Why Not Wye? When Combining Two Signals Into One Is Not A Good Idea

Anything that can be hooked-up wrong, will be. You-know-who said that, and she was right


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.

Dennis Bohn is vice president of research and development for Rane Corp., and is a noted technical writer. For more articles of this nature, see the Rane Pro Audio Reference at www.rane.com, and note that these materials are also available for purchase in book form. Courtesy ProSoundWeb and Rane. Used by permission.

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Tweaking DSP for Stage Monitors: Tips & Tricks To Maximize Potential

Some problems cannot be completely cured, but with some DSP horsepower and modern test equipment, they can be minimized

November 17, 2009, by Bruce Main

bruce dsp

In the days when digital signal processing (DSP) first stalked the arena, it was the guys at front of house that had all the fun. They would ignore their comm light for long periods of time while staring intently at asymmetrical crossovers on their laptop displays.

But now that DSP is ubquitous, the guys at the other end of the snake are beginning to experience the joys of audio in the digital domain.

The real beauty of DSP in monitorland is in optimizing wedges and side fills so that little or no equalization is needed to minimize feed back.

You might have a rack full of graphics, but the show will start out with all of them set flat. These units will only be used for on-the-fly feedback reduction or to ”personalize” a mix.

And with DSP, a preset can be designed for each model in your inventory, particularly useful if you have multiple wedge types. So let’s get on to the tweaking part.

Very few monitor cabinets are inherently “flat”. There are compromises made due to factors like box size versus low frequency driver selection, box size versus horn selection, low frequency driver dispersion versus. crossover frequency, etc.

Your favorite microphones also have frequency anomalies, especially in off axis response. Combine the two and things can get quite complicated.

Many of these problems cannot be completely cured, but with some DSP horsepower and modern test equipment, they can be minimized.

The first step is to achieve flat on-axis response from your monitors. I would highly recommend that if you don’t already own (Gold-Line) TEF, (Rational Acoustics) Smaart, or at least a real-time analyzer (RTA) then beg, borrow or steal some audio analysis equipment for this portion of the procedure.

Measured with TEF, a high-frequency device before equalization. (click to enlarge)

At an AES convention a few years ago, an equipment manufacturer that I have a relationship with set up a loudspeaker, an EQ and a pink noise source in their booth. Passersby were invited to try to equalize the pink noise by ear to achieve a flat response from the loudspeaker.

One person was spot on, but most of the results were pretty scary, even though the participants were all audio professionals. We need accuracy of +/-1 or 2 dB to give us a truly flat baseline to work from.

Some may have the ears to accomplish this, but most of us don’t. If you’re using an RTA, be sure to get one that displays in increments as small as 1 dB.

Again with TEF, the data of a high-frequency device after equalization. (click to enlarge)

Measuring the low frequency response of a loudspeaker minus the room’s acoustic contribution is a somewhat tricky proposition. A 40 Hz wavelength is approximately 28 feet long.

In order to measure those frequencies properly with a time windowed measurement system the window has to be long enough to contain at least one full wavelength.

Unfortunately that means that it is long enough to contain room reflections that contaminate the measurement. At higher frequencies this is not a problem because the wavelengths become short enough for the windowing to provide anechoic measurements.

Because real-time analyzers are time blind they include room reflections at all frequencies.

Testing Methods
There are a couple of ways to deal with this. One is to do your testing outdoors far enough from any buildings to minimize reflected energy.

My favorite approach comes from Don Keele, one of the really smart guys in our industry. He places the measurement microphone about one inch from the center dome of the woofer, takes a measurement, then places the mic one inch from the port, takes a measurement, and then sums the two responses.

The signal-to-noise ratio of the measurement is improved greatly because of inverse square law gains that result from being so close to the source.

A word of caution here: if the device under test has a maximum output of 120 dB at one meter, it’s output at one inch will approach 136 dB. This may be enough to do bad things to your expensive measurement microphone.

So start out at a fairly low volume level and work your way up. This test will give you a good idea of what is going on from 200 Hz on down.

The first parameter I set is a high-pass filter to prevent signals that the box is not capable of reproducing from wasting power and potentially damaging components.

Then correct any large EQ anomalies with parametric filters. The microphone can be moved to normal listening distance for this and all subsequent tests.

Make sure that the distance is at least three times the longest dimension of the box under test. This puts us in the far field. I do this test with no crossover engaged for the low-frequency device.

If you test from 200 Hz to, say, 5 kHz, you get a good idea of the total low frequency response curve of the box.

When you’re done, a configured system should look something like this on the DSP software. (click to enlarge)

Next, look at the upper range of the device’s response curve. There will be an obvious point where the amplitude drops off or the speaker gets into breakup modes represented on the test display by narrow notches and/or peaks.

