The Application Support Group includes individuals with extensive experience in all types of sound reinforcement applications, from the smallest clubs or houses of worship to the largest stadiums. ASG members have backgrounds in theatrical sound & lighting design, design/build contracting, touring/live sound reinforcement, studio recording, and system engineering and alignment. All are trained in the use of AutoCAD, EASE, and SIA SmaartLive software. Individual members are proficient in the use of the latest DSP applications, including, but not limited to, products from BSS, Shure, Crown, Crest, QSC, White Instruments, Rane, and many more.

Phone: 1.800.992.5013 or 508.234.6158
Contact Online
Fax:  508.266.6202
Also, utilize EAW’s online forums moderated by the ASG team: http://forums.eaw.com/

Bear in mind that extensive design questions can usually be answered in a more timely fashion by phone.

Also, be aware that ASG serves many EAW customers. We will promptly respond to your initial request so you know that we are working on your project, but please allow up to 1-2 weeks for a complete recommendation depending on the size and complexity of the project.

There is no exact answer to the question of how much amplifier power you should use for a particular loudspeaker. Actually, there are three separate and very distinct issues regarding selecting amplifier power for loudspeakers.

• Loudspeaker Power Handling Rating - The power handling rating in EAW's specifications means that the loudspeaker has passed our standard power handling test. In this test the loudspeaker is "exercised" to a point of damage or failure. The power rating resulting from this test is intended to be used as a point of comparison with the power ratings of other loudspeakers. This rating does not necessarily correspond to the best amplifier size to use nor is it a measure of the "safe" amplifier size to use under actual operating conditions.
• Preventing Loudspeaker Damage - Preventing damage to or failure of a loudspeaker is not a function of amplifier size nor the loudspeaker's power rating. Preventing damage is a function of operating an audio system so that a loudspeaker is not stressed beyond its limits. If an audio system is operated improperly, damage to or failure of a loudspeaker can occur even with an amplifier sized well below the loudspeaker's power rating. Contrarily, if an audio system is operated properly, damage to or failure of a loudspeaker can be avoided even with an amplifier sized well in excess of the loudspeaker's continuous (or RMS, average, etc.) power rating.
• Selecting an Appropriate Amplifier Size - The amplifier for your loudspeaker should be sized according to both the sound levels required and the type of audio signals that will be reproduced. If you are unsure of how to determine these things, consult a qualified professional or contact EAW's Application Support Group. As a rule of thumb, where the full capability of the loudspeaker is needed to achieve appropriate acoustic output levels, EAW recommends an amplifier that is twice the loudspeaker's power handling specification. This allows the amplifier to reproduce peaks 6 dB above the specified power handling. However, this recommendation does NOT guarantee trouble-free operation, and assumes that operation of the loudspeaker can be properly controlled. It is the responsibility of the audio system operator to ensure that all equipment in the system is operated within its capabilities. That is the only way to ensure that loudspeakers do not get stressed beyond their limits to the point of damage or failure.
  
  For a more detailed discussion of this topic, please download our Amplifier Power paper (PDF file, 112 KB).

EAW's older power ratings and newer Accelerated Life Test rating are similar to the "rms" or "continuous" power ratings used by other loudspeaker manufacturers. As such, these ratings are all reasonably comparable. However, because of the different ways manufacturers do power testing, differences between these ratings of up to about +/-20% should not be considered significant.

Processor settings are available as downloadable PDF files on the Processor Downloads page.

The input connectors on the loudspeaker will be one of the following types with the pin connections as listed. Because of possible production changes, check the input panel labeling to verify proper connections. For barrier strips, the proper connections are marked for each terminal on the loudspeaker input label.

HPF is the common abbreviation for a high pass filter, while LPF is the common abbreviation for a low pass filter. High and low pass filters allow certain frequencies to "pass" through them while rejecting others. As its name implies, a high pass filter passes frequencies above its filter frequency and reduces the level of those frequencies below it. A low pass filter passes frequencies below its filter frequency and reduces the level of those frequencies above it.

HPFs and LPFs are generally defined by three characteristics: a cutoff frequency, a topology, and a slope. The cutoff frequency is the frequency where the response of the filter falls to some level below that of the unfiltered ("passed") frequencies. This level is generally 1/2 the voltage of the unfiltered frequencies or -6 dB. The topology determines the shape of the filter's frequency response. The most commonly used filter topologies are Butterworth, Linkwitz-Riley, and Bessel. The slope of the filter defines how fast the level is reduced beyond the cutoff frequency. This is usually defined as dB per octave (dB/oct). Common filter slopes are 6, 12, 18, and 24 dB/oct.

