Article

Monday, February 06, 2012

PreSonus Adds New Control Options To StudioLive Mixers

PreSonus has announced new updates to its StudioLive Series digital mixers, including a number of features not found on any other digital mixer from any manufacturer. 

New features include:

QMix. Up to 10 musicians can simultaneously control their PreSonus StudioLive monitor (aux) mixes using an iPhone or iPod touch and PreSonus’ QMix app, a free download from the Apple App Store. QMix/VSL is the only solution that allows multiple users to each control their own aux from separate iPhones.

Smaart Engine Technology. PreSonus has begun incorporating Rational Acoustics Smaart Measurement Technology for sound-system analysis and optimization directly into PreSonus Virtual StudioLive remote-control/editor/librarian software.

With Smaart technology and VSL, you’ll be able to precisely identify nasty feedback frequencies and get your loudspeakers to play nicer with the room-all without having a degree in acoustical engineering.

The first version of VSL to incorporate Smaart technology will be part of PreSonus Universal Control 1.6, which is expected to be available later this spring.

Universal Control 1.5.3 and StudioLive Remote 1.2. Universal Control 1.5.3 features an improved version of Virtual StudioLive that supports the new QMix iPhone app, including QMix permissions (so that each user controls only one specified aux mix) and the ability to name aux buses.

Universal Control 1.5.3 also adds VSL features that work with PreSonus StudioLive Remote 1.2 for iPad to enable SL Remote permissions so that iPad users can only control front-of-house mixer features or a specified aux. Tap tempo has been added to both VSL and StudioLive Remote.

VSL adds the ability to copy and load channels, copy main mix to aux mix (and aux to aux), link channel faders so that they can move together, and make your StudioLive mixer default to Fader Locate Mode once a fader has been adjusted in VSL or in StudioLive Remote for iPad.

PreSonus

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Friday, January 27, 2012

Everything You Wanted To Know About Sound Level Meters (SLMs)

The primer: what, how, why, what's available, techniques, applications and more

A sound level meter (SLM) is a device used to make frequency-weighted sound pressure level measurements displayed in dB-SPL.

0.0 dB-SPL is the threshold of hearing, and is equal to 20uPa (microPascals). This correlates to what one would aurally perceive when in a deep cave or in a large anechoic chamber.

Packaged as a single-function handheld test device, SLMs are intended to be held at arm’s length during measurements (to reduce the effects of the body on the measurements) or secured to a tripod stand for more stability.

All SLMs feature an omnidirectional measurement quality condenser microphone, a mic preamp, frequency weighting networks, an RMS detector circuit, averaging circuits, the meter display, AC and DC outputs used to feed other measurement devices or for recording (see Figure 1, below).

Most SLMs have the same set of user adjustments, including SPL range selection, A and C weighting filters, slow and fast detector response, and minimum or maximum SPL.

The SPL range switch provides a balance between minimizing the preamp noise level and measuring a wide range of sound pressure levels.

Most of the commonly available SLMs measure from about 30 to 130 dB-SPL and do this in 3-4 ranges.

 
The more advanced and expensive SLMs feature removable microphones, 1-octave and/or 1/3-octave filter sets, additional weighting filters including B, D and Flat or Linear (no filter), additional detector response options (Impulse and Peak), averaging (over time) and data logging or storage (either on-board, as computer files or both).

Almost all SLMs are designed and specified to perform to one of four internationally standardized levels of accuracy:

 
Note that these tolerances are at 1 kHz, the standard calibration frequency for SPL measurement. In order to ensure the flatness of the SLM there are additional tolerances specified for various frequency bands and microphone classes as well.

Class-0 SLMs are employed primarily to calibrate other SLMs and may be used for very high precision noise measurement in controlled spaces and/or for academic research.

Class-1 and Class-2 SLMs are most widely used by acousticians, sound system professionals, industrial designers/ manufacturers and researchers in academia and government. Measurements made with these levels of accuracy are generally acceptable as evidence in the resolution of legal disputes.

Class-3 SLMs are restricted to noise survey meters and dosimeters.

Microphone Sizes
Most general purpose SLMs are provided with a 1/2-inch free field microphone, permanently attached to the SLM body.

Higher quality SLMs with removable microphones may be outfitted with capsule sizes ranging from 1/8-in to 1-in.

The smaller microphone capsules (1/8-inch and 1/4-inch) have three primary advantages:
—They are capable of higher SPL measurements;
—They remain omnidirectional up to higher frequencies;
—And they present less disturbance of the air due to their reduced physical size (the mic itself has less impact on the measurement).

Larger 1-inch capsules exhibit less self-noise and are therefore best suited to measurement in very quiet spaces, of very low-level noise sources, and of low-frequency energy.

Microphone Classes
One can also choose between three different classes of measurement microphones: Random Incidence, Free Field, and Pressure.

In reality, these three classes of microphone are all fundamentally “pressure response” microphones and are designed to sense pressure - not pressure gradients or particle velocity.

The difference is primarily in how they pick up sound at high frequencies, and this relates to how they are positioned relative to the sound source.

At higher frequencies (where wavelengths become so small as to equal, or be smaller than, the size of the microphone diaphragm) the direction in which the microphone is aimed - relative to the direction of the sound being measured - will have an effect on the accuracy of the measurement.

Random Incidence microphones are intended for use in a diffuse sound field where the sound is arriving from all directions due to a high level of reflections.

This type of microphone must be angled approximately 70 degrees to the sound source in order to yield accurate measurements.

Free Field microphones are designed for use in an open space (free of reflections) and should be held at 0 degrees to (pointed directly at) the sound source.

Pressure microphones are designed specifically for measurements in couplers and closed cavities such as plane wave tubes, ear cavities in dummy heads and for noise measurements in air ducts.

Some SLMs provide a switched correction filter allowing one type of microphone to be used in place of the other.

Note: In general, a “measurement quality” microphone differs from a recording or sound reinforcement microphone in its flat response, the absence of an onboard preamp, and its less rugged construction.

Detector Averaging Time
The standard averaging time choices provided by the majority of SLMs are Fast and Slow.

The Fast setting has a 125milliseconds (mS) averaging time (developed in 1950 to match the ballistics of the moving coil meters used at that time) and is useful for steady state (not dynamic) sound.

For measurement of more dynamic sound (when the Fast setting results in fluctuations of 4 dB or more), the Slow setting provides a 1 second averaging time resulting in much more consistent and stable readings.

It is this setting (along with an A-weighting curve) that is most frequently used for measurement of pop music SPLs.

Some higher priced SLMs also provide Impulse response time (four times faster than the Fast setting) used for extremely dynamic and short term (impulse) sound and peak response time, which accurately responds to signal pulses as short as 50 microseconds.

Frequency Weighting Filters
Because we humans do not hear frequencies in a linear manner (as illustrated in the often referred to Fletcher-Munson curves, aka: equal loudness contours), sound level measurements made with a flat response do not accurately reflect how we perceive sound.

When measuring pop music sound reinforcement systems (with the program signal containing generous amounts of bass energy), broadband readings would be artificially high without the frequency shaping provided by the A-weighting filter.

C-weighting, with its more gradual low-and high shelving filters, is used for very high SPL sound system and noise measurements.

On SLMs without a Flat setting, the C-weighted setting may be used to feed a spectrum analyzer such as Rational Acoustics Smaart for frequency analysis of the broadband sound.

Table 2 below, and then Figure 2 below that, show the frequency response for A and C weighting filters, plus two other weighting filters employed for more exacting and specific noise measurements.

 
All weighting curves are equal in level at 1,000 Hz, the standard reference frequency that SLMs are calibrated to using acoustic calibrators.

Note On SPL Annotation: When communicating or recording (speaking or writing) SPL values, it is necessary to annotate what - if any - weighting curve has been employed. A-weighted SPL is written or stated as “XX.X dBA” and C-weighted measurements as “XX.X dBC”. Such annotation eliminates the need to provide the “SPL” suffix.

The absence of weighting notation implies that a Flat setting (no weighting filter) was used and in this case the suffix “SPL” must be used to differentiate from other, non-acoustic decibel values.

Historical Note: The A, C and B frequency weighting curves were first developed for specific sound levels based on equal-loudness contours.

Because equal-loudness contours were developed from research utilizing pure tones, it later became evident that the weighting curves chosen for SLM measurements at certain levels do not correlate very well to how we actually hear complex sounds.

 
More recently it has been found that the A-weighting curve provides a realistic approximation of how we hear over a fairly wide range of sound level measurements.

