generally justified by better acoustic performance.
An extra HPF section may be present in an "N-way" loudspeaker crossover to protect the lowest-frequency
driver from frequencies lower than it can safely handle. Such a crossover would then have a bandpass filter
for the lowest-frequency driver. Similarly, the highest-frequency driver may have a protective LPF section
to prevent high frequency damage, though this is far less common.
Recently, a number of manufacturers have begun using what is often called "N.5-way" crossover
techniques for stereo loudspeaker crossovers. This usually indicates the addition of a second woofer that
plays the same bass range as the main woofer but rolls off far before the main woofer does.
Remark: Filter sections mentioned here is not to be confused with the individual 2-pole filter sections that a
higher order filter consists of.

[edit] Classification based on components

Crossovers can also be classified based on the design approach; by the type of components used.

[edit] Passive

A passive crossover
A passive crossover is made entirely of passive components, arranged most commonly in a Cauer topology
to achieve a Butterworth filter. Passive filters use resistors combined with reactive components such as
capacitors and inductors. Very high performance passive crossovers are likely to be more expensive than
active crossovers since individual components capable of good performance at the high currents and
voltages at which speaker systems are driven are hard to make, and expensive. Polypropylene, metalized
polyester foil, paper and electrolytic capacitors are common. Inductors may have air cores, powdered metal
cores, ferrite cores, or laminated silicon steel cores, and most are wound with enamelled copper wire. Some
passive networks include devices such as fuses, PTC devices, bulbs or circuit breakers to protect the
loudspeaker drivers from accidental overpowering. Modern passive crossovers increasingly incorporate
equalization networks (e.g., Zobel networks) that compensate for the changes in impedance with frequency
inherent in virtually all loudspeakers. The issue is complex, as part of the change in impedance is due to
acoustic loading changes across a driver's passband.
On the negative side, passive networks may be bulky and cause power loss. They are not only frequency
specific, but also impedance specific. This prevents interchangeability with speaker systems of different
impedances. Ideal crossover filters, including impedance compensation and equalization networks, can be
very difficult to design, as the components interact in complex ways. Crossover design expert Siegfried
Linkwitz said of them that "the only excuse for passive crossovers is their low cost. Their behavior changes
with the signal level dependent dynamics of the drivers. They block the power amplifier from taking
maximum control over the voice coil motion. They are a waste of time, if accuracy of reproduction is the
goal."[2]
Alternatively, passive components can be utilised to construct filter circuits before the amplifier. This is
called passive line-level crossover.

[edit] Active
An active crossover contains active components (i.e., those with gain) in its filters. In recent years, the most
commonly used active device is an op-amp; active crossovers are operated at levels suited to power
amplifier inputs in contrast to passive crossovers which operate after the power amplifier's output, at high
current and in some cases high voltage. On the other hand, all circuits with gain introduce noise, and such
noise has a deleterious effect when introduced prior to the signal being amplified by the power amplifiers.

Typical usage of an active crossover, though a passive crossover can be positioned similarly before the
amplifiers
Active crossovers always require the use of power amplifiers for each output band. Thus a 2-way active
crossover needs two amplifiers—one each for the woofer and tweeter. This means that an active crossover
based system will often cost more than a passive crossover based system. Despite the cost and complication
disadvantages, active crossovers provide the following advantages over passive ones:
 a frequency response independent of the dynamic changes in a driver's electrical characteristics.
 typically, the possibility of an easy way to vary or fine tune each frequency band to the specific
drivers used. Examples would be crossover slope, filter type (e.g., Bessel, Butterworth, etc.),
relative levels, ...
 better isolation of each driver from signals handled by other drivers, thus reducing intermodulation
distortion and overdriving
 The power amplifiers are directly connected to the speaker drivers, thereby maximizing amplifier
damping control of the speaker voice coil, reducing consequences of dynamic changes in driver
electrical characteristics, all of which are likely to improve the transient response of the system
 reduction in power amplifier output requirement. With no energy being lost in passive
components, amplifier requirements are reduced considerably (up to 1/2 in some cases), reducing
costs, and potentially increasing quality.

Active crossovers can be implemented digitally using a DSP chip or other microprocessor. They either use
digital approximations to traditional analog circuits, known as IIR filters (Bessel, Butterworth, Linkwitz-
Riley etc.), or they use Finite impulse response (FIR) filters. IIR filters have many similarities with analog
filters and are relatively undemanding of CPU resources; FIR filters on the other hand usually have a higher
order and therefore require more resources for similar characteristics. They can be designed and built so
that they have a linear phase response, which is thought desirable by many involved in sound reproduction.
There are drawbacks though—in order to achieve linear phase response, a longer delay time is incurred
than would be necessary with an IIR or minimum phase FIR filters. IIR filters, which are by nature
recursive have the drawback that if not carefully designed they may enter limit cycles resulting in non-
linear distortion.