HPFs and LPFs have two distinct applications: As Bandstop filters and as Bandpass filters. A special use of Bandpass filters is for crossovers.

Bandstop Filters:
These are used to eliminate frequencies above or below a certain frequency that are not useful for reproduction. This may be because the loudspeaker is incapable of reproducing them or the frequencies do not exist in the audio signal.

For example, speech contains very little information at frequencies above about 8 kHz and below about 150 Hz. This means these two frequency ranges essentially useless for reproducing speech. In this case you would set a HPF for about 150 Hz and an LPF for 8 kHz to eliminate them from the microphone signal path.

Bandpass Filters:
In the example above, the combination of the HPF and LPF created a single band-pass filter. A bandpass filter always consists of an HPF and LPF working together to pass a range of frequencies and reduce the level of any frequencies above and below this range.

Crossover Filters:
Crossovers are filters made from HPFs and LPFs. They divide up the frequency spectrum into the various frequency ranges (bands) needed by the transducers (also commonly referred to as a "drivers"). Crossovers provide high and low frequency bands for 2-way systems and high, mid, and low frequency bands for 3-way systems.

Typically, crossovers are used in conjunction with bandstop filters. These are used to filter out frequencies near to and beyond the high and low frequency limits of the human hearing range. This is a highly recommended practice.

The bandstop filters are usually engaged in the mixing console, main equalizer, or the electronic crossover. For example, in a 2-way system the high pass crossover filter works with an LPF, set near the highest hearing frequency. Working together, these two filters make up a bandpass filter that passes frequencies from the crossover point to the upper limit of hearing. The low pass crossover filter works similarly with an HPF that filters out unwanted very low frequencies.

In the case of a 3-way system, the mid range crossover output is also a bandpass filter. It is formed by the midrange high and low pass crossover filters themselves. This bandpass filter passes only the midrange frequencies.

Thus, in most instances and applications, a loudspeaker crossover as used in a complete audio system, is really made up of two or more bandpass filters, each consisting of an HPF and LPF.

The answer to this question is usually not what the questioner is looking for. The crossover point can be defined as the frequency at which the responses of two filters, an HPF and an LPF, cross one another. More practically, it is the point where one transducer takes over from another going up or down in frequency. This point is really the center of a frequency range over which both transducers contribute to the sound. This is called the overlap area.

To understand just "where" this frequency is, it is useful to know that any transducer is, in fact, a bandpass filter. Every transducer has inherent high and low pass filters. They are mechanical rather than electronic, but their effects on the signal are the same. Each filter has a cutoff frequency, topology and slope. When a crossover filter is used with a transducer, both filters combine to make a new frequency response curve. When you measure the individual transducers with their crossover filters and overlay the response curves, one can readily see the overlap area and determine the center point of that area. Thus, the correct answer to the question is that the crossover point is determined acoustically from the combined response of the crossover's and the transducer's filters.

The real question that needs to be asked is, "What are the crossover settings for (insert EAW loudspeaker model here)". These settings, in conjunction with the acoustical response of the transducers, determine the crossover point(s) for the loudspeaker. One cannot simply state a "crossover point" for setting a proper crossover network. Please see the paper Processor Setting Fundamentals (PDF, 559k) for a more thorough explanation of this topic.

These are the names given to three different types or topologies of filters. They are most often applied in crossover networks. Each one has different characteristics, but there is no definitive answer as to which is the best to use. The choice of which one to use depends entirely on the filter characteristics of the transducer. It is entirely possible to have a crossover between two transducers where the LPF for the lower frequency transducer and the HPF for the higher frequency transducer each have different topologies, slopes, and cutoff frequencies.

As filters, the Butterworth is considered the "gentlest", The Bessell has the most interesting phase shifts and the Linkwitz-Riley the sharpest cut-off. Because there are phase issues associated with all filters, it is hard to describe the differences apart from specific applications.

This is simply a way to state the bandwidth of a filter. The bandwidth is the frequency range (passband) a filter covers. Common Q values and the bandwidths they represent are shown in the following chart:

Q and bandwidth are two different ways to describe the range of frequencies affected by a parametric EQ filter. The two quantities are mathematically related and both state the same thing. Simply put, a "low Q" and a "high bandwidth" filter covers a wide range of frequencies, whereas a "high Q" or "low bandwidth" filter covers a narrow range of frequencies.

Often crossover filters are looked at as mathematical abstractions and implemented as tuned to a certain frequency as the "crossover point". While this seems the obvious method for dividing up frequencies between LF, MF, and/or HF transducers, it ignores a significant fact: loudspeaker transducers are also filters. By nature, they each have "built-in" low and high pass filters.