Display
With very few exceptions, modern SLM’s are provided with LCD displays with 3- or 4-digit readouts. Most of these provide 1/10-dB resolution except in the case of 3-digit displays when they go over 100 dB-SPL.

Prior to LCD technology, analog meters were employed and these suffered from all the inherent shortcomings of this method of display: slow rise and settle time, susceptibility to damage from pinging (over loading) and overall mechanical fragility.

Another shortcoming of analog meters is the difficulty in establishing stable readings due to the range selecting that is common in most SLMs. One must take the range that is selected and then add or subtract from the meter position accordingly.

Measurements made when the sound pressure level has changed and is out of range are prone to be inaccurate, as well. Some feel that fast impulse noise measurements are easier to read with an analog SLM than with a “digital” SLM.

Both styles of display are difficult to read in dim lighting but some SLM’s feature backlit LCD’s for this condition. And either type of display may be quite difficult to read while at arm’s length if the display size is too small.

Equivalent Continuous Sound Level
Although we live sound folk generally believe that we are able to accurately determine the overall sound pressure level when we view the displayed real time values on our SLM’s, these readings are actually no more than a glimpse, or snapshot, of the varying sound pressure levels that we are exposed to.

Due to the dynamic content of the sounds we normally work with (this applies to both speech and music) and our own human shortcomings, we are unable to mentally integrate the frequently changing SPL’s and determine an accurate mean SPL value.

Research into hearing safety and annoyance and the standards that have been developed from this research are based on the accumulative exposure to sound energy over periods of time.

To this end, integrating SLMs have been developed to provide measurements that are conducted over a selected time period and averaged so that they provide an accurate equivalent energy value, designated as Leq (LAeq or LCeq when weighting is employed).

Integrating SLMs also provide Lmax and Lmin values, and the user can choose the time period that the Leq averaging takes place (normally from several seconds to 24 hours).

More and more organizations and governments are adopting this as a more accurate means for establishing risk of hearing damage, community noise standards, annoyance issues, etc.

Although stand-alone integrating SLMs are considerably more expensive than basic SLM’s, some software is available to provide this statistical averaging from the AC output of any SLM.  and both the TerraSonde Audio Toolbox and Neutrik Minilyzer provide this function along with other SPL measurements and other useful audio system measurements.

Relative Vs Absolute Measurement
Absolute measurements are those made with calibrated SLMs that provide actual sound pressure levels referenced to 20uPa (0 dB-SPL).

Such absolute SPL measurements may be used with and accurately compared to those provided by other calibrated and equal quality SLMs.

Conversely, measurements made with un-calibrated SLMs or those with of low quality (including less then linear frequency response) do not accurately indicate actual sound pressure levels.

SLMs that have been incorrectly user-calibrated or have simply lost their calibration over time also do not provide accurate absolute SPL values.

Relative measurements are simply comparisons between sound pressure levels and do not require absolute calibration nor other “precision” performance characteristics such as a flat frequency response.

Non-calibrated and lower-quality SLMs do provide meaningful comparative measurements as long as the same SLM is employed for these measurements and the measurements are made in the same manner.

Example: A Radio Shack SLM, which is specified as providing +/-2.0 dB-SPL accuracy and which has a less-than linear frequency response, is still able to provide realistic indication of the changes in relative sound pressure levels even though it does not provide accurate measurement of absolute sound pressure levels.

Likewise, the SLM function contained in Smaart may be used “as is” (without calibration) for accurate relative SPL measurement.

One relevant and valid relative SPL measurement that can be made in sound reinforcement work is determining the even-ness of coverage provided by a loudspeaker system.

In this case virtually any SLM is able to indicate any deviations that may exist in the sound pressure levels as the SLM is moved about the coverage area of the loudspeaker system.

In a similar manner we can observe changes in relative SPL as the gain of a sound system is brought up and down and we can accurately determine changes in SPL as a noise source (machinery, for example) is turned on and off.

Available SLMs
As stated previously, SPL measurements made with Class-2 and Class-3 SLM’s provide sufficiently accurate (within +/- 1.0 dB) readings for what we require in sound reinforcement work.

Until recently, SLM devices that met this level of accuracy were prohibitively expensive ($1,200 and up).

Most of the old classic SLM’s are cumbersome to set up and, due to their inherently fragile construction, are seldom used for touring sound or on-site installation work.

Without question the most common SLMs used by live sound folks have been those made by Radio Shack. These appear to represent a very good value, are easy to use and are readily available almost anywhere in North America.

In researching for this article I have found that, along with all of the sound system operators and contractors who own a Radio Shack SLM (or two), there are large numbers of other users including university physics/acoustics and biology departments, municipalities, hi-fi enthusiasts, and theater system installers and property management companies.

And there are at least a few websites devoted to using the Radio Shack SLM for tweaking hi-fi, project studio and sound reinforcement loudspeaker systems and for simple noise measurement.

The original Radio Shack model 33-2050 utilizes an analog meter display and has been in production for the past few decades, and several years ago, the model 33-2055 was introduced, providing a “digital” (alphanumeric) LCD display, recessed and better-protected switches and a maximum/minimum measurement function.

Otherwise, these two models employ the same microphone, base circuitry and battery complement.

 
Alternative SPL Measurement Tools
One alternative to the stand-alone SLM’s described in this article is the SLM function included with some of the better portable RTA’s (real time analyzers) that are often employed for live sound and acoustic measurement.

Models from Ivie, Audio Control, and GoldLine all provide accurate broadband sound level measurement (Class-2 or Class-1, depending on the microphone used) plus octave and 1/3-octave spectrum analysis.

This combination of measurements is very useful for determining both the SPL and the frequency components of the sound/noise under con sideration.

 
Most, if not all, of the current crop of computer-based electroacoustic measurement system provide an SPL function along with their frequency and time-related measurements. Of these, SIM from Meyer Sound and TEF from Goldline are sold with microphones and are calibrated at the factory.

Smaart (any version), SpectraFoo, SysSid, PC-RTA and other similar systems utilize third-party microphones and preamps that must be calibrated by the end-user, using an acoustic calibrator or piston.

Once calibrated, these systems provide very good accuracy and feature a number of variables (weighting filters, etc).

Unfortunately, they are not really “handheld” nor nearly as portable as stand-alone SLMs. And (most importantly) they do not retain their calibration from site to site and therefore require an acoustic calibrator for continued use in measuring absolute SPL’s.

Another product that is a variation of a SLM deserves mention here. SPL “management” systems have been available for perhaps a decade and are directly related to the issues of sound system volume levels.

These simply employ microphones and a software-based SLM system that provides either an obnoxious visual warning or more extreme reactive functions (such as muting of the sound system signal!) when the predetermined SPL limit has been reached or surpassed.

Calibration
In situations where the accuracy of sound level measurements must be recorded and/or verified, it is necessary to calibrate the measurement system both before and after the measurements are made.

Software-based electroacoustic measurement systems that feature an SLM function also need to be calibrated if used for absolute SPL measurements.

An acoustic calibrator provides the standard 94 dB or 114 dB (or both) 1 kHz test tone and should also include the proper size opening or adaptor rings for the microphone(s) you intend to employ.

As is the case with SLMs, calibration devices are manufactured to several levels of tolerance (see table near beginning of article) and they have historically been prohibitively expensive.

Recently there have been several acoustic calibrators introduced that provide very good accuracy for sound system and acoustics measurement.

SLMs may also be calibrated with a pistonphone, but the cost of this device is prohibitively high and the high level of accuracy it provides is not practical for most general sound system related measurements.

Why We Use Them
The simple and snappy answer as to why we use SLMs: so we can know what’s going on.

The need for accurate and repeatable sound level measurement is due simply to the tendency for humans to aurally perceive sound in a very subjective manner. This applies to all aspects of sound perception in the audio field: frequency content, mix “correctness”, system EQ, dynamics, etc.

At some point or another we will disagree on the relative loudness being produced plus these other subjective sound qualities.

Whereas other electroacoustic measurement tools remain very subjective in how we interpret the data they provide, a reasonably accurate SLM provides an irrefutable means to objectively verify how loud the sound is (or is not).

SLM measurements may also be used to verify exactly how much the sound level has been changed (as noted in the section on relative versus absolute measurement).

Anyone involved in live sound system work is aware that there is an ongoing and increasing problem with the issue of volume levels produced in public performance, and specifically, in pop music reinforcement.

It’s wise for anyone involved in this field to become familiar with the actual measured numbers, not to mention the need to continually monitor what we are doing. Issues of personal health and legal liability are an ongoing and increasing concern.

Measurement of unintended noise levels and/or the effectiveness of acoustic treatment are additional exercises that most live sound folks eventually will become involved in.