Use parametric filters to flatten the response as much as possible within the useable range of the device. Choose an upper crossover point for the woofer that filters out the nasty modes and only utilizes the relatively flat part of the speaker response.

The high-frequency test is next. I prefer doing this test with no crossover engaged, however, the sweep frequency must be started at a high enough frequency to avoid damaging the driver.

Check the manufacturer’s recommendation for the lowest suggested crossover point and start your sweep there. If full range pink noise is being used as the test signal, start with the crossover engaged to protect the driver.

Using the parametric filters in your DSP of choice, correct for frequency response anomalies.

If the box is using a constant directivity horn you may need to use a shelving filter to increase the high frequency output above 2 to 3 kHz. Get the response as flat as you can across the full frequency spectrum.

Remember, if you leave a 3 dB peak in the response and it happens to coincide with a 3 dB peak in the vocal mic response, it will cost you 6 dB of headroom.

You should discover an overlapping frequency range where the low- and high-frequency devices are behaving in a fairly linear fashion. The crossover can be set anywhere within that region. As a general rule, if the horn is small, set the crossover towards the high end of the overlap zone.

Larger horn mouths provide pattern control down to lower frequencies, so if the horn is larger, you can set the crossover point lower while maintaining good directivity from the device.

Next, the levels and time alignment between the low- and high -requency sections should be set. With the mic on-axis and centered between the horn and woofer, do a full range sweep. Set the crossover outputs so the average volume level is the same across the entire frequency spectrum.

Then look at the frequency and phase response at the crossover frequency. Pretty ugly, eh?

Using an impulse response or ETC measurement, look at the arrival times for the two devices. Set the alignment delay on the DSP to eliminate the time arrival offset.

Now look at the frequency and phase again. Better? You will need to do a little fine tuning to get the flattest possible phase line.

If you’re using an RTA, this part is harder. Try inverting the polarity of the high output on the crossover. You should see a notch at the crossover frequency. Adjust the delay until the notch is at its deepest. Reset the high polarity back to normal.

If the time offset is correct the notch should disappear. If your RTA has a 1/12th octave mode, it will be easier to see.

Some real time analyzers have a loudspeaker timing analysis feature as well. Using asymmetrical crossover slopes can produce better (or worse) off-axis response.

Experiment with this if you have time, but this magazine isn’t long enough to cover all the possible permutations. Program in some brick wall limiters just before the little red lights on the amps start to dance.

Less From Two Than One?
Save these settings as a preset in your DSP of choice and repeat the process for each type of monitor wedge in your inventory as well as side fills and drum monitor rigs. You can also use these settings as a basis for multiple wedge setups.

But remember that when you use multiple cabinets of any sort, comb filtering will occur because of the time arrival differences. These peaks and notches are non-minimum phase. That means that they are not “EQ-able”. (Is that a word? It is now!)

Because of this, sometimes it’s possible to get less output with two wedges than with one.

But riders being what they are, go ahead and do a preset for dual wedge setups. The crossover and time alignment settings will remain the same, but you may get some summing in the low frequencies.

Use the RTA to check the frequency response because a TEF sweep will be too frightening to look at. If you need a preset for absolute maximum output with a particular vocal mic, try putting it on a stand exactly as if you were setting up for a show. Plug the mic into the test microphone input on the test rig.

The response will be a combination of the speaker under test and the off axis response of the microphone. EQ the response to be as flat as possible. It may not sound pretty, but it will get loud. (At least until the singer cups the microphone, sealing off the back of the cartridge, turning it into an omni).

Voila! A look at the final measurement result, courtesy of TEF. (click to enlarge)

You may also want to do presets for full range response or one with a higher frequency on the low-cut filter for vocal only. Sometimes it is beneficial to attenuate the lows for an acoustic set to avoid exciting acoustic guitars or pianos. Save a preset and switch back and forth at the appropriate times.

With the current crop of DSP devices, it’s not uncommon to find configurations like four input, eight output that work perfectly for either four two-way mixes or two three-way side fills and a cue wedge. Look for routing flexibility and the ability to store lots of presets.

But most of all, listen to the units. Audio quality varies as much with digital equipment as with analog.

It’s too easy to buy this type of device based on a laundry list of features and functions when the most important thing is great sound.

Bruce Main has been a systems engineer and FOH mixer on and off for more than 30 years. He has also built, owned and operated recording studios and designed and installed sound systems. Courtesy ProSoundWeb.

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700 MHz Wireless Changeover: If not now, WHEN?