For a number of good reasons, transducers usually end up being operated near to the extremes of their passbands where their filter characteristics come into play. These filters do not have the characteristics of the textbook filters one can construct from electronic components. The actual crossover filter is a combination of the passive or electronic crossover filters plus the transducer's filter characteristics. Thus, using passive or electronic mirror-image, symmetrical filter crossover swill not result in optimized crossover filters.

Typically asymmetrical crossover filter types, slopes, and cutoff frequencies are needed to precisely match specific transducer filter characteristics through a crossover region. This means that the HPF and LPF will normally be set at different frequencies. Those frequencies can be "underlapped" (LPF lower than the HPF) or "overlapped" (LPF is higher than the HPF), depending on the transducer characteristics. These settings in conjunction with the transducers' outputs will create a flat response across the crossover region without holes or excess energy.

There are typically one of two reasons for such filter settings.

• A PEQ can be used to reduce the level of a transducer anomaly that, while outside its normal operating range, still affects the overall performance as a bump or ripple in the frequency response.

• A PEQ can be used to modify a transducer's phase response or output level near the crossover region. Optimum summation of two transducer's outputs occurs when their phase responses have a similar value and slope at a particular frequency. Placing a filter at or near crossover can be used to create a phase and/or magnitude change that optimizes the summation of the sound from the two transducers.

EAW processor settings assume that all the amplifiers in your system have identical voltage gains. Note that this does not mean identical input sensitivities. Voltage gain is the ratio of input to output voltage and remains fixed, regardless of the load on the amplifier. For example, two amplifiers have different maximum output capabilities but identical voltage gains. In this case, the same input voltage will produce the same output voltage from both amps.

The output gains in EAW's loudspeaker processor settings account for the nominal transducer input sensitivities and wattage needs. This means that these settings assume amplifiers with the same voltage gain.

If the amplifiers have different gains, then the balance in the processor settings between LF, MF, and HF will no longer be correct. Fortunately, to restore the correct balance, the gain differences between amplifiers in a sound system is easily accounted for.

1. Determine the voltage gain of each amplifier in your system. If this is not in an amplifier's published specifications, contact the manufacturer for assistance.

2. Now determine which amplifier has the lowest gain - all other amps must be matched to the gain of this amplifier. You may match the amplifiers in one of two ways.

1. Turn down the input attenuators on all higher gain amps until their gains match that of the lowest gain amplifier. The input attenuator labeling is calibrated in dB of voltage attenuation on many manufacturer's products, making this option a good deal easier. You will have the same gain when a given input signal produces the same voltage output level on all the amplifiers.

2. Determine the difference in dB between the higher gain amplifiers and the lowest gain amplifier and reduce each output gain within the loudspeaker processor for that amplifier by the difference. For example, suppose the HF amplifier has 32 dB of gain and the LF amplifier has 35 dB of gain. The difference is 3 dB. Simply reduce the LF output gain by 3 dB from the factory processor settings. The LF output gain is reduced because it feeds the higher gain amplifier and this must be matched to the lower gain amplifier.

You will get results. However, neither you nor EAW can predict just what those results will be. A crossover is part of the engineering design of a loudspeaker. If you choose to set up a "regular" set of crossover filters without doing acoustical measurements to determine the effects and make appropriate adjustments, then you have not properly tuned the loudspeaker. The results can range from acceptable to poor. However, you will not achieve optimum performance from the loudspeaker. Among other things you may adversely affect its frequency response, phase response, dispersion pattern, power handling, and time alignment.

Crossover design is a critical part of a loudspeaker's performance. Yet, many loudspeaker manufacturers design crossovers within narrow guidelines of price or performance. This can leave the end-user the task of loudspeaker equalization to compensate for crossover deficiencies. EAW does not expect system operators to make our loudspeakers sound great. Our loudspeakers are intentionally engineered to produce great sound right out of the box. To accomplish this, EAW incorporates the time consuming - but superior - iterative process of development that includes creating complex, asymmetrical crossover settings to optimize total system performance.

An iterative process repeats a cycle of operations, beginning each new cycle with the results of the previous one. With each cycle (iteration), the end results moves closer to the "ideal", or "model" result. For EAW loudspeaker systems, the ideal result is flat on-axis response and linear power response. This latter parameter ensures smooth off-axis performance.

The iterative process begins with the measurement of the acoustical and electrical response of the individual transducers in the enclosure using a dedicated laboratory measurement system. The data is then fed into a proprietary software measurement program. Based on the data thus obtained, EAW engineers build a prototype crossover network. The loudspeaker is measured again and the new data is fed back into measurement program. The network is refined through this iterative process until optimal total system performance is achieved.