How We Use Them
For basic every day sound system level measurement, using an SLM is fairly straightforward.

The standard SLM employs a free field microphone and is intended to be held at arm’s length and tipped up slightly towards the sound source.

As you move away from the source and into the reverberant or diffuse field, or when measuring in a multiple source sound field such as on stage, the angle of incidence of the microphone becomes much less of a concern.

When measuring noise levels at installation sites, some care should be taken. If measuring the noise level of a rack room, for example, be sure that the SLM is positioned away from walls, floors and other large boundaries.

Signal-to-noise levels are seldom an issue for loudspeaker system measurement because we can simply turn the system up, more than far enough above the noise floor.

But when measuring the noise from a loudspeaker system (hum, hiss and/or the fan noise that may exist in self-powered systems), or from a mixing console and its power supply, etc., or from other common noises such as HVAC systems, vacuum cleaners, power tools, etc., the unintended noise must be at least 10 dB-SPL lower than the noise being measured.

When measuring and logging levels to help resolve a dispute, both A and C weighted measurements should be provided, as should the distance from the source and any other detailed notes.

Also don’t forget that a non-annotated measurement (“00.0 dB-SPL”) will be interpreted to be flat (unweighted).

General Measurement Guidelines
When conducting sound pressure level measurements there are a few basic rules that apply:

• Keep the SLM at arm’s length, aimed in the direction of the sound source(s) and positioned away from room boundaries. By the way, the mixing surface on a large (40-plus channel) mixing console will effect SPL measurement when the SLM is laid down on the mixing surface or angled against the meter bridge.

• For almost all sound system measurements, use the A-weighting filter and Slow response setting.

• Be sure that you’re measuring within the range that you have selected. SLMs with multiple range positions and those with analog displays, in particular, may provide incorrect readings when the SPL is over/under the selected range.

• Prior to making SPL measurements, ensure that the SLM has been turned on for a few minutes and has reached room temperature.

• Wind and air-blowers will effect SPL measurements.

• Be aware of where you are taking your measurements. For example: if you measure from within the area where the bass energy from the sound system has combined (aka “power alley”) or within a standing wave null, the readings will be non-representative of the SPLs for the majority of the audience.

Tom Young is principle consultant for Electroacoustic Design Services in Connecticut, and he has designed hundreds of systems for churches and similar spaces. Tom’s also the moderator of the Church Sound Community Forum here on ProSoundWeb.

More articles by Tom Young on PSW:
Time Is On Your Side (If You Want It To Be) With Sound System Alignment
Microphone Techniques For Drums In Contemporary Worship Environments
The Aux-Fed Subwoofer Technique Explained

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Thursday, January 26, 2012

PreSonus StudioLive Mixers Now Outfitted With Smaart System Analysis Tools

PreSonus has begun incorporating Rational Acoustics Smaart Measurement Technology directly into the software used to control its StudioLive digital mixers.

PreSonus Virtual StudioLive (VSL) remote-control/editor/librarian software will now incorporate Smaart Spectra and Smaart Locator, powerful tools for sound-system analysis and optimization, as part of PreSonus Universal Control 1.6, expected to be available later this spring.

Smaart is not a single technology but an evolved collection of audio measurement tools and techniques. Using Smaart technology, users can tap into the power of the StudioLive mixer’s EQ to improve the sound of their system.

With Smaart-enhanced VSL users can view the spectral content of their mix in real time, and easily make changes.

Clicking on the Graphic Equalizer button in Universal Control 1.6, Smaart Spectra’s Real Time Analyzer activates Spectrograph’s algorithms, displaying the spectral content of whatever is routed though a particular graphic EQ.

Users can also activate a Real Time Analyzer, much like the plug-in used in PreSonus’ Studio One 2.

In addition, the Smaart Spectra Spectrograph display can help to precisely identify feedback frequencies, enabling even less experienced users to more easily tune their loudspeakers to the room.

Smaart Spectra graphs a continuous series of spectrum measurements, showing frequency on one axis, time on another, and level indicated by colors - making it particularly useful for quickly identifying feedback frequencies, which can be easily addressed using StudioLive GEQs.

“This is just the start of a beautiful relationship - PreSonus products are about to get a lot Smaarter,” says PreSonus chief technology officer Bob Tudor. “This is the real thing, trusted by acousticians and live sound engineers the world over. We’re thrilled to be incorporating Smaart Spectra and Smaart Locator into Virtual StudioLive.”

PreSonus

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Wednesday, January 25, 2012

Loudspeaker Sensitivity: What’s A Watt Anyway?

Shedding some light on the sensitivity specification and how it may translate to the real world performance of a loudspeaker system

The specification of a loudspeaker’s sensitivity is probably one of the most common, yet perhaps one of the most misunderstood.

It’s common to see the magnitude response of a loudspeaker system reduced to a single number as a sensitivity rating.

This is perhaps at the heart of the confusion.

One would think that this metric should give some indication as to how loud a particular loudspeaker will be when reproducing a signal.

One may also think that two loudspeakers with the same sensitivity rating will be equally loud when reproducing the same signal. Each of these assertions is only partially true.

A loudspeaker’s sensitivity can give an indication of its output level but only for a signal with a specific bandwidth and spectral content.

Similarly, two loudspeakers with the same sensitivity may not output the same SPL when excited by the same signal if the frequency response limits of the two loudspeakers are different. Let’s look at the underlying cause of each of these effects, bandwidth, and the role it plays, and also look at why sensitivity may no longer need to be referenced to a watt.

According to the standard IEC60268-5, a loudspeaker’s sensitivity is determined by measuring its output when driven by a band limited pink noise signal with a Vrms equal to the square root of the loudspeaker’s rated impedance and referencing this SPL to a distance of 1 meter.

The bandwidth of the pink noise is limited as a function of the effective frequency range of the DUT (Device Under Test). This is done to ensure that the test signal is confined to a portion of the frequency spectrum in which the DUT has appreciable output.

If a particular loudspeaker isn’t capable of reproducing signals below 150 Hz it does no good to excite it with such signals other than to generate heat. The same holds true if the loudspeaker can’t reproduce signals above some high frequency limit.

A high-resolution transfer function measurement of the DUT can also produce an identical sensitivity rating when the average magnitude is calculated on a log frequency basis.

As an example, let’s look at Figure 1. Here we see the on-axis response of a loudspeaker. Its sensitivity rating is shown as the straight line.

The length of this line coincides with the upper and lower frequency limits of the pink noise used to measure the sensitivity rating.

The spectral content of this noise signal is shown in Figure 2.

If a signal with different spectral content, but the same broadband level were used to drive this loudspeaker, would it result in the same SPL as the sensitivity?

It’s impossible to determine this without knowing both the spectral content of the signal and the response of the loudspeaker. (Note that 20 Hz to 20 kHz, or in the case of Figure 1, 110 Hz-8.3 kHz, does not specify the response of a loudspeaker. A graph of the response curve really needs to be known.)

With knowledge of these, we can certainly make an estimate to answer this question.

The spectral content of three different signals is shown in Figure 3.

One is the band- limited pink noise signal used to determine the sensitivity of the loudspeaker. The others are speech and a shaped noise signal having approximately the same spectral content as the speech. This speech-shaped noise is used instead of speech as its RMS level is more consistent as a function of time than actual speech.

Thus, it will be easier to determine the SPL output by the DUT with this signal. All three signals have approximately the same broadband RMS level. From approximately 200-800 Hz the speech-shaped noise signal has greater level than the pink noise signal.

Above and below this frequency region the pink noise signal has much greater level than the speech-shaped noise signal.

Comparing this to the response of the loudspeaker in Figure 1 we see that the loudspeaker has limited output below 150 Hz. The greatest output in the response of the loudspeaker occurs in the 300 Hz-3 kHz region.

If the speech-shaped noise signal were used to drive the loudspeaker with the same broadband level as the noise we could reasonably expect the broadband SPL to be greater than when driven with the pink noise signal.

This is exactly what happens.

The sensitivity of the loudspeaker is 97.1 dB. When driven with the speech-shaped noise the SPL is 98.1 dB, an increase of 1.0 dB.

This results from the higher level of the speech-shaped signal in the frequency region where the loudspeaker has higher output capability compared to the rest of its pass band.

Conversely, if the low-frequency band-limited pink noise shown in Figure 4 were used to drive the loudspeaker it is reasonable to expect that the SPL would be less than when driven by the noise signal.

This results from the low-frequency pink noise signal having a higher level in the frequency region where the loudspeaker has lower output capability.

