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Wednesday, May 02, 2012
When Eyes And Ears Don’t Agree: Perspective On The Role Of Measurement
Sound system design and implementation is first an engineering practice
In my previous article (here), I addressed some of the caveats of measuring sound fields in enclosed spaces.
The conclusion was that the eyes and ears do not always agree when it comes to sound quality.
If making acoustic measurements is so difficult, why bother? Why not tune the system based solely on listening? Because there are several very good reasons for including our eyes in the system tuning process!
We humans base our concept of reality upon the evidence presented by the five senses to the brain. Our concept of “blue sky” or “green grass” is the result of the tuning of our visual system and the programming of our brains. If our visual system were tuned to sense a different part of the electromagnetic spectrum, then our perceptions about reality would change.
Our sense of sight is programmed with base-line reference values from the day we’re born. We know what the sky should look like, and we know what the grass should look like. As far as we know, “green” and “blue” mean the same thing to everyone.
This is why we can walk into an electronics retailer, gaze at a wall of television sets, and pick out the one that has the “best” picture. The best one is the one that conforms the closest to our base-line programming of what certain colors should look like.
NO “ABSOLUTES”
Our hearing system has no such set of “absolute” references. Most of the sound events that we hear each day are man-made. Depending upon your walk in life, the sounds that you hear most often are probably different than the sounds that I experience. Only those who have undergone special sensory training have a concept of what the “correct” sound of a violin, piano or any musical instrument is - and even that is controversial among the “experts.”
The realm of sound is a subjective world, one that is devoid of absolute standards that form benchmarks for what we perceive. I may feel that a Gibson guitar sounds the “best,” and you may feel that a Fender sounds better.
There is no absolute benchmark to justify either position, so it simply ends up being a matter of opinion.
Loudspeakers are much the same way. I may feel that the “Blowsound 2000” sounds much better than the “StackNBlast Z-12,” and you may have the opposite opinion. Without a standard there is no way to justify either position, so we just have to agree to disagree.
Measurement provides the standard by which we can justify an opinion of what sounds “best.” Because loudspeakers ideally are reproducers of sound (as opposed to producers), and since we can measure what goes in (electrical energy) and what comes out (acoustical energy), we have a benchmark by which to judge the accuracy of the loudspeaker.
If a loudspeaker “sounds good” when playing back a Rob Zombie CD, yet its measured magnitude response looks like the Swiss Alps, then we can brand it as “inaccurate” with good authority. Accuracy is not necessarily a prerequisite for (subjectively) “good” sound.
The human auditory system is the most powerful analyzer that we have at our disposal. It is a two-channel, frequency-dependent, localizing data collection system with on-board algorithms that give it subjective perception. Even the most powerful analyzers can only emulate a few of these characteristics.
TWO SHORTCOMINGS
The importance of the listening process in tuning sound systems should not be underestimated. In spite of its strengths, our hearing system has two serious shortcomings with regard to adjusting the sound system’s response: it is not calibrated, and it is not consistent.
The lack of calibration means that we cannot listen to a sound and state with accuracy how loud it is. At best, we can offer a subjective impression of “pretty loud”, “ear-splitting”, etc.
Even a highly trained listener has difficulty identifying an absolute sound level to within 3 dB of its true level, which is a power ratio error of 2-to-1. So, if you guessed that the SPL were 87 dBA, and you were off by 3 dB, you missed by a 2-to-1 power ratio - a pretty large error! Absolute level measurements are trivial for analyzers - most of which can measure sound levels to within a fraction of a decibel.
The response of the human auditory system is not consistent. It changes with exposure, which means that after you listen to loud sounds for a while, the characteristics of your hearing system change.
The stapedius tendon (try saying that three times fast!) attaches to the bones of the middle ear and dampens their motion when your ears are exposed to loud sounds over a span of time. So, what sounds fine early in the show is not likely to sound the same by the end. This “threshold shift” is nature’s way of protecting us, but as with all protection mechanisms, it can be circumvented.
We’ve all experienced performances that get louder as the evening wears on. This is caused by the sound operator tracking the threshold shift with the main fader of the system. Visual feedback (the meter’s mixer) can prevent this from happening. This is, of course, a measurement. Humans are subject to listener fatigue - a condition that makes it very difficult to listen objectively after prolonged exposure to sound at any level.
A good night sleep will “reset” our hearing system and render us capable of critical listening. Analyzers have no such malady. They don’t get tired as the day wears on. So when the analyzer and your brain start to disagree, it may be time for a rest.
REALITY DISTORTION?
We work in an industry where a lot of money is made by distorting reality. You can drop a thousand dollars on a processor that essentially increases the harmonic distortion of your system or fills its response with deep notches, and feel like you have improved its sound quality.
Yet no one buys a processor for a television that makes the grass look blue or the sky look green. When the goal of a system is accurate sound reproduction (this isn’t always the goal) then accurate has no meaning unless a benchmark exists. This is where measurement comes in.
Consider the following scenarios:
1) You’re at an airport and you hear a perfectly awful announ-cement come over the sound system. Why did it sound bad? The initial reaction is usually to blame the loudspeaker, since it is where the bad sound came from.
But a good loudspeaker that is fed bad program material will still sound bad. Perhaps the problem is an overdriven amplifier, or poor micing technique on the part of the talker. How would one find out?
Once again, we return to measurement. If I feed the loudspeaker a known stimulus and it can reproduce it with good fidelity, then the problem lies elsewhere in the system. The process is repeated until the offending component is found, which could ultimately be a gate agent with a bad head cold. A sound system is only as good as its weakest link, and measurement is necessary to test the links.
2) A manufacturer may complete a run of loudspeakers, and find that two picked at random for a listening test sound dramatically different. Which one is the most “correct?” A measurement can provide the answer. Loudspeaker manufacturers measure each loudspeaker that comes off the line to assure that it falls within a set of tolerance values that were established during the model’s design.
Only measurements can verify that the replicas are identical to the original. No one wants to buy a loudspeaker whose only validation of performance was “Bill’s ears in Quality Control”. Bill may have been tired when my loudspeaker rolled off the line!
3) You’ve been called to tune a sound system which the client complains lacks “presence”. Most people would start boosting the high-frequency tone controls or the house equalizer to “restore” it.
But what if the system simply lacks the bandwidth to reproduce full-range music? A mixer with all of the high-frequency tone controls fully clockwise and a “smiley face” on the house EQ is probably deficient in bandwidth. Either that, or the sound guy just retired from 20 years on the road as “monitor engineer” for a heavy metal band. Some simple measurements can reveal whether the system is capable of what you are asking it to do.
4) The congregation at a local house of worship complains about poor intelligibility from the house system. Three different people have been consulted about the problem, and each of them suspects a different cause. Now how do we really get to the bottom of this?
The answer, of course, is measurement. It’s possible to spend a vast amount of time and money “fixing” the wrong problem. None of us would submit to surgery because our doctor suspects that we need it. We rely on X-Rays and CAT scans to reduce the risk of an incorrect diagnosis.
ORDER THE COMBO
Most sound system chores require a combination of listening and measurement. One without the other can yield completely unsatisfactory results. The two used together can quickly bring a system to its fullest potential, and also reveal the shortcomings of a sound system that might be addressed by equipment upgrades, changes to room acoustics, etc.
Those that don’t measure are operating in a world without references where “anything goes.” This approach is fine for purely artistic endeavors, but sound system design and implementation is first an engineering practice. Loudspeaker selection and placement is mostly science, while selecting what color to paint them is mostly art.
“Mostly” leaves room for the other, but points to the dominant process. Measurements assure that the science has been satisfied, paving the way for the artistic use of the sound system.
Pat and Brenda Brown own and operate SynAudCon, the leading independent professional audio education source, with training sessions held around the world and online. For more info go to www.synaudcon.com.
This identity depends on the physical characteristics of the sound’s medium (the matter or substance that supports the sound). Let’s take an A at 440 Hertz produced at 60 decibels: we can immediately tell if the sound was emitted from a violin, saxophone, or piano.
Yet, even though the instrument is different, it’s the same note and the same amplitude. The difference is in the sound production: string, air column, etc..
Plus, the sound isn’t generated by the same “tool”: a bow for violin strings, a reed and an air column for the sax, and felt covered hammers that strike the piano strings. It’s the different physical characteristics of the medium and the « tool » that determine the characteristic sound waves in each case.
Later we will also see how a sound chamber adds another dimension to this definition.
Waveform
The most basic waveform is a sine wave (sinusoid) (Figure 1). It could be considered the atom of sound.
Figure 1: A sine wave (or sinusoid). (click to enlarge)
Pure sinusoidal sounds are rare (i.e., tuning forks, drinking glasses being rubbed) and were considered to have strange powers over human behavior at one time. Most sounds that surround us are of a more complex nature.
This means that inside a sound, that we perceive as being unique, there is a superposition of many sine waves that have, in a way, fused together to become one sound.
It’s the nature of this superposition itself that determines the resulting waveform (Figure 2) and that is responsible for its timbre. This is called a spectrum.
Figure 2: Above, a square wave; below, a sawtooth (or saw) wave. (click to enlarge)
Spectral Representation
There are many ways of graphically representing sound. For instructional purposes we have chosen to use a spectrogram for its clarity and simplicity.
Horizontally: time in seconds. vertically: frequency in Hertz. A sine wave (sinusoid) at 100 Hertz is represented by a horizontal line at a height corresponding to 100. (Figure 3)
A harmonic sound at 100 Hertz is represented by superimposed lines corresponding to sine waves of 100, 200, 300: n x 100 Hertz. The length of the lines represent the length of the sound.
Figure 3: Spectral representation. (click to enlarge)
Noise
Let’s imagine a case where all sine wave frequencies that are perceptible to the human ear (from 20 Hz to 20 kHz) and having the same amplitude, are “mixed” into one sound signal.
We get what is called “white noise,” or in other words, “hiss.” If the white noise is very short we would perceive it to be a kind of short percussive sound. Consonants belong to this category, in the same way that a sound medium that receives the attack of the “tool” which “kick-starts” it, produces as noise. This noise corresponds to the time it takes for the sound wave to stabilize and take its final form.
The “rubbing” of a bow on a string is similar to a hissing sound, while a hammer hitting a piano string is similar to a percussive sound. These notions will be dealt with in greater depth when we get to envelopes and transients. In the case where a series of noise frequencies is contained between certain limits we will refer to them as noise bands.
If a zone is particularly swollen in energy, then we can speak about colored noise around that zone. Pink noise is white noise with a power density that decreases by 3 dB per octave.
Figure 4: Different types of noise. (click to enlarge)
Harmonic Sounds
Having already highlighted the superimposed or complex aspect of sound, we are now going to focus on a specific category of frequencies in a sound spectrum: harmonics.
A harmonic sound is a sound which contains sine waves that obey the mathematical law called the Fourier series. This law translates as follows: A complex periodic signal is made up of a certain number of component frequencies that are integers of the fundamental frequency.
An example of a harmonic sound: a sound at 100 Hz in which the component waves are 100, 200, 300, 400, 500 and 600 Hz. The perceived pitch is the lowest frequency: 100 Hz.