In August 2008, the FCC proposed a rule change prohibiting all Part 74 devices (this includes wireless microphones, intercom, and monitors) in the 700MHz band shown above as early as February 18, 2009. We are still waiting for the FCC to issue the final rule, and thus a specific date. When the DTV transition was delayed until June 12, 2009 we thought maybe that would be the date. Now that has come and gone and we still have no guidance from the FCC for a ‘vacate by’ date. The new owners of the spectrum are urging the FCC to require Part 74 devices migrate out of the spectrum no later than February 18, 2010. It is not a question of IF a ban will be imposed but WHEN. Technical presentations to the FCC have indicated there is a strong possibility of interference between Part 74 devices and the new commercial and public safety uses. Public safety has already deployed networks in 40+ markets.

If you have equipment in this spectrum, there are several reasons you should be migrating out of the 700Hz band.

  1. You do not want to risk interfering with public safety life saving operations.
  2. Most 700MHz gear for the US market is outdated and you will benefit from new features and bandwidth.
  3. Replacement parts are already becoming obsolete as reputable manufacturers curtailed or eliminated US models operating in this range two years or more ago.
  4. The financial impact will be immediate if you wait till you receive interference or a “violation” notice.

Planning, budgeting and replacing over the next few months will mitigate the burden on your finances. Plus several manufacturers are offering trade-in rebates that expire in December 2009. Links are also shown for manufacturers that can change the frequency range or re-band certain models.

We often get asked, “Will the FCC really come looking for devices operating in the 700MHz band?” The short answer is YES. Unlike other countries, the FCC does not have a fleet of detector vans searching cross-country for violators. Enforcement will mostly be based on complaints received from the new owners of the spectrum. The FCC already has a presence at major events so it is reasonable to assume they will be monitoring the 700MHz band for unauthorized use. The FCC can also utilize selective detection/enforcement localized around a geographic or high profile area. It is extremely important to understand that FCC fines are steep, typically starting at $10,000. (Sending a junk fax or telemarketing to someone on the ‘Do Not Call’ list is $4500 per fax/call.) Should you ever find the Enforcement Bureau ‘knocking on your door’ you are well advised to cease doing whatever brought them there and avoid a repeat visit.

An example of the interference problems you may encounter in the 700MHz band already exists. Qualcomm bought TV channel 55 (716-722MHx) nationwide in a previous auction and have already deployed a video streaming service call Media Flo or Flo TV in 40+ cities/areas. That number is likely to double by the end of 2009 and significantly increase coverage in 2010. Other users of the spectrum are likely to deploy similar broadband transmissions like the Qualcomm signal shown on the left. If is close enough and strong enough you will not find a usable frequency in that channel.

Most users with 700MHz band equipment have/will upgrade to equipment that operates in a portion of the remaining UHF TV channels 14-51 working around DTV signals. Planning is critical to getting the right frequency range for your location.

© Professional Wireless Systems

David McLain | The Mixer Guy | CCI SOLUTIONS
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Churches affected by FCC changes on wireless-microphone frequencies

DESOTO, Texas (ABP) -- While the digitization of television signals June 12 garnered wide publicity, many churches were left unaware their wireless microphone systems also were affected.

The Federal Communications Commission auctioned off the lower half of the 700-megahertz band to several telecommunications companies and reserved the upper half of that spectrum for law enforcement, fire and safety operation and municipalities. Many of the sound systems in churches operate in that 700-megehertz frequency.

So, what has the impact been on churches?

“That’s going to vary widely on where the church is located,” said Rex Campbell, media-services manager for the Baptist General Convention of Texas. Churches in rural areas may not notice any problem, he noted.

“In the more urban areas, it’s a little different. The more crowded the airwaves are, the more likely there will be interference,” he explained.

Even churches in cities may not see an immediate difference, Campbell said, because it will take a while for the telecommunications companies to make use of the bandwidth they have purchased.

But beyond the question of efficiency, churches also have wrestled with issues of legality. Some experts, such as Tim Hendrix, a senior accounts manager of Ford Audio Video in Dallas, insist it technically became illegal June 12 to use a sound system that falls into the 700-megahertz range.

But finding a definitive answer from the FCC -- either by wading through regulations on the agency’s website or by phoning to ask -- can be problematic.

Hendrix noted it technically has been illegal for several years to operate any wireless-microphone system without a license from the FCC, but the market exploded so quickly that it exceeded the government’s ability to enforce its regulations.

“I don’t know how the FCC would ever enforce it, but eventually if someone continues to use a wireless mic in that 700-megahertz range, they will get nothing but static,” Hendrix said. But he added no one knows when that interference may begin.

He also agreed that churches in urban areas may feel the effects first.

Wireless-microphone manufacturers no longer sell systems in the affected range, Hendrix said. Most are offering rebates to churches that want to trade their old systems in for new ones in a different range. Those rebates, however, are prorated based on the age of the existing system.