The SPL produced by the low-frequency pink noise is 94.9 dB, a decrease of 2.2 dB.

Now let’s compare two different loudspeakers. Figure 5 shows loudspeaker A compared to loudspeaker B. Notice that they both have the same sensitivity, 97.1 dB.

Loudspeaker B, however, has greater low frequency and high frequency extension than loudspeaker A.

Because of this the bandwidth of the pink noise used to determine the sensitivity of loudspeaker B is greater than the bandwidth of the noise used for loudspeaker A (Figure 6).

As a result, the mid-band level of the noise for loudspeaker B is slightly less than that of the noise used for loudspeaker A. It’s a bit difficult to see but upon careful observation the black trace can be seen to be an average of 0.5 dB below the red trace from approximately 100 Hz-10 kHz.

This is due to the greater bandwidth of the signal used for loudspeaker B (black trace). Remember that the broadband levels of both these signals are identical.

So what happens when each of these loudspeakers is driven by the broadband pink noise signal (20 Hz-20 kHz) also shown in Figure 6? As each of the loudspeakers used in this example are markedly not flat in their mid-band response there may be some tonal, and potentially measurably, differences in the SPL.

Hopefully, the reader can put these issues aside for the moment. All other things being equal, the loudspeaker with the greater effective frequency range (low- and high-frequency extension) should have greater SPL output.

Loudspeaker B should have slightly greater output when driven by this broadband pink noise signal. In fact, loudspeaker B measured 0.8 dB greater than loudspeaker A, 97.0 dB compared to 96.2 dB.

From these examples one should be able to see that the SPL generated by a loudspeaker is a function of both the loudspeaker’s transfer function and the spectrum of the signal being reproduced.

Several acoustical room modeling programs take this into account when calculating the SPL produced over an intended audience area. They may allow for the selection of pink noise, some sort of speech spectrum, or a user-defined spectrum.

This should aid the sound system designer, while still at the drawing board stage, to better understand the potential SPL capabilities of the sound system with the typical program material the system is likely to be reproducing.

The other item I mentioned at the beginning of this article was referencing sensitivity measurements to one watt being dissipated by the DUT. There are several reasons why I think that this is not beneficial with modern sound systems.

First, it is somewhat cumbersome to determine how much voltage is required across a particular DUT such that the input current drawn from the driving source yields 1 watt. This can be done using dual channel FFT measurement systems and an appropriate current monitor or probe.

But would this give us useful information for the design and/or specification of loudspeakers or sound systems?

We can simplify this measurement procedure so that we don’t concern ourselves with the dissipation of a real watt by the DUT.

Instead we apply a voltage across the DUT that would dissipate one watt in a pure resistance having the value of the rated impedance of the DUT.

This certainly is easier, but again, does this give us useful information for the design and/or specification of loudspeakers or sound systems? Perhaps.

My thought is that more useful comparative information would be gained by applying the same voltage across the DUT regardless of its impedance.

The majority of amplifiers used in sound systems today are of a constant voltage type. That is to say, their output voltage remains constant independent of the load placed on them. Of course, the load must be within the specified operational limits for a given amplifier.

The salient point is that for a given drive voltage, a lower impedance loudspeaker will have greater SPL output than a higher impedance loudspeaker; all other items being equal.

Shouldn’t this be reflected in the sensitivity specification of the loudspeaker? Why then would one want to use a 2.0 Vrms signal to drive a 4-ohm loudspeaker and a 2.83 Vrms signal to drive an 8-ohm loudspeaker to determine their respective sensitivities?

Think about it this way; let’s connect two virtually identical loudspeakers to an A/B selector switch driven by the same amplifier.

The only difference between these loudspeakers is that one is half the impedance (rated at 4 ohms) than the other (rated at 8 ohms).

When switching between these two loudspeakers the output voltage of the amplifier does not change, however, the current drawn from the amplifier does.

This results in the loudspeaker with the lower rated impedance producing greater SPL.

Measuring and specifying sensitivity with the same voltage, regardless of the impedance of the DUT, would accurately reveal the SPL differences that occur.

From these examples, I hope that it’s clear that the input signal and the magnitude (frequency) response of a loudspeaker will determine the SPL generated, not just the sensitivity rating of the loudspeaker.

It’s much better to have knowledge of the loudspeaker’s response in the form of a graph than a single sensitivity number. The latter may be derived from the former.

Charlie Hughes has worked at Peavey Electronics and Altec Lansing. He currently heads up Excelsior Audio Design & Services; a consultation, design and measurement services company based near Charlotte, NC. Charlie is a member of the AES, ASA, CEA and NSCA. He is an active member of several AES and CEA standards committees.

More articles by Charlie Hughes:
Using All-Pass Filters To Improve Directivity & Magnitude Response
Loudspeaker Measurement: An Overview Of EASERA SysTune
Using Limiters To Enhance LF While Still Keeping Things Under Control

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Monday, January 23, 2012

Measurements Without Computers: Leave The PC At Home And Get Better Data

Combining the hearing process with the measurement process, allowing the benefits of each to be exploited

It is difficult to describe how something sounds.

The terms are vague and subjective, and often have different meanings to different people.

Sound is meant for listening, and the listening process is the most power diagnostic tool for the audio technician.

Measurement systems yield lots of data, but it can be difficult to interpret.

It makes perfect sense to combine the hearing process with the measurement process, allowing the benefits of each to be exploited.

The process of convolution allows measured data to be evaluated by listening. The method that I use requires the room impulse response (RIR) to be saved as a Wave file, the default file format for several measurement applications. The RIR WAV file can be listened to in a wave editor, or convolved with anechoic program material in a special program (read on!).

Of course, all of this requires a personal computer. This can be a hassle, as computers can be awkward and unwieldy when used to evaluate different listener locations. The setups can be time consuming, requiring very long cables, interface boxes, etc.

Ironically, the power of the personal computer can allow the RIR to be gathered without it. Here is one method to gather the system RIR without a PC, and how to post-process the data to evaluate it and listen to it.

The Signature
The RIR is without a doubt the single most important acoustical measurement.

It’s also one of the oldest, having been implemented by hand claps, balloon bursts, starter’s pistols and even yachting canon fire. Within it lies a wealth of information about the sound system and acoustic environment.

The principle is simple – whatever the room does to your hand clap, it will also do to your voice, or to a musical instrument played at that location in the space. The RIR is the “signature” that the room places on the sound.

Once the RIR is gathered at a listener position, it can be analyzed to reveal information about the direct sound, early and late reflected fields, and the diffuse field of the room.

If the sound system is used to excite the room, the RIR will also contain information about the loudspeaker’s response and resultant performance as it relates to speech intelligibility and music clarity.

In short, most system/room characteristics that can be heard at a listener position will be included in the RIR for that position.

The freeware GratisVolver (www.catt.se) can be used to convolve the RIR with anechoic music or speech.

“Convolution” means to take one file (time or frequency data) and encode it with the characteristics of another (time or frequency data).

If one file is dry program material, and the other is an RIR gathered in a room, the convolution will yield what the dry program material would sound like when encoded with the acoustic characteristics of the live space for that specific position.

Ideally, the RIR should be two channels gathered with a stereo microphone. This preserves many of the localization cues required by listeners to pinpoint a sound in three-dimensional space.

The benefits of convolution are obvious:
• Speech or music can be evaluated without making actual recordings of talkers or instruments.

• The RIR can be modified to simulate acoustical changes to the space. Convolution can then be used to evaluate the effects of the changes.

• There are signal-to-noise advantages over recordings. If the RIR is gathered in a noise-immune manner, then the convolutions will have a lower noise floor than actual recordings made in the space.

• The RIR is a complete documentation of the room/system response at one listener position. It can be reprocessed in the future as other processing algorithms become available.

Regardless of what measurement platform you use, a properly gathered RIR should have the following characteristics:

• When displayed on a dB vertical scale, the data should fill the screen from top left to bottom right.

• The full decay of the system must be measured, meaning that the decaying tail must not be truncated.

• For convolution purposes, good signal-to-noise ratio is required. I like to have at least 90 dB of dynamic range, and see the decaying energy fade into the noise floor at -80 to -90 dBFS. Less can work, but there may be some artifacts in the playback.

Applications & Processes
Deconvolution is the opposite of convolution.

A known stimulus is played into a room and the response is recorded.

The difference between the two files is the RIR, which represents what the room/system did to the sound as it passed from source to receiver.

One way that the RIR can be obtained is by a complex mathematical process called deconvolution. A number of software applications exist that can deconvolve two files.

GratisVolver can yield the RIR by convolving a sweep recorded in the room with an inverse version of the original sweep. This is deconvolution.