The following component waves (2 x 100, 3 x 100, 4 x 100, etc.) are calculated on integers and are called harmonics. The lowest frequency, on which they are based, is called the fundamental.
The number, or “rank,” of a harmonic is the integer by which the fundamental is multiplied. For example, the 3rd harmonic would be the one at 300 Hz. (Figure 5 a-c)
Figure 5a: Spectral representation of a harmonic signal with 4 harmonics.
Figure 5b: The sine wave components of the signal, top to bottom.
Figure 5c: The combination of the 4 sine waves.
The pitch of a harmonic sound is easily perceptible to the ear, and these sounds usually have an “in tune” quality about them. That’s why melodic musical instruments are designed with the goal of producing harmonic spectrums.
Noises, like those we referred to earlier, are aperiodic signals. They are characteristic of percussion instruments, for example.
Distribution Of Energy In The Spectrum
Regions of relatively great intensity in a sound spectrum are called formants. In the case of a band of consecutive frequencies it is referred to as a formant zone between x and y Hertz (Figure 6).
Figure 6: A harmonic sound at 100 Hz with a formant at 300 Hz.
This distribution of energy plays an important role in the perception of timbre, as do the number of components in the spectrum, their distribution, and its regularity or non regularity. Figure 7 provides some graphic examples.
Figure 7
a - violin: a hiss noise at the attack, harmonic spectrum
b - flute: harmonic spectrum
c - piano: noise of the hammer attack, percussive sound and spectrum not quite regular in its harmonics
d- warm sound: few harmonics but regular distribution of the energy from low to high
e - piercing sound: harmonic sound with a lot of intensity in the highs
f - Hollow sound: few harmonics in the mids
g - nasal sound: weak lows, intense mids, weak highs
h - non harmonic sound: like a non-tuned bell
I - square signal, odd harmonics: like a clarinet sound
Equalizing On A Console
It’s the EQ section of a console that will allow us to tweak or correct timbre. Depending on the model, the EQ section is more or less sophisticated and offers different possibilities of adjustment.
We won’t be dealing with simple high/low EQ knobs or switches that you can find on hi-fi amplifiers or entry level mixers which are only meant to adapt a sound to a specific listening area.
We’re more concerned with the EQ controls that are found on small modern digital models or part of most major recording software. Keep in mind that EQ is mainly used for one reason—to correct, and not in the hope of improving the recorded signal: you can never turn a mediocre recorded sound (due to bad placement of the mic or even the quality of the mic itself) into a great sound by just using EQ.
Equalizers split the audible frequency range (20 Hz to 20 kHz…) into many sub-ranges. Thus one generally talks about highs, medium highs, low mids, and lows.
The first thing to do, then, before tweaking any knobs, is to determine in which frequency range the problem lies, then after that, the nature of the problem. Is it due to too much coloring that wasn’t detected during the recording process, a parasite due to the environment, or a masked effect due to the presence of other instruments?
What Does It Look Like?
Equalizers are harmonic and partial filters. Their specificity lies in the fact that they not only can get rid of component frequencies, but that they can also amplify chosen frequency zones. Of course, if there isn’t anything in the signal in that range, only hiss will be added!
Good EQ sections generally have 4 bands, and each offers at least 2 controls: frequency adjustment and gain. These are called semi-parametric. There’s often a third setting called the bandwidth or “Q” which has the purpose of enlarging or tightening the frequency range (bandwidth) of the filter.
When this third control is present, the EQ is then called a parametric equalizer. Frequency adjustment will be tweakable between the upper and lower limits of the sub-range of the filter (with software these limits no longer exist!)
The gain knob defines, in dB, how much the filter will effect the chosen frequency. As we can see here in Figure 8, borrowed from Cubase, this gain can be positive or negative. We can also see that the curve of the bandwidth can be wider (hump shape) or narrower (peak shape). This shape corresponds to the bandwidth which is adjusted by the Q setting.
Figure 8: Two EQs in Cubase. Note the cut frequency represented by the point in the curve, and the width of the curve, which corresponds to the Q of the filter.
How To Modify Timbre
Keep in mind that all EQing on an instrument will be destructive with respect to the recorded sound, just as the latter is also, in many cases, an imperfect copy, of the original. So one must be careful!
Before touching anything, think about what you want to accomplish with EQing, i.e., “I want a ‘warmer’ sound, I want to cut the bass, I want my instrument to stand out in the mix, I want to get rid of that annoying resonance that came from the studio…”
The spreadsheet below is offered as a sort of “quick guide” chart. Don’t forget, however, to listen: your ears are the ultimate judge.
Los Teignos is a “rêveur à temps partiel” and contributor to Audiofanzine.
For more audio/sound related content and resources, go to Audiofanzine.
Power Lines: Determining When “Isolated Grounding” Is Needed
Options and errors in AC wiring
If electrical wiring, from main breaker panel to outlet, consists of Romex and plastic J-boxes, an “isolated” or “technical” ground system is already in place. This is the case In most, but not all, residential
wiring.
However, when wiring consists of metallic conduit and J-boxes, as in most commercial buildings, an isolated safety-grounding scheme can sometimes reduce audio system noise. It is most applicable in situations where conduit may come in contact with building steel, water pipes, gas pipes, or other structures because they may be grounded at distant locations (perhaps even the building next door or across the street) and will inject noise current into the safety ground system.
Special insulated ground or “IG” outlets (distinguished by a triangle marking on their face and, most popularly, orange in color) are used, which intentionally insulate the green safety ground terminal from the outlet mounting yokes or saddles.
Therefore, safety grounding is not provided by the J-box and conduit, but by a separate insulated green wire which is routed back to the main electrical panel.
Of course, the J-box and conduit must also be properly grounded by other, usually existing, means. Code requires that this or any safety ground conductor be routed in the same conduit or cable as the line and neutral circuit conductors. Although not explicitly stated in Code, this practice prevents loop inductance from limiting fault current, assuring fast breaker response should a fault occur.
Most often, wiring is not “daisy-chained” to outlets on the same branch circuit, so noisy leakage current from one device couples less to others on the same branch circuit. However, inductive coupling from phase conductors to the ground conductor (the major source of ground voltage differences between outlets) is not reduced.
Technical grounding practices are covered by NEC Article 250-74 and its exceptions. An excellent reference for system grounding, with emphasis on equipment racks, is a white paper by Middle Atlantic Products (to which I was a contributor) available at www.middle atlantic.com/pdflPowerWhitePaper4_07.pdf.
A potential problem with isolated grounding is that unaware users may connect a signal cable between a piece of equipment powered by an isolated ground outlet to another piece of equipment powered by a non-isolated ground outlet. Noise now enters the isolated ground system via the signal interconnect.
This problem is very common in large computer networks and, like most computer noise problems, will likely be blamed on something else.
The resultant ground loop and circulating noise currents defeat the purpose of the isolated ground system.
THE NEUTRAL-GROUND SWAP
National Electrical Code recommends that premises wiring be sized such that regulation of the most distant outlet on a branch circuit is 5 percent (6 volts of drop for 120-volt circuits) or better under full load. This means that 3 volts are dropped across both line and neutral conductors.
On the other hand, safety ground wiring normally carries well under 100 rnA of cumulative equipment leakage currents. This could occur for a worst-case scenario of 20 pieces of equipment, each having a three-prong AC plug and 5 rnA leakage current (maximum allowed by UL). In this scenario, total voltage drop over the length of the safety ground conductor would be only about 20 mv.
But, as shown in the schematic directly below, a simple outlet wiring error that swaps the neutral and safety ground conductors allows load current to flow in the safety ground wiring.
The middle outlet has N-G swap wiring error. (click to enlarge)
When equipment load currents of 15 or 20 amps flow in the safety ground wiring, voltage drops as high as 3 volts can occur over its length (assuming the safety ground wiring is the same gauge as line and neutral).
Although the outlet is still functional and safe, this error can cause system ground noise to increase by a factor of 100 or about 40 dB!
Simple outlet testers cannot find the problem because they test only for voltages at the outlet.
Clamp-on AC ammeter. (click to enlarge)
Since neutral and safety ground are bonded together at the main breaker panel, they are indistinguishable to these testers. Even more sophisticated testers cannot reveal the error.
However, if nominal loads (say 100-watt light bulbs) are plugged into each of the outlets on the branch circuit, a clamp-on ammeter like the one shown here at left can quickly reveal and locate the mis-wiring by measuring current at points A, B, and C.
Referring to the schematic, abnormally high currents would be measured at locations A and B but not C.
Bill Whitlock has served as president and chief engineer of Jensen Transformers for 20 years. Read more articles by Bill about best AC and electrical practices here.
FiberPlex Technologies Introduces WDM16 Active Wave Division Multiplexer
FiberPlex Technologies has released the WDM16, a 16-channel active wave division multiplexer that does not require optical wavelength matching, which allows users to use standard off the shelf equipment without worrying about having to have equipment operating at specific wavelengths.
The WDM16 provides (16) Small Form Factor Pluggable (SFP) ports that can be loaded with any sort of optical laser of any speed or mode.
The WDM16 allocates the individual wavelengths for the CWDM by taking each input in (from 155MB to 2.5GB), regardless of wavelength, internally translating that wavelength to the assigned wavelength internally in the WDM16 for that channel, and multiplexing up to (16) of these channels for a maximum of 40GB on a single fiber.
The WDM16 is designed for use with the FiberPlex FOM series modules, FiberPlex Shadow enabled devices, or virtually any third party fiber optic equipment.
Is is equally at home as a stand-alone system allowing the user a quick, easy and economical way to increase the bandwidth of existing fiber optic infrastructure, or to limit the amount of fiber infrastructure necessary in any new construction where adding more fiber is cost prohibitive.
Available fully redundant, hot swappable power supplies insure reliability. Redundancy is carried even further to the channel level, where any single channel can fail without taking down the whole unit.
Additionally, the default failure mode of the communications microprocessor is to have all channels operate and fans run full. The WDM16 is ready for any mission critical application.
Wave division multiplexing can free up coveted fiber pairs, when dark fiber is scarce, and in new or retrofit installations it can provide considerable savings compared to the labor costs to pull new or extra fiber runs. The WDM16 is the ultimate solution for expansion of your fiber infrastructure.
Acoustically Incompetent: The Need For Architects To Learn To Listen…
The issue is well understood, and the knowledge to solve the challenge already exists
It’s been 112 years now, and you’d think it’s been long enough. Yet some of the brightest guys in America keep making the same dumb mistakes over and over again.
And ignoring the issue hasn’t made it go away either - it just keeps popping up like Baby Boomers and their anticipated Social Security payments…
Still, you’d think someone given the responsibility of designing our great facilities would want people to be able to converse and enjoy listening to music in them. Sadly, that is far less often the case than necessary.
At the most basic level, sound bounces around unless it’s absorbed or diffused. Too many bounces and our brains get confused and we can’t enjoy the space. Too much intrusive noise and we get confused too, and the issue only gets worse as we age.
The cure is simple and well known. Go buy absorption and diffusion and sprinkle it liberally around a room, starting with the ceiling, floors, and walls. Absorption is cheap; diffusion more expensive. Yet neither is a rare or exotic item; they are both widely available and allow both performers and listeners to enjoy the space. You know, carpet works well if deployed properly.