[edit] Mechanical
This crossover type is mechanical and uses the properties of the materials in a driver diaphragm to achieve
the necessary filtering. Such crossovers are commonly found in full-range speakers which are designed to
cover as much of the audio band as possible. One such is constructed by coupling the cone of the speaker to
the voice coil bobbin through a compliant section and directly attaching a small lightweight whizzer cone to
the bobbin. This compliant section serves as a compliant filter, so the main cone is not vibrated at higher
frequencies. The whizzer cone responds to all frequencies, but due to its smaller size only gives a useful
output at higher frequencies, thereby implementing a mechanical crossover function. Careful selection of
materials used for the cone, whizzer and suspension elements determines the crossover frequency and the
effectiveness of the crossover. Such mechanical crossovers are complex to design, especially if high fidelity
is desired. Computer aided design has largely replaced the laborious trial and error approach that was
historically used. Over several years, the compliance of the materials may change, negatively affecting the
frequency response of the speaker.
A more common approach is to employ the dust cap as a high frequency radiator. The dust cap radiates low
frequencies, moving as part of the main assembly, but due to low-mass and reduced damping, radiates
increased energy at higher frequencies. As with whizzer cones, careful selection of material, shape and
position are required to provide smooth, extended output. High frequency dispersion is somewhat different
for this approach than for whizzer cones. A related approach is to shape the main cone with such profile,
and of such materials, that the neck area remains more rigid, radiating all frequencies, while the outer areas
of the cone are selectively decoupled, radiating only at lower frequencies. Cone profiles and materials can
be modeled in FEA software and the results predicted to excellent tolerances.
Speakers which use these mechanical crossovers have some advantages in sound quality despite the
difficulties of designing and manufacturing them, and despite the inevitable output limitations. Full-range
drivers have a single acoustic center, and can have relatively modest phase change across the audio
spectrum. For best performance at low frequencies, these drivers require careful enclosure design. Their
small size (typically 165 to 200 mm) requires considerable cone excursion to reproduce bass effectively,
but the short voice coils required for reasonable high frequency performance can only move over a limited
range. Nevertheless, within these constraints, cost and complications are reduced, as no crossovers are
required.

[edit] Classification based on filter order or slope

Just as filters have different orders, so do crossovers, depending on the filter slope they implement. The
final acoustic slope may be completely determined by the electrical filter or may be achieved by combining
the electrical filter's slope with the natural characteristics of the driver. In the former case, the only
requirement is that each driver has a flat response at least to the point where its signal is approximately
−10dB down from the passband. In the latter case, the final acoustic slope is usually steeper than that of the
electrical filters used. A third- or fourth-order acoustic crossover often has just a second order electrical
filter. This requires that speaker drivers be well behaved a considerable way from the nominal crossover
frequency, and further that the high frequency driver be able to survive a considerable input in a frequency
range below its crossover point. This is difficult in actual practice. In the discussion below, the
characteristics of the electrical filter order is discussed, followed by a discussion of crossovers having that
acoustic slope and their advantages or disadvantages.
Most audio crossovers use first to fourth order electrical filters. Higher orders are not generally
implemented in passive crossovers for loudspeakers, but are sometimes found in electronic equipment
under circumstances for which their considerable cost and complexity can be justified.

[edit] First order

First-order filters have a 20 dB/decade (or 6 dB/octave) slope. All first-order filters have a Butterworth
filter characteristic. First-order filters are considered by many audiophiles to be ideal for crossovers. This is
because this filter type is 'transient perfect', meaning it passes both amplitude and phase unchanged across
the range of interest. It also uses the fewest parts and has the lowest insertion loss (if passive). A first-order
crossover allows more signals of unwanted frequencies to get through in the LPF and HPF sections than do
higher order configurations. While woofers can easily take this (aside from generating distortion at
frequencies above those they can properly handle), smaller high frequency drivers (especially tweeters) are
more likely to be damaged since they are not capable of handling large power inputs at frequencies below
their rated crossover point.
In practice, speaker systems with true first order acoustic slopes are difficult to design because they require
large overlapping driver bandwidth, and the shallow slopes mean that non-coincident drivers interfere over
a wide frequency range and cause large response shifts off-axis.

[edit] Second order

Second-order filters have a 40 dB/decade (or 12 dB/octave) slope. Second-order filters can have a Bessel,
Linkwitz-Riley or Butterworth characteristic depending on design choices and the components used. This
order is commonly used in passive crossovers as it offers a reasonable balance between complexity,
response, and higher frequency driver protection. When designed with time aligned physical placement,
these crossovers have a symmetrical polar response, as do all even order crossovers.
It is commonly thought that there will always be a phase difference of 180° between the outputs of a
(second order) low-pass filter and a high-pass filter having the same crossover frequency. And so, in a 2-
way system, the high-pass section's output is usually connected to the high frequency driver 'inverted', to
correct for this phase problem. For passive systems, the tweeter is wired with opposite polarity to the
woofer; for active crossovers the high-pass filter's output is inverted. In 3-way systems the mid-range driver
or filter is inverted. However, this is generally only true when the speakers have a wide response overlap
and the acoustic centers are physically aligned.