Using deconvolution, you can get the RIR without using an impulsive stimulus.

A slowly swept sine wave offers some significant advantages:

• Significantly more energy is fed to the room. This offers dramatic signal-to-noise benefits.

• It’s less likely that you will overload your recorder, since the level of a sine wave doesn’t fluctuate like a noise or impulsive stimulus.

• The simple nature of the waveform allows the recording to be done with most file formats (i.e., WAV, MP3). Complex waveforms can be significantly altered by the compression schemes used by some formats.

• It is relatively easy to set recording levels.

Beware. Since sine waves generate significant power (that’s why you get a better signal-to-noise ratio), be careful not to burn up the loudspeakers used for the test.

The Equipment
There are lots of possibilities here. My setup consists of things that I already had. GratisVolver includes the forward and inverse sweeps needed for the test. You will need to supply a few hardware items.

For my stimulus file, I created a WAV file that includes the 14-second forward sweep and a short speech track. The total duration is about two minutes.

I use a media player (i.e., iPod) to play the file into the system. As a separate unit, it eliminates the need to string long cables in the venue to excite the system. The file is loaded on to a stereo portable handheld recorder. My mic is a Crown SASS, ideal for this application since it emulates a human listener. All of this fits neatly in the SASS case.

The data is gathered by playing the sweep file into the system and simply recording it. The 14-second log sine sweep is recorded at each desired measurement position. The speech track is used to provide a reference for what the measurement position sounded like. It can later be used as a reference to compare the convolved files to. The whole process takes about two minutes per listener location.

The Possibilities
Now the fun begins! The RIR can be opened in any acoustics measurement package that recognizes the WAV file format, which is about all of them and processed to yield the various acoustic measures.

It can be convolved with dry program material in GratisVolver. The user is free to exploit the strengths of all of these platforms (plus their ear-brain system) to analyze the data. It also allows for easy comparison of the measurement platform algorithms such as reverb time and clarity scores.

You will learn which measures are more meaningful to the way that you think about sound, and that help you draw meaningful conclusions from the data. Such files can be exchanged with colleagues for consultation, and can serve to document the before - and - after the performance of a sound system renovation.

They are also excellent for archival purposes. How would you like to have an RIR of the Fogg Lecture Hall where Sabine derived his famous equation? We could return to that space virtually and listen to it, as well as many others.

RIR testing is nothing new. We just have a few new tools that help us collect and analyze. All serve to make us better measurers and better listeners, ultimately resulting in better sounding systems.

Pat & Brenda Brown lead SynAudCon, conducting audio seminars and workshops around the world in addition to providing web-based training at www.synaudcon.com.

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Monday, December 19, 2011

Equalizing the Room - What it Really Means

You can't change the room. You can, however, equalize the response of the loudspeaker system. More than semantics, equalization has very real practical consequences.

“I am going to equalize the room.” We’ve all heard that statement so many times that we scarcely think about what it literally means. We know that in practical terms it means adjusting an equalizer to suit your taste. It may be done with the latest high-technology analysis equipment, voodoo magic or simply tweaking away “until it sounds right.”

In any case, are we really “equalizing the room”? What exactly are we doing? There are lots of disagreements on this topic but all agree on one thing: You cannot change the architecture of the room with an equalizer.

You can, however, equalize the response of the speaker system. Where the room fits into all this is a matter of debate; It is much more than semantics and has very real practical consequences on our approach to sound system alignment.

What do equalizers “equalize” anyway?
Let’s assume that we have a speaker system with a flat (or otherwise desirable) free field frequency response. That is to say, it requires no further equalization. There are three categories of interaction that will cause the frequency response to change, to become, for lack of a better word, “unequalized.”

The first of these interactions are between speakers. When a second speaker is added the combination results in a modified frequency response at virtually all locations. This is true of all speaker models and all array configurations, regardless of any claims to the contrary.

The summation of the two responses varies the frequency response at each position, depending upon the relative time arrival and level between the two speakers. As additional speakers are added the variations in response increase proportionally.

The second category is the interaction of the speaker(s) with the room. These are generally termed coupling, reflections or echoes. The mechanism is similar to the speaker interaction above. The response varies from position to position, depending upon the relative time arrival and level between the direct and reflected sound.

Both of the above effects are the result of a summation in the acoustical space of multiple sources, either speaker and speaker, or speaker and reflection. Therefore the solutions for these interactions are very closely related.

The third interaction is caused by the effects of dynamic conditions of temperature, humidity and changing absorption coefficient. However, the effects of these interactions are small by comparison with the other two, so we will not touch on them further here.

Are any of these problems solvable with an equalizer? The answer is a qualified “Yes”. The magnitude of the above problems can be reduced by equalization, and substantial progress can be made toward restoring the original desirable frequency response.

If equalizers were totally ineffective, then why have we been loading these things into our racks for the last 35 years? However, in a practical sense the equalizer can only provide complete success in equalizing the response when applied in conjunction with other techniques such as architectural modification, precise speaker positioning, delay and level setting.

To what extent is the speaker/room interaction equalizable? This has been a matter of debate for more than 15 years. In particular the advocates of various acoustical measurement systems have come down hard on these issues.

What we are doing is equalizing, among other things, the effects of the room on the speaker system. Why is this controversial? It stems from the historical relationship of equalizers and analyzers. Let’s turn on the way-back machine and take a look.

Early analysis
In ancient times (the 1970s), the alignment of sound systems centered around a crude tool known as the Real-Time Analyzer (RTA) and a companion solution device, the graphic equalizer. The analyzer displayed the amplitude response over frequency in 1/3 octave resolution and the equalizer could be adjusted until an inverse response was created, yielding a flat combined response.

It takes a negligible skill level to learn to fiddle with the graphic EQ knobs until all the LEDs line up on the RTA. It is so simple that a monkey could do it, and the result often sounded like it.

Although these tools were the standard of the day, they have severe limitations, and these very limitations can lead to gross misunderstanding of the interaction of the speakers to each other and the room, resulting in poor alignment choices.

One such limitation is the fact that the RTA lacks information regarding the temporal aspects of the system response. There is no phase information nor any indication as to the arrival order of energy at the mic.

The RTA cannot discern direct from reverberant sound, nor does it indicate whether the response variations are due to loudspeaker interaction alone and loudspeaker/room interaction. Therefore the RTA provides no help in terms of critical speaker positioning, delay setting or architectural acoustics.

Second, the RTA gives no indication as to whether the response at the mic is in any way related to the signal entering the loudspeakers. The RTA gives a status report of the acoustical energy at the microphone, with no frame of reference as to the probable causes of response peaks and dips.

These peaks and dips could be due to early room reflections or speaker interactions, which can respond favorably to equalization. However, the irregularities in response could be from late reflections, noise from a forklift engine or reflections from a steel beam in front of the loudspeaker.

The equalizer will be ineffective as a forklift or steel beam remover, but the RTA will give you no reason to suspect these problems. A system that is completely unintelligible could look the same as one that is clear as a bell.

Third is the fact 1/3-octave frequency resolution is totally insufficient for critical alignment decisions. In addition, there is the misconception that a matched analyzer/filter set system is desired. It is not. The analyzer should be three times the resolution of the filter set in order to be able to provide the visible data needed to detect center frequency, bandwidth and magnitude of the response aberrations.

A 1/3 octave RTA is only able to reliably determine bandwidths of an octave or more. What appears as a 1/3 octave peak may be much narrower. What appears as a broad 2/3 octave peak, may actually be a high narrow peak placed between the 1/3 octave points. What will your graphic equalizer do with this?

Unfortunately the absence of this critical information lulled many users into a sense of complacency predicated on the belief that equalization was the only critical parameter for system alignment. In countless cases, equalizers were employed to correct problems they had no possibility of solving, and could only make worse.

Graphic equalizers have no possibility of creating the inverse of the interactive response of the speakers with the room. Simply put: “You can’t get there from here.”

The audible results of all this tended to create a generally negative view of audio analyzers. Many engineers concluded that their ears, coupled with common sense could provide better results than the blindly followed analyzer.

As a result, though RTAs were often required on riders, they only received cursory attention on show day.

Modern Analysis
Technological progress led to the development and acceptance of two analysis techniques in the early 80s: Time Delay Spectrometry (TDS) and dual-channel FFT analysis. Both of these systems brought to the table whole new capabilities, such as phase response measurement, the ability to identify echoes and high-resolution frequency response.

No longer could an unintelligible pile of junk look the same as the real McCoy on an analyzer. The complexity of these analyzers required a well-trained, highly skilled practitioner in order to realize the true benefits.