We should all agree that a good sound system cannot fix a bad acoustical space. Neither can a great one. No amount of amplifiers and loudspeakers can “fix” a large room with insufficient acoustical absorption, no matter how loud it plays or how well its pattern is controlled. Even with the most exotic line arrays, the room will sound far better if properly treated to optimize the reverberation time relative
to performance expectations.
Yet for years architects in the U.S. have wrongly believed that noise and reverberation problems can he cured with exotic sound reproduction systems. They can’t. There is no $300,000 sound system that sounds good in a tiled restroom. Nor is there a three dollar sound system that can. Isn’t it. funny how modern restaurants using the exact same materials as our restrooms get the same “aural flush” result?
Did you know any acoustician can calculate and predict the results accurately long before the building is built?
One needn’t look very far to understand why it’s difficult to communicate in most modern buildings in the U.S. - it’s the fault of our architects. Their training is lousy.
How lousy? Apparently architects are no longer required to take Latin. Had they done so, they would realize that the root word in “auditorium” is not seismic retrofit; nor design/build; nor cost/plus; nor value engineering, nor even LEED.
Here’s a hint:
auditorium
1727, from L. auditorium “lecture room,” lit. “place where something is heard,” neuter of auditorius (adj.) “of or for hearing,” from auditor “a listener,” from audire “to hear” (see audience).
One might assume that a space dedicated to the idea that something should be heard would have a primary emphasis on noise reduction, reverberation control, and maximizing
speech intelligibility. Not so in American architecture. Even with seats costing $200 per evening for prime events now, our architects continue to treat acoustics as an inconvenient afterthought.
Why so? I’ve concluded there are a number of answers behind this debacle.
Many architects in the U.S. live exclusively in a visual world. It’s often all about the pretty picture in a magazine and on the web. Many European architects live in a visual and aural world and realize that the design of a facility affects the quality of sound reproduction.
Our architectural schools do not teach the subject properly, if at all. One of our more prestigious architectural schools’ offers a total of 123 total classes in its curriculum. Only one of them, “Design for the Luminous and Sonic Environment” appears to have an emphasis on the aural environment. Even in that one we take a back seat to lighting. Typical.
Ever looked an architect straight in the eye and asked them what they budgeted for interior acoustical treatments up front? Nine times out of ten the answer is nothing.
Architects routinely ignore their acoustical consultants input, and put in them in the unenviable position of having to justify their recommendations ad nauseum. Ever see a lighting designer having to justify their lamp selections in a similar manner? Nor have I.
Our architects need to better understand which materials have the best acoustical absorption. Wood is good, but not great for absorption. Fiberglass is two to three times better. There is no building code compliance enforcement for intelligible speech, thus it is not a priority for many architects. There is for fire sprinklers.
If the sprinkler system doesn’t work, the building doesn’t get a Certificate of Occupancy. True, there are some emergency evacuation standards that are just beginning to address the issue, but the lack of an acoustic code means a lack of enforcement. We regulate everything from tire tread wear to pajama flammability, but not basic audio quality in our society.
More expensive project labor means less expensive materials are used. Seen much granite used in buildings recently? Less expensive materials imply lower weight materials, resulting in less capability to attenuate sound transmission between rooms.
Perhaps some of that is our fault as sound system suppliers. My suspicion is that few in our profession are aware of how to calculate speech intelligibility. We have no control on the amount of fiberglass or diffusion installed in a building. Many have never bought any absorption or diffusion in their entire career.’
Providing a quality aural experience requires a quality acoustical space first and then a quality sound system to perform well. Go straight to “Audio Jail,” do not pass “Go,” and do not collect $200 if you think you can get away with an all-hard-surfaced interior, no matter how tightly you control your loudspeaker directivity.
In the meantime, architects in the U.S. need to step up to the plate. The issue is well understood, and the knowledge to solve the challenge already exists. No more research needs to be done. Get yourselves properly trained.
The first quantitative acoustically engineered building opened to the public in 1900! Boston Symphony Hall has been making money for a century now, and it’s well past time our architects use technology properly to improve acoustical performance throughout North America in every single building.
Remember Zappa’s Law: There are two things that are universal: Hydrogen and Stupidity. The inability to communicate successfully in our facilities falls in the latter.
John Mayberry has workedin and around the A/V business for more than a quarter of a century. Find him online at here, and check out his blog here.
Clear-Com Launches HelixNet Networked Partyline Intercom System
Clear-Com has announced the worldwide release of the HelixNet Partyline, the industry’s first networked partyline intercom system with a set of unique capabilities for achieving greater efficiency, cost-savings and flexibility from set up to operation and maintenance.
HelixNet Partyline provides digitized Clear-Com sound and central administration of the entire system (firmware upgrades and maintenance) from the Main Station with a single cable and flexible cable options, with the ability to leverage an existing cable infrastructure.
The initial release of HelixNet Partyline consists of the HMS-4X Main Station, HBP-2X HelixNet beltpack, HLI-2W2 two-wire interface module and the HLI-4W2 four-wire interface module. The system begins shipping in June of 2012.
“For over 40 years, the Pro Audio community has been using the common, three-pin XLR microphone cable to carry audio for analog partyline systems,” says Chris Barry, product manager at Clear-Com. “In order to preserve our customers’ investments in intercom systems and cabling infrastructure, we had specifically designed HelixNet Partyline to transmit four channels of digital quality audio, plus program and power for beltpacks, over a single, shielded twisted-pair cable (ex. microphone cable, Cat-5 or Cat-6 cable). This capability alone is unprecedented in the history of intercom technology.”
The HelixNet Partyline system also offers many unique features to create a much higher audio quality, increase efficiency during the set up and maintenance processes and simplify operations.
HMS-4X HelixNet Main Station and Interface Module
• High channel density and high user capacity. The sleek 1RU HMS-4X HelixNet Main Station fits into any standard 19” rack and can provide power and four channels of audio to support up to 20 digital beltpacks.
• No hum. No buzz. Unlike standard analog systems, the all-digital HelixNet system is immune to electro-magnetic interference and ground loops.
• Highly flexible and offers intuitive user operations. System settings and menus are quickly accessible. Firmware maintenance and upgrades can be achieved easily via USB ports.
• Greater connectivity with existing analog intercom systems and audio devices. The expansion bay in the Main Station allows optional HLI-2W2 two-wire and HLI-4W2 four-wire interface modules to connect easily with existing analog intercom systems and audio devices, while preserving the high audio quality that is free of hum and noise.
HBP-2X HelixNet Beltpack
• High channel density and selectable channels to save resources. The rugged, ergonomically-designed HBP-2X HelixNet Beltpack is a two-channel beltpack that can access two of any four system channels and program audio over a single cable, along with individual level control. Networked audio is distributed over a single, shielded twisted pair, keeping the number of cables required, low.
• Easy to operate and read. Optimally positioned buttons and volume knobs are easy to locate, identify and control on the beltpack. Channel labels are simple to read on the high-contrast 10-character OLED displays. Beltpacks can be set up in daisy chain or star configurations with no need for active split boxes.
• Durable and flexible. HelixNet Beltpacks are highly durable, fabricated from lightweight cast aluminum, and come with a sturdy beltclip, rubber bumpons and an integrated strap guide to offer a variety of practical mounting options.
Differences, Cause & Effect And Consequences Of Polarity And Phase
The terms "polarity" and "phase" are often used as if they mean the same thing. They do not.
Polarity and Phase - these terms are often used as if they mean the same thing. They are not.
POLARITY: In electricity this is a simple reversal of the plus and minus voltage. It doesn’t matter whether it is DC or AC voltage. For DC, Turn a battery around in a flashlight and you have inverted or, more commonly stated, reversed the polarity of the voltage going to the light bulb. For AC, interchange the two wires at the input terminals of a loudspeaker and you have reversed the polarity of the signal coming from that loudspeaker.
PHASE: In electricity this refers only to AC signals and there MUST be two signals. The signals MUST be of the same frequency and phase refers to their relationship in time. If both signals arrive at the same point at the same time they are in phase. If they arrive at different times they are out of phase. The only question is how much are they out of phase, or stated another way, what is the phase shift between them?
The important point to note in these definitions is that you can reverse the polarity of one signal and you can measure this change. You need two signals to measure a phase shift.
For convenience, the word “speaker” will be used in place of the more correct term “loudspeaker” in the rest of this article.
A picture is worth 1,000 words… but a few words of explanation can help.
The following figures show the differences and some consequences of polarity and phase. Figures 1 through 12 show graphs of sine wave signals. Actually it is a sine wave from one signal source split two ways. Except for figure 1, one of the splits is “processed” by reversing its polarity and/or by delaying it (phase shifting it) as described. To put this in the real world, imagine two speaker systems side-by-side, each reproducing one of the signal splits. (More precisely, the graphs show what you would see on an oscilloscope looking at the output of a mixing console with each split going to a separate input after one of the splits has been “processed”.)
The vertical scale in the graphs is in arbitrary units of -2 to +2 with lines at each 0.5 interval. If you like, consider this as -2 to +2 volts. Because phase shifts are measured in degrees, the horizontal scale in the graphs is labeled in degrees with a vertical line at each 90-degree point. One full cycle or period of a sine wave is 360 degrees.
Assume that the signals shown are 1 kHz sine waves, in which case each vertical line represents 1/4 millisecond of time. Sound travels in air about 3.4 inches (85 mm) in 1/4 millisecond so each vertical line also represents this distance. Note that in the graphs the signals all start 1/4 millisecond or more from the left so you can clearly see when each signal starts. (The importance of this will be seen in figure 9.) There is no signal along the flat line from -90 to 0 degrees.
Signals In Polarity, In Phase
Figure 1: This shows 3 periods or 3 cycles of two simple sine waves. Both are +/-1 volt high at their peaks = total of 2 volts. One is shown in blue the other in red.
Figure 1: Sine Waves in Fig. 1 Added.
Figure 2: This is what happens when the two are combined (= added together). This is exactly what would happen on a line exactly between the two side-by-side speakers. The two signal beings being in phase and in polarity add up so the peaks are now at the +/- 2 volt lines = 4 volts or twice the original signals. Acoustically this is an increase of 6 dB = 20 x log(1+1).
Figure 2: Two Sine Waves - Same Polarity & Phase.
Signals Out of Polarity
Figure 3: This is like figure 1 but the second sine wave, shown in red, has been reversed in polarity. As you can see the + and - voltage points are exactly opposite from the first sine wave, shown in blue. This would be accomplished by reversing the +/- input connection on the speaker reproducing the red sine wave.
Figure 3: Two Sine Waves - Red = Polarity Reversed.
Figure 4: This is what happens when the two are combined. Each point of the two signals being in phase, but opposite polarity, adds up to zero. Acoustically this is an infinite decrease of output. Because you can’t take the log of 0 assume the difference is actually 0.0.01 volts (the dots = 58 more zeros). 20 x log of this number is -1200 dB. That should be pretty quiet. You can’t easily hear this with two speakers because of having two ears. But using a very carefully positioned microphone to measure this in a place with no sound reflections, you would find almost no signal.