[edit] Third order

Third-order filters have a 60 dB/decade (or 18 dB/octave) slope. These crossovers usually have Butterworth
filter characteristics; phase response is very good, the level sum being flat and in phase quadrature, similar
to a first order crossover. The polar response is asymmetric. In the original D'Appolito MTM arrangement,
a symmetrical arrangement of drivers is used to create a symmetrical off-axis response when using third-
order crossovers.
Third-order acoustic crossovers are often built from first- or second-order filter circuits.

[edit] Fourth order

Fourth-order crossover slopes shown on a Smaart transfer function

Fourth-order filters have an 80 dB/decade (or 24 dB/octave) slope. These filters are complex to design in
passive form, as the components interact with each other. Steep-slope passive networks are less tolerant of
parts value deviations or tolerances, and more sensitive to mis-termination with reactive driver loads. A 4th
order crossover with −6 dB crossover point and flat summing is also known as a Linkwitz-Riley crossover
(named after its inventors), and can be constructed in active form by cascading two 2nd order Butterworth
filter sections. The output signals of this crossover order are in phase, thus avoiding partial phase inversion
if the crossover bandpasses are electrically summed, as they would be within the output stage of a
multiband compressor. Crossovers used in loudspeaker design do not require the filter sections to be in
phase: smooth output characteristics are often achieved using non-ideal, asymmetric crossover filter
characteristics.[1] Bessel, Butterworth and Chebyshev are among the possible crossover topologies.
Such steep-slope filters have greater problems with overshoot and ringing[3] but there are several key
advantages, even in their passive form, such as the potential for a lower crossover point and increased
power handling for tweeters, together with less overlap between drivers, dramatically reducing lobing, or
other unwelcome off-axis effects. With less overlap between adjacent drivers, their location relative to each
other becomes less critical and allows more latitude in speaker system cosmetics or (in car audio) practical
installation constraints.

[edit] Higher order

Passive crossovers giving acoustic slopes higher than fourth-order are not common because of cost and
complexity. Filters of up to 96 dB per octave are available in active crossovers and loudspeaker
management systems.

[edit] Mixed order

Crossovers can also be constructed with mixed order filters. For example, a second order lowpass
combined with a third order highpass. These are generally passive and are used for several reasons, often
when the component values are found by computer program optimization. A higher order tweeter crossover
can sometimes help compensate for the time offset between the woofer and tweeter, caused by non aligned
acoustic centers.

[edit] Classification based on circuit topology

Series and parallel crossover topologies. The HPF and LPF sections for the series crossover are
interchanged with respect to the parallel crossover since they appear in shunt with the low & high
frequency drivers.

[edit] Parallel
Parallel crossovers are by far the most common. Electrically the filters are in parallel and thus the various
filter sections do not interact. This makes two-way crossovers easier to design because the sections can be
considered separately, and because component tolerance variations will be isolated. Parallel crossovers also
have the advantage of allowing the speaker drivers to be bi-wired. In the years before computer modeling,
simplistic three-way crossovers were designed as a pair of two-way crossovers, but the advent of iterative
design software has taught that this old technique creates excess gain and a 'haystack' response in the
midrange output, together with a lower than anticipated input impedance.

[edit] Series
In this topology, the individual filters are connected in series, and a driver or driver combination is
connected in parallel with each filter. To understand the signal path in this type of crossover, refer to the
"Series Crossover" figure, and consider a high frequency signal that, during a certain moment, has a
positive voltage on the upper Input terminal compared to the lower Input terminal. The low pass filter
(LPF) presents a high impedance to the signal, and the tweeter presents a low impedance; so the signal
passes through the tweeter. The signal continues to the connection point between the woofer and the high
pass filter (HPF). There, the HPF presents a low impedance to the signal, so the signal passes through the
HPF, and appears at the lower Input terminal. A low frequency signal with a similar instantaneous voltage
characteristic first passes through the LPF, then the woofer, and appears at the lower Input terminal.

[edit] Derived
Derived crossovers include active crossovers in which one of the crossover responses is derived from the
other through the use of a differential amplifier. For example, the difference between the input signal and
the output of the high pass section is a low pass response.[4] Thus, when a differential amplifier is used to
extract this difference, its output constitutes the low pass filter section. The main advantage of derived
filters is that they produce no phase difference between the high pass and low pass sections at any
frequency.[4] The disadvantages are either
 (a) that the high pass and low pass sections often have different levels of attenuation in their stop
bands, i.e. their slopes are asymmetrical,[4] or
 (b) that the response of one or both sections peaks near the crossover frequency,[5]
or both. In case (a), above, the usual situation is that the derived low pass response attenuates at a much
slower rate than the fixed response. This requires the speaker to which it is directed to continue to respond
to signals deep into the stopband where its physical characteristics may not be ideal. In the case of (b),
above, both speakers are required to operate at higher volume levels as the signal nears the crossover
points. This uses more amplifier power and may drive the speaker cones into non-linearity