Advocates of both systems stressed the need for engineers to utilize all tools in their system, not equalizers alone, to remedy the response anomalies. Delay lines, speaker positioning, crossover optimization and architectural solutions were to be employed whenever possible. And now we had tools capable of identifying the different interactions.

But on the issue of “equalizing the room” a division arose. All parties agreed that speaker/speaker interaction was somewhat equalizable. The critical disagreement was over the extent the loudspeaker/room interaction could be compensated by equalization.

The TDS camp advocated that speaker/room interaction was not at all equalizable and therefore, the measurement system should screen out the speaker/room interaction, leaving only the equalizable portion of the loudspeaker system on the analyzer screen. Then the inverse of the response is applied via the equalizer and that was as far as one should go.

The TDS system was designed to screen out the frequency response effects of reflections from its measurements via a sine frequency sweep and delayed tracking filter mechanism, thereby displaying a simulated anechoic response. The measurements are able to clearly show the speaker/speaker interaction of a cluster and provide useful data for optimization.

Such an approach can be effective in the mid and upper frequency ranges where the frequency resolution can remain high even with fast sweeps but it is less effective at low frequencies. Low frequencies have such long periods that it is impossible to get high-resolution data without taking long time records, thereby allowing the room into the measurement.

For example, to achieve 1/12th octave resolution, the equivalent to the Western Tempered Scale, one must have a time record 12x longer than the period of the frequency in question. For 30 Hz you will need a 360ms (12x30ms). If fast sweeps are made to remove echoes from the measurement, the low frequency data has insufficient resolution to be of practical use.

Dual-channel FFT analyzers utilize varying time record lengths. In the HF range, where the period is short, the time record is short. As the frequency decreases, the time record length increases, creating an approximately constant frequency resolution.

The measurements reveal a constant proportion of direct sound and early reflections, the most critical area in terms of perceived tonal quality of a speaker system.

The most popular FFT systems utilize 1/24th octave resolution, which means that the measurements are confined to the direct sound and the reflections inside a 24 wavelengths time period across the board. This is a good practical level of resolution, allowing us to accurately equalize at around the 1/8 octave level.

With the FFT approach, more and more of the room enters the response as frequency decreases. This is appropriate because at low frequencies the room/speaker interaction is still inside the practical equalizability window.

For example, the arena scoreboard reflection is 150 ms later than the direct signal. At 10 kHz, the peaks and dips from this reflection are spaced 1/1500 of an octave apart. At 30 Hz, they will be only 1/3 octave apart. Thus the scoreboard is in the distant field relative to the tweeters, and applying equalization to counter its effects will be totally impractical.

An architectural solution such as a curtain would be effective. But for the subwoofers, the scoreboard is a near-field boundary and will yield to filters much more practically than the 50 tons of absorptive material required to suppress it acoustically.

Many years ago, the FFT camp boldly stated that the echoes in the room could be suppressed through equalization. Unfortunately, these statements were made in absolute terms without qualifying parameters, leaving the impression that the FFT advocates thought it was desirable or practical to remove all of the effects of reverberation in a space through equalization.

While it can be proven from a theoretical standpoint that the frequency response effects of a single echo can be fully compensated for, that does not mean it is practical or desirable. The suppression can only be accomplished if the relative level of the echo does not equal or exceed that of the direct and that no other special circumstances arise that cause excess delay. (Excess delay causes a “non-minimum phase” aberration and is outside the scope of this article.)

If the direct level and echo level are equal the cancellation dip becomes infinitely deep and the corresponding filter required to equalize it is an infinite peak. As we know from sci-fi movies, bad things happen when positive and negative infinity meet up.

Compensating for the response requires adjustable bandwidth filters capable of creating an inverse to each comb filter peak and dip in the response. As the echo increases, you will need increasing numbers of ever narrowing filters.

A 1ms echo corrected to 20 kHz will require some 40 filters because there are 20 peaks and 20 dips varying in bandwidth from 1 to .025 octave. A 10 ms echo would need 400 with bandwidths down to an 1/400 octave.

Obviously, it would be insane to attempt to remove all of the interaction at even a single point in the hall. In the practical world, we have no intention of attacking every minuscule peak and dip, but instead will go after the biggest repeat offenders. The narrower the filters are, the less practical value they have because the response changes over position.

Practical Implications
It is indeed possible and practical to suppress some of the effects of speaker/room interaction. If this was not possible, it would be standard practice to equalize your rig in the shop, put a steel case around the EQ rack and hit the road. The practical side of this is that we must be realistic about what is attainable and what are the best means of getting there.

The variations in frequency response due to both speaker/speaker interaction and loudspeaker/room interaction will always change with position. Once you have seen high-resolution data at multiple positions, you can never go back to thinking that your equalization will solve problems globally.

A system that has the minimal amount of the above interactions will have the greatest uniformity throughout the listening environment and, therefore, stand to gain the most practical benefit from equalization. If it sounds totally different at every seat, let’s just tweak the mix position and head to catering.

To minimize the speaker/speaker interactions requires directional components, careful placement and precise arraying. In areas where the speakers overlap, time delays and level controls will minimize the damage in the shared area. To minimize loudspeaker/room interaction, the global solutions lie in architectural modification (it’s curtain time), the selection of directionally controlled elements and precise placement.

Finally you are left with equalization. For each subsystem with an equalizer, map out the response in the area by placing a microphone in as many spots as you can and see what the trends are.

In particular, measure around the central coverage area of the speaker. Stay away from areas of high interaction, where the response will vary dramatically every inch.

Examples of this include the seam between two cabinets in an array or very close to a wall. Each position will be unique, but if you place filters on the top four to six repeat offenders you will have effectively neutralized the response in that area.

Conclusion
Modern analyzers are capable of displaying a dizzying array of spectral data. But little practical benefit will come to us if we continue with the antiquated approach of the RTA era. To fully take advantage of the benefits of equalization, we must fully comprehend how to identify the mechanisms that “unequalize” the system.

With modern tools, it becomes possible to analyze the response such that the interactive factors of speaker systems can be distilled and viewed separately. This allows the alignment engineer to prepare the way for successful equalization by using other techniques that reduce interaction and maximize uniformity in the system.

“Equalizing the room” will remain in the domain of architectural acousticians, but with advanced tools and techniques, we can proceed forward to better equalize the speaker system in the room.

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Rational Acoustics Debuts New Version Of RTA-420 Measurement Microphone

Offers heavy-gauge steel construction and measurement performance similar to high-end mics costing many times more

The new version of the Rational Acoustics branded RTA-420 is a budget-friendly measurement microphone with an impressively flat response curve for its price, and as a result, it works well as a “starter” mic, a spare, or a “B” rig mic for use in tough conditions where the user might not want to deploy a more expensive mic.

The RTA-420 offers heavy-gauge steel construction and measurement performance similar to high-end mics costing many times more.

It is supplied with a mic clip and a bright yellow windscreen, which is helpful in seeing the mic in dark venues. It also includes a custom-designed soft-sided carrying case that accommodates and organizes two mics plus their accessories.

Like the previous version RTA-420, the new model is PIN 3 HOT.

The RTA-420 is available for purchase via the Rational Acoustics online store, at a retail price of $89 (USD). In addition, customers can contact their Rational Acoustics dealer or distributor for local pricing and availability.

Rational Acoustics

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Renkus-Heinz Announces 2012 EASE Training Class Schedule

All classes are based on EASE 4.3 for Windows

Renkus-Heinz announced the EASE Level 1 Training Class schedule for 2012.

The classes, conducted by Renkus-Heinz application engineers Jonas Domkus and Jim Mobley, cover the basics of using EASE 4.3 and EASE JR 4.3, with specific focus on room modeling techniques and basic electro-acoustic analysis.

Each three-day session is held at the EASE Learning Lab, located at the Renkus-Heinz headquarters in Foothill Ranch, CA, with classes offered quarterly throughout the year.

All classes are based on EASE 4.3 for Windows. Ownership of and basic familiarity with EASE 4.3 is a prerequisite, and students are encouraged to bring their own laptops.

2012 classes are scheduled on February 20-22, May 14-16, August 20-22 and November 12-14, 2012.

Go here for more information and/or to register for the courses.

Renkus-Heinz

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Tuesday, December 13, 2011

Rational Acoustics Introduces Smaart I-O Measurement-Grade 2×2 USB Audio Interface

Designed and built specifically for use with Smaart v.7 measurement software

Rational Acoustics has just introduced the new Smaart I-O, a measurement-grade 2×2 USB audio interface designed and built specifically for use with Smaart v.7 measurement software. 