Figure 4: Sine Waves in Fig. 3 Added.
Signals Ot of Phase
Figure 5: The second sine wave, shown in red, starts 1/4 millisecond later (90 degrees later) than the first one, shown in blue. Put another way, the second signal has been delayed by 1/4 millisecond.
Figure 5: Two Sine Waves - Red = Phase Shifted 90 Degrees.
Figure 6: This is what happens when the two are combined and it’s pretty interesting. First notice that the peaks are almost at the +/-1.5 volt lines. The value is actually +/-1.414 volts. This is a 3 dB increase. This would be like listening to two speakers but the one reproducing the red sine wave is 3.4 inches (85 mm) further away from you than the other. The first thing you hear is only from the speaker reproducing the blue sine wave. The black line starts when the sound from the second speaker is heard and this line is the combined signal of both speakers.
Figure 6: Sine Waves in Fig 5 Added
Suppose the speaker reproducing the red signal were only 2.25 inches (57 mm) further away. The signals would be shifted by only 60 degrees. The increase for the combined signal would be about 4.5 dB. So the amount of phase shift is important.
The second thing to notice is what happens at 1/4 millisecond or 90 degrees after the blue signal starts when the second signal “kicks” into the picture represented by the line turning black. There is a distinct change in the waveform.
The third thing to notice is that the entire waveform after the “glitch” is shifted in time compared to figure 7 about 45 degrees = average of 0 and 90 degrees.
Signals Out Of Phase And Polarity
Figure 7: The second sine wave, shown in red, is a combination of the sine wave in figures 3 and 5. The signal not only has its polarity reversed but it is shifted in phase by 90 degrees compared to the first signal, shown in blue. In this case the speaker reproducing the red sine wave has its +/- input connection reversed in polarity and is 3.4 inches (85 mm) further away from you than the one reproducing the blue sine wave.
Figure 7: Two Sine Waves - Red = Phase Shifted 90 Degrees & Polarity Reversed.
Figure 8: This is what happens when the two signals are combined. The picture is similar to figure 6 with two important differences. First the “glitch” at the point where the second signal starts is different. This is the point where the line turns black. Second is that the entire waveform is shifted by 45 degrees again but this time to the left of the original signal.
Figure 8: Sine Waves in Fig. 7 Added.
The “Glitches”
The glitches in figures 6 and 8 give an indication of what happens during the onset of a signal. While the so-called steady state portion of the combined signal (shown by the black portion of the lines) looks the same except for the amplitude change, these glitches will affect the transient attack of sounds. This is not to say that either will sound horrible, but a phase shift between otherwise identical replicas of a sound WILL make a difference in the sound of the initial transient attacks, depending on the frequency and amount of phase shift.
This is exactly the kind of phenomena that can occur in the crossover region of a speaker. This is because the distance from each driver to the listener is usually different and the crossover itself shifts the phase of the signal between the drivers. Speaker designers are often faced with a choice between something like what you see in figures 6 and 8. Neither is “correct” so a designer can only choose the one that “listens” better. Just looking at these two, I would bet the waveform in figure 8 might sound better and the choice would be to reverse the polarity of one of the drivers. These crossover “glitches” occur only over a small range of frequencies where both drivers reproduce the sound. It is well accepted by designers that this kind of “improvement” is sonically more significant than the fact that frequencies above and below the crossover point may be out of polarity.
Signal Phase Shifted 180 Degrees
This is where many get into trouble in thinking that phase and polarity are the same thing, meaning that it is often assumed that a 180 degree phase shift and reversing the polarity are the same.
Figure 9: In this figure each sine wave lasts for only 2-1/2 cycles. The second sine wave, shown in red, is shifted in phase 180 degrees from the first shown in blue. This is what would happen if the speaker reproducing the red sine wave were about 6.8 inches (170 mm) further away from you than the one reproducing the blue sine wave. You can see that between the 180 and 900 degrees the signals LOOK like they are simply out of polarity but they are NOT. It is VERY important to note that if you could not see the beginning or the end of these signals you could not tell whether they were out of polarity or 180 degrees out of phase. Too often this is what causes confusion between a polarity reverse and a 180 degree phase shift.
Figure 9: Two Sine Waves - Red = Phase Shifted 180 Degrees & Polarity Reversed.
Figure 10: This is the result of combing the two signals. Unlike figure 4 where the signals are simply out of polarity, and completely cancel, there are clearly two positive halves of a sine wave visible before and after the two signals cancel along the black line between 180 and 900 degrees. The first is from the blue sine wave in figure 9 that occurs before the start of the red sine wave. The second is from the red sine wave in figure 9 that continues after the blue sine wave stopped.
Figure 10: Sine Waves in Fig. 9 Added.
Signal Phase Shifted 180 Degrees And Reversed In Polarity
Figure 11: This is the same as figure 9 but the polarity of the red signal is reversed from figure 9.
Figure 11: Two Sine Waves - Red = Phase Shifted 180 Degrees & Polarity Reversed.
Figure 12: This is the two signals in figure 11 combined. Between the 180 and 900 degrees, the signals add much like in figure 2. However there are significant differences in the overall 90 to 1080 degree signal. The first 1/2 sine wave of this signal is only from the blue sine wave from figure 11. The last 1/2 sine wave is only from the red sine wave in figure 11. You can clearly see that both of these 1/2 sine waves are only 1 volt at the peaks. This is a clear difference from figure 2 where all the peaks reach 2 volts.
Figure 12: Sine Waves in Fig. 11 Added.
The reason is that the two signals in figure 11, even though identical, are offset by 180 degrees. They add together only between 180 and 900 degrees when both are being heard. More importantly, during this time period DIFFERENT parts of the same signal have added together. For example you can see that between 180 and 360 degrees it is the second 1/2 of the blue signal’s first complete sine wave that adds to the first 1/2 of the red signal’s first complete sine wave.
Real Audio Signals
Sine waves are easy to look at to dramatically show the difference between polarity and phase. Armed with this knowledge you can look at figures 13 through 18 that show something like a real audio signal where the effects of polarity and phase are more difficult to see.
The signal shown in these figures was a generated by a mathematical algorithm that produces something close to a pink noise signal. Pink noise contains all frequencies with an equal amount of energy in each octave band. Real audio signals don’t look much different than pink noise (but one would hope they sound better!). The scales on these graphs are arbitrary. You can look at the vertical scales as +/-3 volts if you like. However, because of the way the signal was generated, there was no way to define absolute time or degrees along the horizontal scales. Suffice it to say that the phase-shifted signal used in these figures was shifted by one data point out of the 240 data points that make up the signal lines.
There is one important thing to understand about phase shift. The amount of time one signal is delayed from another will have different effects at different frequencies. Assume there is a 1 millisecond time difference between two identical signals. At 500 Hz the result will be as shown in figure 10 because at 500 Hz the 1 millisecond time difference is a phase shift of 180 degrees. The signals are offset by 1/2 a cycle. At 1 kHz the signals will be offset by 1 complete cycle. In other words you would hear one cycle from the first signal then both combine then you’d hear the one cycle from the second signal after the first stopped. This is similar to what is shown in figure 12 (which shows only 1/2 cycle) but is not the result of the same conditions that were used to make figure 12. At 250 Hz the effect would be as shown in figure 6 because a 1 millisecond time difference corresponds to a 90 degree phase shift at 250 Hz or an offset of 1/4 cycle. At lower frequencies the phase shift would be even less and the signals would tend to add as in figure 2, approaching but never quite reaching the 6 dB increase shown in that figure.
Contrary to phase, polarity affects all frequencies the same way. It makes the positive portions negative and the negative portions positive. Put another way, it simply flips the signal over the same way at all frequencies. With these things in mind, examine figures 12 through 18
Effects of Polarity and Phase On “Real” Audio Signals
Figure 13: This shows a pink noise signal generated as noted above.
Figure 13.
Figure 14: This shows both the original signal in blue and what happens when an identical but phase shifted signal is added to it, as shown in red. The red signal is similar to the combined signal shown in figure 6. Note the increases in signal level and the changes in the waveform (many glitches). However you can also see the combined signal follows the original fairly closely.
Figure 14.
Figure 15: This shows both the original signal in blue and what happens when the phase shifted signal is also reversed in polarity and combined with it, as shown in red. In this case there are huge differences between the original and combined signal.
Figure 15.
Figure 16: To better understand what is going on, this figure shows an averaged or integrated version of the pink noise signal in figure 13. This is basically what would you would see if you graphed the readings from a typical SPL meter for the signal in figure 13.
Figure 16.
Figure 17: This shows the averaged signal from figure 16, in blue, and the averaged combined signal from figure 14, in red. Note that there are primarily level differences (mostly increases). Otherwise the two lines look very similar.
Figure 17.
Figure 18: This really shows what is going on in figure 15. The blue line is the averaged signal from figure 16. The red line is the averaged signal from figure 15. The red line shows that the out of polarity and phase-shifted signal approaches a straight line. Because you are looking at a broad frequency range, you are seeing a severe cancellation of the lower frequencies due to the polarity reversal. However, unlike the low frequencies, the upper frequencies do not completely cancel due to the phase shift. The red line contains primarily high frequency energy. In the blue signal the higher frequencies are the small “bumps”. These can be clearly seen in the red signal and most of them correspond to those in the blue signal.
Figure 18.
Figure 18 is a prime example of what you would hear if you stand exactly between two speakers playing the same signal (i.e. mono) with one speaker out of polarity. The bass will disappear. But, there will always be a difference in distance between you and the speakers due to the spacing of your two ears and probably a slight overall difference in distance between you and each speaker. A difference in distance means a difference in the time arrival and thus there will be phase shifts between the sound from the two speakers. The amount of shift will vary with frequency. Because of the shorter wavelengths at high frequencies, the phase shifts allow most of the highs to be heard. They may be out of polarity but the effect is like what is shown in figure 8. Also, in a room you would also hear sound reflections from the floor, walls, and ceiling. You would only hear something like the red line in figure 18 outdoors away from any reflective surfaces or in an anechoic chamber.
Figure 19.
The small distance between your ears and any small difference in distance from you to each speaker do not cause appreciable phase shifts at low frequencies. This is because of the considerably larger wavelengths. The difference in your distance from each speaker might be only 1 inch (25 mm). However, the wavelength of even a 1 kHz sound is roughly 1 foot (300 mm) and at 100 Hz roughly 10 feet (3 m). At the lower frequencies the polarity difference predominates because the phase shifts due to the difference in your distance from the speakers is very small compared to the wavelengths of the low frequencies. Thus the lower frequency signals, being nearly in phase but out of polarity, will cancel like in figure 4. The lower the frequency the less the phase shift between the two speakers and the greater the cancellation.
A Polarity / Phase Field Trip!!
(As with all physical exercise, check with your doctor first, who might not recommend you do this for some reason.)
Find two railroad tracks, lie across them, and wait.
Two trains, one on each track, come along. Both are right side up and both hit you at exactly the same time. The trains are in polarity and in phase.
The same thing happens again and both trains hit you at exactly the same time. However, this time one train is upside down.
That is a polarity reversal.
The third time both trains are right side up but one hits you first and the other hits you shortly after the first. That is a phase shift.