The I-O features two high-quality, active balanced inputs with 50 dB of computer adjustable gain in precision 1 dB steps. These input gains are monitored directly by Smaart allowing the user to retain accurate SPL calibration while varying measurement signal input levels.

The inputs employ a Neutrik combo jack (XLR – 1/4-inch) to accommodate both mic and line level input signals, each with switchable 48-volt phantom power on the XLR inputs for measurement mics and a 20 dB pad on the 1/4-inch to accommodate line level signals.

The Smaart I-O also features two active balanced XLR outputs capable of providing +8.2 dBu (2 Vrms) max level signal for playback or excitation sources.

Powered by the USB port of your computer (or via the optional 5 VDC jack), the Smaart I-O provides a compact and portable measurement-quality input for a Smaart rig.

Smaart I-O Features

Two (2) Neutrik XLR – 1/4-inch combo jack inputs
—Computer controllable 50 dB analog gain in precision 1 dB steps
—Max input (at minimum gain): +6 dBu (XLR),  +26 dBu (1/4-inch)
—Active balanced with 2k ohm impedance (XLR), 65k ohm (1/4-inch)
 
Two (2) Active Balanced XLR Outputs
—150 Ohm impedance
—Max Output : +8.2 dBu (2.0 V rms) w/ 100k ohm load
 
Frequency Response
—Magnitude :  16 Hz – 20 kHz, +/- .25 dB
—Phase:  16 Hz – 20 kHz, +/- 10 deg
 
Sample rate clocks of multiple I-O’s can be linked to allow for device aggregation in Smaart’s dual-channel measurements.
 
Bus powering via USB for convenient, single-cable connection to the computer.
 
Optional 5 VDC jack in cases of insufficient USB power
 
Dimensions:  1.75-in (h) x 7.25-in (w) x 4.25-in (d)

For more info about the new Smaart I-O, and to download the control application for Smaart v7, click here.

Rational Acoustics
 

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Monday, December 12, 2011

First-Ever Live Sound International Compact System Demo Proves A Big Success

While the 12 participating manufacturers primarily chose to highlight flown compact line arrays, others took a different approach

More than 650 worship production personnel and live sound operators converged on the Dallas Convention Center in early November to check out the first-ever Live Sound Compact Systems Demo, held in conjunction with the WFX Conference & Expo.

The demo provided the first-ever opportunity to directly listen to, evaluate, and compare a dozen compact loudspeaker systems in a controlled listening environment.

In addition, all participating companies had representatives on hand to provide technical details and pricing information.

The event was established in a full-size exhibit hall of the convention center, with a 150-foot by 150-foot demo space draped off to define the area and to help eliminate reflections. Carpet that covered the entire demo area furthered this goal.

While the 12 participating manufacturers primarily chose to highlight flown compact line arrays, others took a different approach. NEXO demonstrated ground-located GEO S12 loudspeakers and companion RS218 subwoofer, while Renkus-Heinz showed the recently introduced IC2 digitally steered system.

Danley Sound Labs flew the SH-96 Synergy Horn, which alternated with a brand-new new prototype on the floor - the compact SH-80 single-12 coaxial system expected to be available soon.

The Format
Live Sound/ProSoundWeb Senior Contributing Editor Craig Leerman served as the emcee for each one-hour full demo session, where all systems were played in a round-robin, random format, supplied with identical audio tracks.

Listeners were able to move from system to system, evaluating what they were hearing and also observing each system’s scale, components and other important details. Each participating company also provided 15-minute exclusive demo sessions, further showcasing their technologies.

The audio tracks, which were selected via voting by all participants, were provided to each system via a digital signal chain. A PC loaded with Wave files of all tracks fed a Focusrite Scarlett 8i6, an 8-input/6-output audio interface, with the SPDIF signal going to a Yamaha LS-9 digital console, which was also used as the system’s master word clock.

From there, signal went out via an Audinate Dante network on Cat-5e to Link DGLink stage boxes, and from there, each system received an analog feed. This system backbone was capably managed by veteran sound professional Tim Weaver of Waco, TX, who served as A2 for the event.

Following the initial system setup and optimization process prior to the demo, Leerman verified that system each was no louder than 102 dB (pink noise C-weighted), measured at a distance of 100 feet. Those levels were then locked into place to insure uniformity.

Leerman also mastered all of the program audio tracks ahead of the event to insure they were at the exact same level.

Valuable Opportunity
“The opportunity to hear a dozen loudspeaker systems playing the same tracks side by side, with a serious attempt at keeping the levels respectable, provided valuable information for us as well as prospective customers,” states Mike Hedden of Danley Sound Labs. “We’re looking forward to participating again next year.”

Mexico-based Sensey Electronics even utilized the demo as the initial launch pad of its products into the U.S. market. “The demo is something our industry needs to do more, and proved a great opportunity to introduce ourselves and our products to a new market,” notes Jerry Colmenaro of Sensey Electionics. “This was a fantastic event.”

Prior to each demo session, all attendees were supplied with a booklet containing the key specifications of each system, including pricing information. And, following each session, prize drawings were held, with lucky attendees walking away with audio products from Audio-Technica, Shure, ADK, Telefunken, Link, and Church Audio/Video.

“It was nice to be able to go to a major trade show and get our products heard, and a great way for the attendees to listen to products from various manufacturers with the same music tracks,” says Hugh Sarvis of WorxAudio Technologies. “We would love to participate again.”

Plans are already underway for another compact system demo – with even more features for attendees – to be held in conjunction with next year’s WFX Expo in Atlanta, September 19-21. More on that as soon as details are available.

WFX Dallas 2011 also proved a rousing success, with team members from over 1,000 churches in attendance. More than 50 conference sessions were presented across six tracks (Audio/Video/Lighting, Design, Facilities Management, Leadership & Management, Social Media & Communication, IFRAA), along with a sold-out lineup of more than 50 Hands-On Training sessions, including all-new advanced-level courses.

Demo Participants:
QSC Audio
Renkus-Heinz
NEXO/Yamaha Commercial
L-Acoustics
WorxAudio Technologies
Outline
Danley Sound Labs
Alcons Audio
dB Technologies
ISP Technologies
Bose Professional
Sensey Electronics

 

 

 

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Wednesday, December 07, 2011

Studio Six Announces New Transfer Function Module For AudioTools Suite Of Applications

Includes a magnitude plot, phase plot, and coherence curve, as well as an accurate impulse response–based delay finder

Studio Six Digital has announced the upcoming release of a new Transfer Function Module within the AudioTools suite of applications for the iOS platform.

The Transfer Function Module includes a magnitude plot, phase plot, and coherence curve, and will also include an accurate impulse response–based delay finder, capable of resolving delays down to a single sample.

It does not, however, include delay tracking, multi-time window based analysis, or any Smaart v.7 code or routines supplied by Rational Acoustics, nor will it communicate with Smaart v.7 in any way.

Notes Studio Six Digital president Andrew Smith, “Rational Acoustics and Studio Six Digital have been working together over the past several years to bring a subset of the Smaart 7 functionality to the iOS platform. However, as we got deeper into the process of porting the Smaart 7 Transfer Function to iOS, it was taking much longer than we had originally thought, and we realized we needed to come up with a different plan.”

“Bringing the subtleties and power of the Smaart 7 Transfer Function to the iOS platform is not trivial,” Smith continues, “so rather than wait to bring a Rational-branded multi-channel product out on iOS, we decided to release our own Transfer Function, which has been in development for more than a year. We are now ready to release a basic, but highly usable Transfer Function, although without the advanced algorithms present in Smaart 7.”

Studio Six Digital’s Transfer Function will require a 2-channel audio interface, such as iAudioInterface2. It will also work with the original iAudioInterface, or any other high-quality 2-channel iOS audio interface. It will not work with the built-in microphone or headset mic input.

For those seeking the ability to utilize Smaart v.7 in a multi-channel capacity on their iOS platform device, Studio Six Digital and Rational Acoustics are in development of a Smaart iRemote application which will allow iPhone/iPod/iPad users to remotely access Smaart measurement data from a configured measurement on any computer running Smaart v.7 on the same network.

Says Rational Acoustics CEO Jamie Anderson, “While not the complete stand-alone multi-channel “Smaart-on-an- iPad” solution that some of our users have been looking for, the iRemote will dramatically increase the functionality of Smaart for those who already own it. Instead of being tethered to a laptop setup at a single point in a venue, users can now roam freely while still having visibility and control of their measurement data.”