The last time the second train is upside down and hits you later than the first. That is both a polarity reversal and a phase shift.
Summary
So there you have it. Although this has only touched on a few areas concerning phase and polarity issues, it is hoped you better understand the difference between the two and a few of the effects of each. Remember that the audio frequency range covers wavelengths of over 30 feet (10 meters) at the lowest frequencies to less than an inch (under 25 mm) at the highest frequencies.
While a reversal of polarity will affect all frequencies identically, a difference in time arrival between two otherwise identical signals will have very different effects on the phase between them. The amount of phase shift will be different at different frequencies and this will depend on how much time difference there is between the arrival of the two signals.
Wye-connectors (or “Y”-connectors, if you prefer) should never have been created.
Anything that can be hooked-up wrong, will be. You-know-who said that, and she was right.
A wye-connector used to split a signal into two lines is being used properly; a wye-connector used to mix two signals into one is being abused and may even damage the equipment involved.
Here is the rule: Outputs are low impedance and must only be connected to high impedance inputs—never, never tie two outputs directly together—never.
If you do, then each output tries to drive the very low impedance of the other, forcing both outputs into current-limit and possible damage. As a minimum, severe signal loss results.
“Monoing” Low End
One of the most common examples of tying two outputs together is in “monoing” the low end of multiway active crossover systems. This combined signal is then used to drive a subwoofer system.
Since low frequencies below about 100 Hz have such long wavelengths (several feet), it is very difficult to tell where they are coming from (like some of your friends). They are just there—everywhere.
Due to this phenomenon, a single subwoofer system is a popular cost-effective way to add low frequency energy to small systems.
So the question arises as how best to do the monoing, or summing, of the two signals? It is done very easily by tying the two low frequency outputs of your crossovers together using the resistive networks described below.
You do not do it with a wye-cord.
Summing Boxes Figure 1 shows the required network for sources with unbalanced outputs. Two resistors tie each input together to the junction of a third resistor, which connects to signal common. This is routed to the single output jack.
Figure 1. Unbalanced Summing Box
The resistor values can vary about those shown over a wide range and not change things much. As designed, the input impedance is about 1k ohms and the line driving output impedance is around 250 ohms.
The output impedance is small enough that long lines may still be driven, even though this is a passive box. The input impedance is really quite low and requires 600 ohm line-driving capability from the crossover, but this should not create problems for modern active crossover units.
The rings are tied to each other, as are the sleeves; however, the rings and sleeves are not tied together. Floating the output in this manner makes the box compatible with either balanced or unbalanced systems.
It also makes the box ambidextrous: It is now compatible with either unbalanced (mono, 1-wire) or balanced (stereo, 2-wire) 1/4-inch cables.
Using mono cables shorts the ring to the sleeve and the box acts as a normal unbalanced system; while using stereo cables takes full advantage of the floating benefits.
Stereo-to-Mono Summing Box Figure 2 shows a network for combining a stereo input to a mono output. The input and output are either a 1/4-inch TRS, or a mini 1/8-inch TRS jack. The comments regarding values for Figure 1 apply equally here.
Figure 2. Stereo-to-Mono Summing Box
Balanced Summing Boxes Figures 3 and 4 show wiring and parts for creating a balanced summing box. The design is a natural extension of that appearing in Figure 1.
Figure 3. Balanced summing box using XLR connectors
Figure 4. Balanced summing box using 1/4-inch TRS connectors
Here both the tip (pin 2, positive) and the ring (pin 3, negative) tie together through the resistive networks shown.
Use at least 1 percent matched resistors. Any mismatch between like-valued resistors degrades the common-mode rejection capability of the system.
Termites In The Woodpile
Life is wonderful and then you stub your toe. The corner of the dresser lurking in the night of this Note has to do with applications where you want to sum two outputs together and you want to continue to use each of these outputs separately.
In other words, if all you want to do is sum two outputs together and use only the summed results (the usual application), skip this section.
The problem arising from using all three outputs (the two original and the new summed output) is one of channel separation, or crosstalk. If the driving unit truly has zero output impedance, then channel separation is not degraded by using this summing box.
However, when dealing with real-world units you deal with finite output impedances (ranging from a low of 47 ohms to a high of 600 ohms).
Even a low output impedance of 47 ohms produces a startling channel separation spec of only 27 dB, i.e., the unwanted channel is only 27 dB below the desired signal. (Technical details: the unwanted channel, driving through the summing network, looks like 1011.3 ohms driving the 47 ohms output impedance of the desired channel, producing 27 dB of crosstalk.)
Now 27 dB isn’t as bad as first imagined. To put this into perspective, remember that even the best of the old phono cartridges had channel separation specs of about this same magnitude.
Therefore stereo separation is maintained at about the same level as a high-quality hi-fi home system of the 1970s.
For professional systems this may not be enough. If a trade-off is acceptable, things can be improved.
If you scale all the resistors up by a factor of 10, then channel separation improves from 27 dB to 46 dB.
As always though, this improvement is not free. The price is paid in reduced line driving capability.
The box now has high output impedance, which prevents driving long lines. Driving a maximum of 3000 pF capacitance is the realistic limit. This amounts to only 60 feet of 50 pF/foot cable, a reasonable figure.
So if your system can stand a limitation of driving less than 60 feet, scaling the resistors is an option for increased channel separation.
Checking out key components of the V-Mixing System
The M-480 digital console is a primary component in the steadily expanding V-Mixing system family from Roland Systems Group that also includes other console models, as well as digital snakes, personal mixers, protocol interfaces and the new R-1000 multi-track digital recorder.
The M-480 is the largest console in the series (other models include the 400, 380 and 300), offering 48 mixing channels plus 6 stereo returns (for a total of 60 channels), left/center/right outputs, 16 aux buses and 8 matrices.
Using proprietary REAC (Roland Ethernet Audio Communication) digital protocol, the system is configurable up to 90 inputs and 90 outputs.
A compressor and gate is provided on all mixing channels, with 6 stereo multi-effects processors and 12 graphic EQs available for patching.
For this Road Test, Roland also provided two of the three available models of snake boxes, as well as some CAT cabling on reels. (More about these facets later.)
Feature Set
The first thing I noticed is how lightweight and compact the console is, even in the flight case that it shipped with. It’s very easy for one person to manage the console in the case, as well as pick up or move the console by itself.
The Roland M-480. (click to enlarge)
Dimensions are 29.6 x 24.3 x 9.1 inches (w x d x h), and weight is 44 pounds. The desk is well built and seems like it can easily handle the “rigors of the road.”
Setting up the console without a snake connected is easy because the rear panel offers 8 patchable inputs and 8 patchable outputs. There is also a stereo pair of RCA input jacks and an XLR input for talkback microphone.
Rounding out the connectors on the rear are an optical and coaxial digital output, USB, MIDI and RS232 jacks, and of course the RJ45 etherCON REAC digital snake ports. A recessed headphone jack and volume control are provided on the front.
The large touchscreen is easy to view and tracks well with adjustments. (click to enlarge)
Getting around on the console is very intuitive, and I found that I didn’t even crack open the manual until I tried to map layers and assign effects. Speaking of the manual, it was very easy to use. The sections are logically laid out, and the text is clear and concise on how to accomplish routing, patching, and tweaking. (I wish more product manuals were as clear!)
During testing, I plugged in a mic and then went to plug in my computer to playback some tracks while I familiarized myself with the controls. That’s when I discovered that the M-480 does not include a digital input on the rear.
What I thought were SPDIF connectors were actually stereo RCA analog line inputs – certainly not a deal breaker for me, but some audio folks want/need a digital input on their console. And, note that digital I/O can be found in the S-4000S snake configuration with AES/EBU I/O.
The shuttle wheel and enter buttons, with a clean layout and adjacent to the screen. (click to enlarge)
After an hour or so, I had the hang of the console, and let an audio buddy who had stopped by my shop check it out. He also liked the layout of the control surface, and the large screen made it easy to see what was going on as he made adjustments.
I guess I’m getting spoiled by my iPad, iPhone, and almost every other newer gadget, because I kept thinking the M-480 sported a touchscreen.
However, while that would be nice and perhaps a bit faster in terms of making adjustments, the console’s jog/shuttle wheel and enter buttons were plenty fast for the type of gigs that I usually do. Once I got the hang of the wheel, I had no trouble easily accessing the various parameters and making adjustments quickly.
Snakes & Cables
It was time to set up the digital snake systems and see how they all fit together.
I started with the smaller S-1608 box, which is well built and can be rack-mounted or used as a stand-alone floor box. A large handle at one end makes it easy to move around.
The S-1608 offers 16 XLR inputs and 8 XLR outputs.
Each input comes with an amber LED showing phantom power status, a green LED for signal, and a red LED for clip.
A master power LED and REAC LED let you know the status of the box at a glance.
Controls include a REAC mode switch to configure the system, and a “mute all inputs” switch that I thought was a handy feature until it was accidentally pressed when plugging in a cable at a gig.
I had forgotten the button was there, and spent a few minutes trying to figure out why I could not get signal from the box to the console.
Next up was the larger S-4000S rack-mountable digital snake head, which arrived in the typical configuration of 32 XLR inputs and 8 XLR outputs. It offers remote controllable mic preamps and redundant Ethernet ports as well as the same features as the S-1608. It’s 6RU and weights 37.8 pounds, and it’s modular package lets you configure the system exactly the way you need it.
The S-1608 and S-4000S digital snake systems. (click to enlarge)
Roland also provided some W100S-R reels, each with 100 meters (328 feet) of Cat-5e cable outfitted with etherCON connectors while weighing just over 11 pounds. I’m a big fan of cable reels and have a bunch of various sized reel systems in my inventory, so I don’t say lightly that the W100S-R is a very good reel, and the cable is rugged yet flexible.
Note that the third available stage box, the S-0816, wasn’t supplied for this evaluation. It’s essentially a handy “front of house” box that could be used with the S-1608 to make a simple 16 input x 8 output point to point snake system.
In The Field
The first gig I deployed the package was a small meeting with a few podium mics and some wireless systems, as well as music playback from my laptop. I plugged the laptop into the stereo RCA jacks on the console and used the console inputs for the wireless units.
The S-1608 and S-4000S digital snake systems. (click to enlarge)
The S-1608 was positioned near the stage, used it to run the podium mics back to the console and the console’s main outputs to my amplifier rack for the loudspeakers. Running the Cat-5 snake was way easier and quicker as compared to my analog audio snakes, and Steve, the A2 at the event, had never even seen the console before but within minutes had all of the basics figured out without looking at the manual.
The next gig was a larger corporate event that included a lot of video and computer playback. During a run through I noticed that the audio track in one feed from video world was not in sync with the video portion shown on the screens. I carry a small digital delay I can patch in just for situations like these, but with a digital delay on every input and output in the console, it was easy to add the required amount of delay and line up the words with the lips on screen.
A week later I did a luncheon award banquet in a ballroom that I’m all too familiar with.
The best place for front of house is usually in the rear right corner of the room where’s there’s power and a house audio tie-in panel available, as well as house lighting controls.
Unfortunately, this requires a 250-foot snake run through the back service hallway, with lots of ladder work running the cable up and over the many doorways between the kitchen areas and the ballroom.
Normally it takes two people about an hour to run an analog snake, but in running Cat-5 from the reel, I did the job myself in less than 20 minutes.
Final Analysis
The final show included a 5-piece dance band performing after a corporate presentation. The load-in schedule was tight, and when the band showed up late, we were really crunched for time. The pre-set libraries of the M-480 provided a great starting place for getting a sound quickly.
Then it was just a matter of tweaking the sound to taste, instead of starting from scratch for every channel. The effects units sound great, and it was simple to adjust any parameter.
The M-480 easily integrates with other components such as the M48 personal mixer. (click to enlarge)
Mixing on the M-480 is easy. Unlike some consoles that require you to go through menu after menu to get to certain parameters, on the M-480 the most often used features are available directly via dedicated buttons and knobs on the surface itself. Then if you need to adjust a parameter, it’s usually only one click to bring up that screen and all of the controls are clearly displayed.
The M-480 also offers some features I didn’t really get a chance to explore including 2-track recording direct to a drive via USB, remote access and offline editing, and the ability to cascade two consoles together to make a much larger desk.
Overall, I’m impressed with the console as well as the entire V-Mixing system. It offers tons of features in a compact package, and at a good price point. MSRP for the system tested: M-480 is $11,795; S-4000S is $6,495; S-1608 is $2,095.
Also note that a new version of the M-480, available as a free update, was announced at the recent Prolight + Sound show last month.
To read Craig’s full review of the M-480, and check out other comments from the community as well as to ask questions, go to the Road Test Forum here on PSW.
Craig Leerman is senior contributing editor for Live Sound International and ProSoundWeb, and is the owner of Tech Works, a production company based in Las Vegas.
Registration Opens For AES Conference On Music-Induced Hearing Disorders
The Audio Engineering Society (AES) has confirmed that registration is now open for its 47th International Conference on Music-Induced Hearing Disorders, which will take place at Columbia College in downtown Chicago, June 20-22, 2012.
The conference presents expert knowledge from audio engineers, academic researchers, medical experts and cutting-edge manufacturers, with a total of 18 papers being presented over two days.
“The conference is a great opportunity for people to learn a wide array of perspectives on hearing health in the music industry,” says Michael Santucci, conference chair and president of Sensaphonics Hearing Conservation, one of the conference sponsors. “We have several presenters coming in from Europe, along with experts from several U.S. universities and manufacturers.
“This is a great opportunity for AES members to gain critical knowledge on the issue of hearing health in the music industry, and to network with the leading experts in the field.”
The papers being presented span a wide range of topics relevant to the music industry, including measurement techniques for in-ear monitors and portable music device, new research in measurement and diagnosis of hearing problems, and new hearing health products. In addition to the papers being presented, the conference will also have trade show booths from its platinum sponsors.
Full program details and secure online registration are now available here.
Costs for the AES 47th International Conference on Music Induced Hearing Disorders are $600 for AES members, $700 for non-members, and $300 for students, and includes conference attendance, premium on-site catering (two light breakfasts, two lunches, one full dinner) and related social events.
Hotel room blocks at attractive rates have been reserved for conference attendees at two nearby downtown Chicago hotels. Attendees are encouraged to reserve their spot early to ensure availability.
Double tracking is a very common recording/production technique for almost any genre of music.
When it comes to rhythm guitars, this technique is almost a standard method of recording with single tracking used only for solos.
It’s also a technique that is often confusing for beginners. Double tracking simply means recording the same part twice and panning each to opposite sides.
The guitarist plays a section of the song perfectly, then repeats it as closely as possible on a second track. This creates a wide stereo spread based on the unique nuances in timing and dynamics of each performance.
It isn’t the same as recording in stereo, using two microphones, a chorus effect, or duplicating and delaying one side. Some of these techniques are ways of “faking” or producing “automatic” double tracking, but they’re simply no substitute for an expertly performed double track. There must be two separate performances for the effect to work.
How To Double Track Guitars
1) Record mono rhythm guitar, with either a microphone on a real amp or virtual amp. This track would be panned center.
2) When a good take is achieved, and any punch ins are finished, go through the recorded track and tighten up any timing issues.
Here’s how it sounds with the first guitar along with drums. The guitar is in the middle.
3) After editing, pan this guitar (and any extra mics for this performance) to the left.
4) That was perfect, now play it again! Make a new track and pan it right.
5) Repeat steps 1 and 2 using the same guitar, pickup selection, amp, mic and any other variables unchanged. Making a change will increase the stereo width but will often result in an unbalanced tone.
This repeats for each section of the song and if there are multiple guitar parts written or two guitarists in the band, usually each will be double tracked.
If there are two guitarists in the band, there could be some confusion. Guitarist 1 plays all his parts twice, guitarist 2 plays all his parts twice.
In a simple song this would mean four tracks for the rhythm guitars. Often this gets up to 12 or 16 tracks pretty quickly. Guitar solos are usually right up the middle or “stereoized” with other techniques to make them pop out.
You have to be careful playing the doubled part; if it’s too far off from the original it will make a unwanted ping-ponging effect especially in headphones.
Quad tracking is exactly the same, but you record each part four times. Each take has to be perfectly in sync or it just sounds like a terrible mess.
Poor Alternatives
So why can’t we just duplicate and delay/shift the recording a little for the same effect? Well, simply because it sounds like crap.
This is what happens when you copy the original mono recording, delay the copy by 20 ms and pan each hard left and right.
It still sounds really bad compared to double tracking. I’m not saying don’t ever use chorus, just don’t use as an alternative to the big wide powerful double track sound.
I hope you have found this article useful.
Jon Tidey is a Producer/Engineer who runs his own studio, EPIC Sounds, and enjoys writing about audio on his blog AudioGeekZine.com. To comment or ask questions about this article go here.
Predicting Array Performance: Hanging The PA Right The First Time
Taking the guesswork out of seeing where your sound is going
Back in the good old days predicting the performance of a group of loudspeakers was a hit and miss proposition. We tried to hit all the people and miss the walls. We were happy if we had enough devices to point a transducer everywhere that needed coverage and enough power to make it good and loud.
Complex interactions between devices operating in the same bandwidth, fine level adjustments for individual devices and precise flying angles were the least of our worries. And even if we were worried about such things, we didn’t have the tools to deal with them.
Then came the digital revolution. With the advent of abundant computer horsepower, remote amplifier control and DSP, our capacity to exercise control over sound system parameters took quantum leaps. At the same time, improvements in test equipment allowed manufacturers to give us meaningful data on the performance of the loudspeakers we were driving.
This data combined with the processing power of the modern personal computer made it possible to actually (GASP!) predict the performance of an array before it was hung. Mark IV Audio (read: Electro-Voice, Klark Teknik, Midas and Altec Lansing) was one of the first companies to bring some of these tools to the masses—the AcoustaCADD program was an early example of sound system modeling software.
They also developed a program called Hang Ten to help Electro-Voice MT-4 owners figure out where to attach flying straps to get the boxes to array properly. And anyone who has herked MT-4s around knows that experimenting with different configurations in the real world just wasn’t that much fun.
Later EV produced a program called ArrayShow, which was extremely useful for demonstrating the summing and cancellation between adjacent cabinets hung or stacked in close proximity.
Bose also had its Modeler software. But these products were manufacturer specific, which limited their usefulness.
The next breakthrough came with the introduction of EASE. Although EASE has a distribution agreement with Renkus-Heinz, its loudspeaker database is an unrestricted club. Anyone can join by testing their loudspeakers in a specific manner and submitting the data in the proper form.
Almost all of the reputable manufacturers have basic cabinet data available on their web sites and product CDs. This allows the system designer to pick and choose different cabinets for different applications, mix manufacturers or even, in the case of some of the big dogs, generate EASE data for their proprietary boxes.
We can use this data to predict coverage and SPL levels in a room, set delay times and volume levels of specific cabinets and even to model complex interactions between devices. EASE also does acoustical predictions including reverb time and intelligibility estimates.
For the audio consultant who has time to painstakingly draw a room and insert all of the appropriate wall and ceiling treatments, this is a great tool. But does this really apply to the touring community?
The line array craze has managed to drag some of us kicking and screaming into the world of predictive software. Line arrays only behave like line arrays are supposed to behave when the cabinets interact properly. The “hang and bang” approach leads to extremely uneven results in the real world. So, almost all of the manufacturers fielding these products have created some software to assist their users.
These are not true modeling packages because, with a few exceptions, they only help you determine vertical splay angles needed to cover angled floors, balconies and the like. The horizontal coverage of most line arrays is a fixed quantity. And level prediction with a line array is frequency dependent in the far field.
The point at which we go from the vaunted 3 dB loss per doubling of distance to typical inverse square law behavior (6 dB loss per doubling of distance) changes with frequency, making broadband SPL predictions difficult. But for most of us using traditional cabinets in traditional clusters, there are some very useful tools out there for making sure we hang what we need to hang and point it where it needs to point.
A program I have employed with success is LARA from Integral Acoustics. It uses EASE data, so most commercially made boxes can be imported. It has a convenient library of pre-constructed rooms that correspond to most of the typical venues we encounter.
They are easily modified to reflect the exact dimensions of a particular space. Defining acoustical treatments is not necessary because LARA treats all surfaces as a perfect absorber. In other words, we are only modeling direct field coverage and not trying to predict how the sound system and the room will interact, so the time required to build a room is reduced to a minimum for the well-prepared house engineer many of the venues have floor plans available on the web.
For those that don’t, getting room dimensions can become part of advancing the show. But even if none of the information is available in advance, this program is so quick that a room can be modeled on site while the truck is being unloaded.
Once the room is built, there are a couple of ways to drop the sound system into the model. There is a library function where you can store all of the cabinet models you are carrying. The speaker locations are chosen by using simple X-Y-Z coordinates. You can build a cluster one cabinet at a time or use pre-designed blocks of speakers that you have defined as clusters.
If you build the array with individual boxes you can change the pitch, roll and azimuth of each box independently. If you build with clusters you must change parameters for the whole unit. Each box or cluster can also be adjusted for output volume and delay.
A chart is generated showing the exact location, height, tilt, volume and delay parameters for each box or cluster. This makes it a snap to put everything where and how it was modeled.
The program generates a color SPL map of the room at chosen frequencies. Audience surfaces can be chosen so that the model only shows coverage where the people are. Or we can look at all surfaces so that sound can be steered away from non-audience areas to minimize the system’s interaction with the room. Individual cabinets or clusters can be turned on or off to see what contribution to the overall coverage is being provided by them.
In addition to SPL maps, the program will provide complex summation maps. These show the constructive and destructive interference patterns that are produced by overlapping coverage patterns. This enables you to see comb filters and the lobing they produce.
These types of tools keep coming along to make our jobs more complicated in some ways but simpler in others. And if we can hang the PA right the first time, we can spend our day tuning, tweaking and sound checking. But the bottom line is better sounding shows with more even coverage through the whole venue.
And that’s what it’s all about, isn’t it? Besides, all this computer modeling will keep us out of the back room on the bus!