Both the Studio Six Digital Transfer Function and the Smaart iRemote are due to be released in mid-December and will be available on the App Store. The Transfer Function Module will retail for $79.99 and the Smaart iRemote for $99.99.

Studio Six Digital

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Tuesday, December 06, 2011

One Spec Fits All? Going Beyond The Surface To Further Understand Audio Products

An observation about human nature and how it applies to the audio business.

Some time ago I was on a trip to Japan, visiting some key customers, and I picked up on something that has been brewing in the back of my mind ever since.

Maybe the light bulb went on because I was in a different country with a different culture, but here was the thing…

I was asked several times by several people: “what’s the maximum number of channels your wireless system can operate simultaneously?”

That’s when the “something” in the back of my mind crystallized: we have a tendency to boil everything down a single specification.

The most readily obvious example that comes to mind is the number of diagonal inches for TV screens. Sure, the size is important, but I think it’s too easy to focus only on that and not pay attention to other things.

Perhaps a savy TV shopper asks about HD factors such as “is it true 1080P?” Again, it’s another spec that may be important, but not as important as other factors.

What about the quality of the picture? Unfortunately, picture quality, while there are many objective factors that can influence it, is largely a subjective thing.

And this is the crux of the issue, I think.

While it is easy to parrot a question like “how many inches?” it’s more difficult to comprehend the finer points. And when it comes to subjective evaluation, we can be wary of showing our ignorance in front of “experts”. (Well, some aren’t wary, but that’s another subject altogether.)

Loudspeakers
Since the debut of “modern” line array technology more than a decade ago, every loudspeaker manufacturer has been faced with the question - “Do you make a line array?” - even when in many cases, a line array is not the right solution for a particular application.

Now, line array has become a marketing buzzword in addition to a loudspeaker design approach, and the majority of manufacturers offer them.

Prior to this change, questions about loudspeakers boiled down to “What is the frequency response?” or “How many watts can it handle?” (I still cringe when I hear people confusing the amount of watts a loudspeaker can handle with how loud it will go.)

These glossy buzzwords also overlooked finer points like efficiency, dispersion, impedance curve, etc. in determining how the loudspeaker might actually perform.

Which brings me to…

Amplifiers
This one is obvious: “How many watts?” Never mind that actual power is largely a factor of the load conditions.

And what about the power supply? Remember those claims by an amplifier company popular in the 1980s? Somehow this manufacturer had found the “magic formula” for generating a (for the time) huge number of watts very efficiently.

But as it turns out, the difference was more about using switching supplies before most other manufacturers had gone there. And for good reason: switching supplies were not up to the enormous task of producing large amounts of clean current continuously in an amp that would actually sound good.

Times have changed. and now we can actually do that.

But the question “How much power?” misses out on many other important factors of amplifier performance such as damping factor, peak versus continuous power, whether or not the amp puts any noise on the power line, etc.

Further, there can be a trap in asking “How light is it?” as the only consideration, assuming that amplifiers all sound the same so the lightest, smallest one is best.

The bottom line is that both weight and output power are important but other issues must be given consideration.

Consoles
In the 1990s, a well-known maker of consoles and mixers led us to focus on channel and input counts as factors above all others. Simply, they introduced more and more mixers with more and more channels at very attractive pricepoints, but at the same time, these units lacked some important features that hobbled their ability to really get the job done.

Churches in particular bought into the mystique and ultimately paid the price. But they weren’t the only ones.

Many users asked the question about channels and compared the dollar cost and could not figure out why consoles that appeared to offer the same number of channels and seemed to have a lot of the same nifty features would cost twice as much or more.

Features like modularity, multiple sweepable EQ bands, bus amps and mix amps with enough headroom, etc. were not nearly as well understood and thus overlooked.

Wireless Microphones
Getting back to what started me on this subject – the question I probably hear more often than any other is “what kind of range do these things have?”

And my answer is always the same: “range of a wireless system is dependent on too many external factors to boil it down to a single number.” Things like interfering signals, antenna position, optimization of the antenna/cable system, the material(s) the walls are made of, line of sight, etc., all have a profound influence on the range of a wireless system.

The same system that might be fine over quarter mile out on the tundra in central Canada might only give you a block in downtown Manhattan.

The Marketing Game
Marketers are tasked with trying to get the message about their products out to their potential customers. Because they know that many customers aren’t as savvy as they could be, some marketers are able to use the single specification angle to influence buying decisions.

I don’t fault companies for making consoles that people want to buy, nor for adding line arrays to their product ranges, nor for stating that a wireless system has “range up to 1,000 feet” in their marketing literature.

But the problem is that in many cases, additional specifications and descriptions that might help people make more educated buying decisions are obfuscated or deliberately left out.

I’ve always felt that if you can’t find the specification, it was purposely left out so that people won’t see it and thus won’t avoid that particular product when they might otherwise. And in my view, this is wrong.

That said, however, we should do a better job of learning about the fundamental principles behind the products we specify, choose and use, in order to foster better results.

In the meantime, I’m selling an amplifier that will do 5,000 watts per channel for $500. The only problem is that the power rating above is at 0.5 ohms and it blows a fuse if the load is any lower than 4 ohms… Any takers?

Karl Winkler is Director of Business Development for Lectrosonics and has worked in professional audio for more than 15 years.

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iSEMcon Introduces New EMX-7150 Measurement Microphone

Includes both free field and derived diffuse field calibration data at no additional cost (calibration chart and ASCII-Data on CD)

iSEMcon has introduced the EMX-7150, a 1/4-inch measurement microphone that has a 10Hz to 20 kHz, IEC61672 class frequency response, designed for room and recording studio analysis, as well as indoor and open-air sound reinforcement measurements.

The EMX-7150 is an omnidirectional type microphone and the first to include the free field and derived diffuse field calibration data at no additional cost (calibration chart and ASCII-Data on CD).

It can be powered with phantom power and handles sound pressure levels up to 145 dB (SPL).

The robust stainless steel body construction uses a watertight state of the art, Neutrik XLR connector and comes complete with a small windshield and spring loaded holding clamp.

Also available is an optional super guard windshield protecting the microphone port from dirt and spraying water as well as trickle water from the back.

iSEMcon

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Friday, December 02, 2011

Listen Technologies Introduces New ADA App For iPhone And iPad

Application to take out the guess work for ADA requirements

Listen Technologies has announced a simple-to-use mobile program for the Americans with Disabilities Act (ADA) Assistive Listening Standards.

The ADA Compliance Assistive Listening Calculator is a complete tool-set for understanding and calculating facility requirements to meet the 2010 ADA Standards For Accessible Design.

Complete with an easy-to-use calculator, e-mail capability, helpful ADA information links, product assistance and quote request button. This application will take out the guess work for ADA requirements.

The mobile app is based on Table 219.3 Receivers for Assistive Listening Systems from Section: 706 Assistive Listening Systems of the Department of Justice Title III of the ADA. The table outlines the minimum number of receivers/assistive listening devices (ALDs) required based on the capacity seating of assembly areas; and the minimum number of ALDs that are required to be hearing aid compatible.

The ADA Compliance Assistive Listening Calculator App is available as a free app here.

Listen Technologies

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Thursday, December 01, 2011

Audix Introduces TM1 Plus Measurement Microphone With Combination Kit

With the TM1 Plus, customers are able to accumulate multiple microphones

Audix has introduced the TM1 Plus, a combination kit that includes the TM1 measurement microphone, threaded acoustic windscreen, shock mount clip, 1/2-inch calibrator adapter and microphone calibration data on CD. 

The data files are a numeric representation of the TM1 frequency response. Together with the provided sensitivity of the microphone, they can be used with a variety of popular software measurement systems to correct the response curve of the microphone. 

“The TM1 has earned very high acceptance among sound engineers and sound designers who rely on measurement systems to properly tune a PA system for live sound venues, theatres, houses of worship, and a wide variety of installations,” says Audix VP of sales Cliff Castle. “Measurement software packages have become very robust and are now able to take simultaneous measurements with multiple microphones, allowing sound designers a very accurate picture of how the PA system is reacting throughout a room in real time. 

“With the TM1 Plus, customers are able to accumulate multiple microphones, essentially creating matched sets once the data files are input into the software measurement programs.” 

The TM1 is a 6mm pre-polarized condenser microphone that is designed, assembled and tested by Audix in Wilsonville, OR. It has a precision-machined four-stage brass body and capsule housing, low noise SMT circuit, nickel plate finish,  field replaceable parts, Switchraft XLR and shock absorbent O-rings. 

All accessory items for the TM1 are included in the TM1 Plus kit. U.S. MSRP is $450. 

Audix

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