Sound Wave Propagation: The Bigger Picture Of What We Hear
Every object in nature has a “preferred” or “natural” frequency at which it will vibrate
Previously (here and here), we’ve been looking at sound on a “microscopic” level, examining particle motion as sound propagates through air.
This time, let’s look at a larger picture of sound wave propagation.
A vibrating object will disturb the surrounding air medium causing localized changes in pressure and particle displacement with the transference of acoustical energy in the form of a sound wave.
Waves can be broadly classified as being either transverse or longitudinal. The distinction for each wave type corresponds to the relationship to which direction the particles in the medium move relative to the direction the wave moves.
For a transverse wave, the particles of the medium move at right angles to the direction of the wave propagation.
Examples of transverse waves include ocean waves, vibrating strings on a musical instrument, and light and other electro-magnetic radiation.
For a longitudinal wave the direction of the wave propagation is parallel (same direction) as the motion of the medium particles.
Sound waves in air are unique in that they propagate as longitudinal waves. Figure 1 shows concepts for transverse and longitudinal wave motion.
Figure 1. Transverse (top) and longitudinal wave motion (bottom) showing transmission medium motion and wave propagation direction. Image credit: Sennheiser. (click to enlarge)
A sound wave in air will propagate away from a source as a wavefront with the speed of 343 m/s. The sound particle vibrations travel outward from the source with the same phase. Sources can take the form of simple geometry, called point or monopole (e.g., single loudspeaker or small hole in a wall), line (e.g., moving vehicular traffic or many closely spaced loudspeakers), plane (e.g., top plate of double bass, large building surfaces, or large two dimensional loudspeaker array), or complex comprising two or more simple sources (e.g., musical instrument or machine).
The wavefront takes on a spherical geometry at a distance from the source that is larger than the source dimensions and are called spherical waves. At farther distances the wavefronts appear to be flat (planar) and are called plane progressive waves. Plane waves are sound waves in the simplest form.
The opposite of a plane progressive wave is the standing wave. And yet another wave type is the cylindrical wave, due to a series of point sources radiating in-phase with each other that results in a line source.
SIMPLE MULTIPLES
Waves, regardless of whether spherical, cylindrical, or plane, can be considered to be simple or complex. A simple wave is a wave that comprises only one frequency, such as a sinusoid. A complex wave comprises a fundamental sinusoid and harmonics, with the harmonics either being simple integer multiples (2, 3, 4…etc.) of the fundamental sinusoid or non-integer multiples (1.35, 2.21, 3.05 etc.), as occurs for many percussion instruments.
Through a mathematical process called Fourier Analysis, we can decompose a complex wave into the fundamental and harmonic frequencies, their relative amplitudes, and phase relationships. Figure 2 shows a Fourier analysis of simple and complex waves.
Figure 2. Fourier analysis of simple (top) and complex waves (middle and bottom). Figures to the left show wave amplitude as function of time. Figures to the right show wave amplitude as a function of frequency (fundamental and harmonics). Top figure is for a tuning fork; middle figure is for a clarinet; and bottom figure for a trumpet. (click to enlarge)
Two or more simple or complex sound waves can combine with each other through an additive process called the law of superposition. The resulting complex wave, assuming that the waves are linearly related, is the sum of the displacements due to each sound source.
Sound waves are linearly related when each is directly proportional to displacement. Non-linear acoustic behavior typically occurs when the source sound pressure level exceeds 140 dB.
While conceptually simple, the law of superposition is complex since the particle displacement (ξ) and particle velocity (u) of each sound wave may not always be in the same direction because the sound waves can arrive from any arbitrary location. Remember too, that particle displacement and particle velocity are functions of time and frequency.
Thus, the law of superposition requires a vector summation of waves. Waves from opposite directions will add momentarily together at a finite point in space and then pass through each other as the wavefronts continue propagating in their respective directions.
Waves can add constructively, resulting in greater amplitude, or destructively, resulting in reduced amplitude. The law of superposition describes this. What determines the resultant amplitude through constructive or destructive addition is the relative phase of the waves.
Waves that are perfectly in-phase (0-degree phase difference) add together with no destructive behavior. Waves that are perfectly out-of-phase (180-degree phase difference) add together to result in effectively zero amplitude. Most complex waves have phase relationships that vary as a function of frequency and do not combine in such simple relationships as described above.
JUST BEAT IT
One interesting wave addition phenomenon is that of beats. Beats occur when two sinusoids of slightly different frequency, typically less than 15 Hz apart, combine at a point in space.
Because the two waves have slightly different frequencies, they will have varying phase relationships, resulting in times when the waves are partially in-phase and partially out-of-phase with each other.
Thus, the waves will add constructively and destructively resulting in slowly varying amplitude.
For example, if the two frequencies are 220 and 229 Hz, the sinusoids will be in-phase and interfere constructively 9 times per second and be out-of-phase 9 times per second, and interfere destructively. The sound level will vary from loud to soft at a rate of 9 Hz.
Most people can perceive beat frequencies up to about 15 Hz. Beyond this value the sensation of “roughness” occurs with no beating. Further separation of the two sinusoids results in perceiving each as a separate frequency. This is one basis for determining the critical bandwidths of the ear.
Figure 3 shows the generation of a beat frequency.
Figure 3. Generation of beat frequency (bottom) from two sinusoids of slightly different frequencies (top and middle). (click to enlarge)
When a sound wave approaches a boundary surface, a portion of the incident energy is reflected and a portion is absorbed by the surface. The absorbed sound is the sum of the dissipated losses within the boundary medium and the portion transmitted through the boundary.
The characteristic impedance of the boundary surface determines the ratio of absorbed sound to incident sound. The physical density of architectural materials is higher than air and results in most of the sound energy being reflected away from the boundary surface.
Two broad classes of sound reflections can occur: standing waves and specular reflections. Standing waves are the result of the law of superposition. Specular reflections are not based on the law of superposition. The sound absorption mechanism described above is applicable to both standing waves and specular reflections.
Standing waves result from interference of two or more waves that repeatedly pass through each other when traveling back and forth between the room boundaries. The result is a wave that appears stationary having regions of maximum amplitude (antinodes) and minimum amplitude (nodes).
ON THE SURFACE
For rooms, the standing waves are referred to as room modes. Three types of room modes occur: axial, tangential, and oblique. Each room mode type is supported by an increasing number of room surface pairs.
Figure 4. Axial standing waves fundamental mode (top), second mode (middle), and third mode (bottom). (click to enlarge)
Axial modes require two opposite room surfaces (one pair); tangential modes require four room surfaces (two pairs); and oblique modes require six room surfaces (three pairs).
Axial modes are the most audible. The tangential and oblique modes are respectively 6 and 12 dB less than the axial modes. Figure 4 shows axial standing waves (room modes).
A specular reflection occurs when the incident angle from the incoming wavefront at the boundary surface equals to the reflected angle from the boundary. This reflection phenomenon only occurs when the wavelength of the incident sound is less than approximately one-fourth the boundary surface dimension.
For the above conditions, the reflections can be approximated as rays and laws of geometrical optics apply. Figure 5 shows simple specular reflection. The wavelength for low frequency sound is often equal to or larger than the room dimensions. When this occurs, there are no specular reflections, and wave acoustics is used for analysis.
Figure 5. Specular reflection where angle of incidence equals angle of reflection. (click to enlarge)
One key concept to remember when sound is incident at a physical boundary is the particle velocity (v) and acoustic pressure (p) are 90° (π/2 radians) out-of-phase with each other. At a boundary, the particle velocity will be zero and the pressure will be a maximum.
This is important when considering sound absorption of materials: the maximum absorption at the lowest frequency of interest will occur at a distance equal to λ/4 from the boundary. At this distance the particle velocity will be a maximum for the frequency corresponding to λ/4. Since most “acoustical” materials are frictional absorbers, a maximum particle velocity will result in the greatest sound absorption.
Resonance is the reinforcement of sound by synchronous vibration. Every object in nature has a “preferred” or “natural” frequency at which the object will vibrate. Imposing an oscillatory force of the same frequency as the object’s natural frequency will cause the object to vibrate at maximum amplitude will little energy input from the exciting force. Changing the “forcing” frequency by a small amount will effectively decrease the resonant response.
DO THE MATH
When examining a system at resonance, we will observe a maximum peak at the resonant frequency (fO).
The height of the resonant peak will depend on the degree of damping within the vibrating system.
The resonant frequency response can be either very sharp, centered around a high amplitude narrow frequency bandwidth (Δf), or quite broad with lesser amplitude.
The desired acoustical response will determine which characteristic is best.
Systems that have a sharp resonance characteristic are called “high Q”; those with a broad resonance are called “low Q”.
The Q term refers to quality factor and can be calculated by the following equation:
where,
Q = quality factor, unitless
fO = resonance frequency, Hz
Δf = frequency bandwidth, Hz, taken as the -3 dB down points about the resonant frequency
Figure 6 shows both high Q and low Q resonance response.
Figure 6. High Q resonance (without damping) and low Q resonance (with damping). (click to enlarge)
RULES OF THUMB
Try to remember the following, or key them into your PDA or computer “cheat sheet.”
- Most everyday sounds we encounter are complex waves comprising many frequencies.
- Simple point sources radiate sound as spherical wavefronts assuming the wavelength is smaller than the source dimensions. At greater distances from the source the wavefronts flatten out and become planar.
- Sounds combine due to the law of superposition and the resultant amplitude depends on the amplitude, frequency, and relative phases of each wave.
- Waves will reflect from room surfaces. Specular reflection requires the wavelength to be at least one-fourth the room surface dimension.
- Resonance is the response of a system when driven at its natural frequency. The sharpness of the resonance will depend on the damping within the system. Rooms have special resonant phenomena called room modes.
Neil Thompson Shade has 30 years of experience in consulting and teaching acoustics, noise control and sound system design. He is president and principal consultant of Acoustical Design Collaborative, Ltd., located in Baltimore, and he has also been taught acoustics, sound system design, computer modeling and related topics at the Peabody Institute of Johns Hopkins University.
The CAD Audio AS32 Acousti-shield was developed as an essential accessory for recording; when effectively utilized, it can substantially reduce unwanted reflections, echo flutter and environmental acoustic interference.
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The AS32 Acousti-shield is available at an estimated street price of $129.
Science Or Snake Oil? The Facts Behind The Hype About Loudspeaker Wire
Marketers must come up with reasons for why you should buy their wire. To claim that their wire is better, they must first identify, in some cases invent, a difference
Too many good folks have been separated from their hard earned money by hyperbolic claims about loudspeaker wire. There will always be people with more dollars than sense, but they don’t last very long in professional audio.
I speculate there aren’t many (if any) of you who would pay thousands, or even tens of dollars per foot for speaker wire.
A very basic practice in merchandising is called differentiation. Marketers must come up with reasons for why you should buy their wire. To claim that their wire is better, they must first identify, in some cases invent, a difference.
This search for a selling proposition has sometimes focused on “skin effect.” It’s a real effect and describes how at very high frequencies, electrons travel in the outer layer or “skin” of signal conductors.
Another related property is that high frequency signals travel faster than low frequencies through the same cable.
These phenomena are dealt with appropriately in very high frequency applications with several techniques. For example, one particular type of botique wire is made up of a large number of very small conductors braided or woven into one cable, producing a large surface area or “skin” for a given cross sectional area.
Another approach for high power high frequency power transfer is to use a hollow conductor, resembling a section of copper tubing. If the electrons are going to ignore the center of the conductor, why pay for it?
This is not an issue for audio professionals, working at mere audio frequencies of 20 Hz to 20 kHz. Perhaps it would be if we were sending audio over many miles, like the telephone company in its pre-digital days. They had to periodically correct for waveform smear.
But at the speed that electricity travels, our typical path distances are much too short to be an issue.
Out Of Perspective
Wire is not very sexy or easy to create real marketing hooks for, but it can actually make an audible difference. The dominant mechanism is simple resistance.
It’s perhaps ironic that the “snake oil” markers of loudspeaker wire will exaggerate some real but insignificant parameter far out of perspective while compromising the real deal.
Forget the hype, what’s important for loudspeaker wire is that it exhibit low impedance that is resistive in nature. If the wire has a significant impedance component (reactance) that changes over the audio frequency spectrum, this can form a simple divider with the loudspeaker’s resistive impedance and cause a frequency response error.
In addition, since loudspeaker impedance will vary quite a bit over frequency, even a perfectly resistive speaker wire will cause errors. The magnitude of this frequency response error will increase proportionately as the wire’s resistance increases.
Purveyors of “funny wire” don’t bother to make claims about useful metrics like resistance since that is already defined by the wire size or gauge (known as “American Wire Gauge” or AWG for short). That would be like advertising how many quarts were in their gallons!
However, frequency response errors caused by wire resistance are one of the very real things that people actually do hear. I find this following anecdote instructive.
From a discussion with one individual who was certain that he heard a significant improvement when using his “Snake-O Special” loudspeaker wire (name changed because I don’t remember it), I determined that the wire gauge he was using was marginal for the length of his run. The wideband loss of volume caused by a wire’s resistance will be very difficult to hear without a side-by-side comparison.
But the difference in amount of loss caused by the loudspeaker’s changing impedance at different frequencies can easily cause a frequency response error that is probably what he heard. It’s easy to imagine how a rising impedance at high frequency could cause a pleasant sounding treble boost. Just listen to how clean and clear these “Snake-O Specials” sound!
There are several strategies to manage these real losses from wire resistance. The obvious one is to throw more copper at the problem. Heavier gauge wire with lower resistance will exhibit lower losses for a given run length.
Another fairly obvious approach is to locate the amplifiers as close as possible to the loudspeakers to keep the run length as short as possible. A third less obvious approach is to scale up the intermediate signal voltages.
Constant Voltage
There are cases, such as in large distributed sound systems where neither of the first two approaches is cost effective.
You can’t afford to put a separate amplifier at every loudspeaker location, and sending sound sources over long distances with acceptable losses would require very heavy gauge wire.
The solution borrows a strategy from high voltage power distribution systems such as the one used by utilities to bring electrical power to our homes.
The power developed within a given load increases with the square of the terminal voltage (E^2/R). However, wire’s losses only increase linearly with current flow, because the voltage developed across the wire is a simple function of its resistance times that current.
Power engineers determined that by raising the voltage carried by transmission lines they could increase the power being carried exponentially while simultaneously reducing the losses due to current flow.
The utility company accomplishes this magic with step-up/step-down transformers. By “transforming” a typical 100-amp at 240-volts residential service, up to tens of thousands of volts at the transmission line the 100-amp draw is reduced to the far more manageable level of 1 amp or so. Wire losses are 1 percent of what they would otherwise be.
Similar manipulations go on in “constant voltage” distributed sound systems but rather than stepping up the voltage to thousands of volts the standard for U.S. systems is 70-volt, with Europe using a slightly higher 100-volt standard. The rest of the world tries to conform to one of those two standards.
Of course, the audio signal isn’t actually held constant. The voltage at rated power is. Both 5 watts and 500 watts constant voltage systems deliver the same nominal voltage for distribution.
The goal in any effective distribution system is to deliver as much power as possible to do useful work in the load and waste as little as possible heating up the wire. In a simple distributed sound system sending a few watts of announcements across a few hundred feet of factory floor, the typical low voltage system could drop as much power in the speaker wire as would reach the loudspeakers.
By stepping up to 70 volts and back down again at each loudspeaker the balance of power delivered versus lost is more respectable.
To put numbers to this concept, say we are trying to deliver 1 watt each to two loudspeakers located 100 feet distant from an amplifier using 24 AWG wire. Because we must count wire losses from the feed coming and going, 200 feet total of 24 AWG exhibits resistance of approximately 5 ohms.
Figure 1: Two different ways of realizing one watt at two loudspeakers. Click to enlarge.
To realize 1 watt at each loudspeaker, there would need to be more than 4 watts into the wire at the amplifier end. (Over 2 watts gets wasted as heat in the wire).
If we first step up the audio to a nominal 70-volt level the current drops to such a low level that the same wire would only waste 0.14 watts while delivering the same 1 watt each to the two loudspeakers.
As useful as constant (high) voltage systems are for managing wire losses, they don’t make sense for point-to-point runs in sound reinforcement systems. The main drawback is the size of the step-up and step-down transformers required.
To put this in perspective, the size of the transformer has to double every time you drop the frequency an octave. To cleanly pass 20 Hz both step-up and step-down audio transformers would have to be three times the size of a conventional amplifier’s 60 Hz power supply transformer.
Keep It Short
The good news for most live sound applications is that we don’t have to tolerate extremely long wire runs. By locating power amplifiers near the loudspeakers we can keep wire runs reasonably short. At these shorter distances we can easily afford heavier gauge wire.
While power losses are now manageable, it is worthwhile investigating the next dominant consideration in sizing loudspeaker wire.
Frequency response errors will be caused by the voltage divider created between the wire’s fixed resistance and the loudspeakers changing impedance versus frequency.
Figure 2 and Figure 3 show two representative loudspeaker impedance plots, pulled from the Internet.
These are not offered as either worst case or typical.
From the impedance plot in Figure 2, if we ignore the extreme low frequency, this loudspeaker exhibits a maximum impedance greater than 17 ohms, with a significant region of the upper bass down around five ohms.
Figure 2: This loudspeaker exhibits a maximum impedance greater than 17 ohms. Click to enlarge.
Meanwhile, Figure 3, while more complex, covers a similar impedance range, with a maximum around 16 ohms and a minimum around six ohms.
To derive a frequency response error we need to compare the drop at maximum impedance to the drop at minimum impedance. The equations below calculate that drop for a given wire resistance.
Note: To simplify this analysis we will assume all loudspeaker impedances to be resistive. While not strictly accurate, loudspeaker impedances will typically be resistive at impedance minimums and any errors caused by load phase angle at the impedance maximums will not be significant for the sake of this analysis.
Minimum Voltage drop= V max = Z max /(Z max +Z wire)
Maximum Voltage drop= V min = Z min /(Z min + Z wire)
Frequency Response deviation= FR max = -20 Log10 (V min/ V max)
Solving for 1-, 0.5-, and 0.1-ohm wire resistance we get:
Spkr 1 (17/5)........ -1.09 dB…... -.57 dB…... -.12 dB
Spkr 2 (16/6)....... -.81 dB…..... -.42 dB…... -.09 dB
Figure 3: While more complex than the loudspeaker in Figure 1, this covers a similar impedance range, with a maximum around 16 ohms. Click to enlarge.
Another related consequence is how wire resistance degrades effective damping factor.
While damping factor is usually though of as a power amplifier characteristic, in reality the wire selection can easily dominate actual damping available at the loudspeaker.
In the above examples the 1-ohm wire would by itself cause a rather weak damping factor of 5 or 6 (regardless of the amplifier’s rated damping factor).
Using the 0.1-ohm wire predicts a more respectable 50-60 damping factor, with some small additional degradation due to the amplifier’s output impedance.
Damping factor deserves a more extensive discussion, but for this exercise we will assume that the amplifier’s output impedance is small with respect to our wire’s resistance.
Gauging Gauge
It’s difficult to predict a precise threshold for audibility of frequency response errors.
Controlled listening tests have suggested that differences as small as a few tenths of a dB can be audible.
To satisfy the dual goals of minimizing frequency response errors and not degrading damping factor for the example loudspeakers selected, I am comfortable with targeting a total wire resistance on the order of 0.1 ohm.
Wire’s resistance varies linearly with length. To keep the total resistance below our target limit of 0.1 ohm we must first project the length of our desired wire run, and then select a wire gauge whose resistance per unit length keeps us within the total resistance budget.
Don’t overlook that the wire length is actually twice the run distance as we must consider the feed to and return from the loudspeaker as effectively in series. We must also add in contact resistance for the connections at all ends.
Lets look at how this works out for a practical example of a 20-foot run. First, we double that to 40 feet to establish the true signal path length.
Then we need to account for contact resistance. I’ve seen Neutrik Speakon (or copies of that connector) rated as low as 1mOhm (1/1000th ohm) per contact when new, and guaranteed
< 2 mOhm over life.
Because there are four connections in our total path lets budget .008 ohms for connections. Subtracting this 0.008 ohms from our 0.1-ohm target leaves us .092 ohms for wire. Dividing this 0.092 ohms by the 40-foot length calculates out to 0.0023 ohms per foot.
Plugging this into the equation for wire gauge:
AWG = 10 ×log 10 R +10 (note R is per 1000 feet)
We get: AWG = 10x log 10 (2.3) +10 = 13.6 gauge
This is a little cumbersome, but once you have established an appropriate gauge for a nominal run length with your specific system. This gauge can be scaled up or down for other run lengths.
Wire resistance changes linearly with length. It changes non-linearly with gauge. A convenient property of wire gauge is that the wire’s resistance will double for every 3-step increase in gauge (AWG). Conversely the resistance will drop in half for a three-step decrease in gauge.
Based on this same example and rounding off to 14 AWG, we can expect similar performance from a 40-foot run using 11 AWG wire, and a 10-foot run would only need 17 AWG. This numbering convention gets a little unusual below “0” AWG.
One step below (larger than) “0” is “00”, and “000” is two steps larger than “0”. I don’t expect to see speaker wires this large, as they would be very difficult to effectively interface with amplifiers and loudspeakers.
Using this example to size wire for your system will get you in the ball park, but it will be more accurate to use actual impedance specifications for your loudspeakers. Manufacturers of professional loudspeakers routinely publish this information.
Remember, use only the impedance max/min deviation within the audio bandwidth of interest. It doesn’t matter what a tweeter’s DC resistance is or a woofer’s 20 kHz impedance, since you won’t be listening to them there.
You also may want to tighten or relax the acceptable frequency response deviation. Better yet, look at your loudspeaker’s typical frequency response and determine if the response errors caused by your wire losses are additive or corrective.
While I don’t suggest trying to dial in corrective equalization using wire losses, if the error is making your system flatter you can afford to be less aggressive in sizing your wire AWG as long as you keep damping and power losses under control.
John Roberts is a long-time professional audio product and system designer and has been writing outstanding technical articles over the past two decades.
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