At Hampton Road Baptist Church in DeSoto, Texas, where Hendrix operates the audio system, the rebate was so small he said it would be almost consumed by the cost of shipping the old system back to the manufacturer.

So, Pastor Jerry Raines and a team of missionaries, while on a mission trip to Brazil later in July, will take the church’s old system to donate for use by churches there.

“These mics are still good anywhere else in the world,” Raines explained. He said the microphones would be given to a local missionary who will then dispense them to churches that have the capacity to use them.

By George Henson
from the Associated Baptist Press, used by permission

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Wireless Mics: Spectrum Reallocation and the TVBD Ruling

Recent changes in Federal Communications Commission (FCC) policy are affecting operation of wireless microphone and monitoring systems in the United States. It’s important for production professionals to understand these changes and apply best practices to ensure reliable operation of their equipment.

On June 12th, 2009 the digital TV transition took place. This transition will NOT have an immediate impact on wireless microphone users, however wireless mic users will lose access to the 700 MHz band at some point.

Wireless microphones primarily operate on frequencies in the UHF TV spectrum. They are considered licensed broadcast auxiliary devices that a broadcaster, or broadcast content provider, may operate on locally vacant TV channels. For example, channel 25 (536-542 MHz) is not used for TV broadcast in Boston. Therefore production may operate approximately eight wireless mics tuned to different frequencies between 536 and 542 MHz.

It’s important to remember that two separate and distinct issues have been progressing in parallel:

1) the Digital Dividend and 2) the TVBD Ruling aka the White Space Debate:

Digital Dividend:

This relates to the reallocation of TV channels 52-69 (698 MHz to 806 MHz), generically called the 700 MHz band. Once analog TV terminates, all full power TV broadcast will be consolidated below channel 52. The 700 MHz range will partially be used for emergency communications in channels 63, 64, 68, and 69. The rights to use the majority of the remaining channels were auctioned to telecom companies such as AT&T, Verizon, and Qualcomm to provide what is being termed as advanced wireless services (AWS). This raised billions of dollars for the federal government and was therefore called the digital dividend. Although the FCC has not made a final ruling, it looks immanent that wireless mics will have to vacate this range. The pro audio industry has been lobbying for a grace period that would allow operation of existing equipment beyond the DTV transition date. Hopefully the FCC will grant this request.

The 700 MHz band can be subdivided in the following manner:

A) Emergency communication channels:

63-64 (764-776 MHz) and 68-69 (794-806 MHz).

B) Auctioned spectrum: 698 – 758 MHz; 776-788 MHz

The telecom companies have not completed building the infrastructure for their AWS. AWS will probably become active at the end of this year only in major cities, and then eventually spread. Functional, 700 MHz wireless mics will continue to work reliably for many months after the DTV transition date, maybe even years in some parts of the country.

C) Block D: (758-763 / 788-793 MHz) scheduled for future auction.

Block D was envisioned to be used as a private – public partnership. It did not solicit a minimum bid in the FCC auction, probably because the winner would have had to share it with municipal agencies. Therefore they are not allocated to any one entity yet. Functionally, these frequencies look clear for wireless mics for the foreseeable future, until the FCC successfully auctions Block D

Bottom line: wireless mic users will lose access to the 700 MHz band. The bright side is that as more analog stations terminate their broadcasts, this will open some TV channels below 698 MHz. TVBD Ruling (The White Space Debate)

In November, the FCC released its rules allowing a new class of unlicensed consumer electronic products to operate in locally unused TV channels, just as wireless mics have done for years. These forthcoming products have previously been referred to as white space devices (WSD) but are now called TV Band Devices (TVBDs). They will mainly be used as broadband access devices.

TVBD are categorized as:

1) Fixed

These are allowed to operate with effective radiating power up to 4W on channels 2-51, with the exceptions of channels 3, 4, and 37. They are prohibited from operating in a channel adjacent to an active TV station.

2) Personal/Portable

Due to their mobile nature, these devices are the most concerning for production professionals. However,

portable TVBDs are restricted to channels 21–51, and are also not allowed in channel 37 (reserved channel for radio astronomy and medical telemetry). They are limited to 100mW operating power or 40 mW if operating in a channel adjacent to an active TV station. This moderate power will reduce their range and therefore the possibility to cause interference.

Licensed operation of wireless mics takes precedence over TVBD. TVBD must coordinate around active licensed wireless mic systems.

The rules include several safeguards to avoid interference to wireless microphones:

Spectrum Sensing

TVBDs must include the ability to listen to the airwaves to sense wireless microphones (in addition to TV stations). Until they can demonstrate through “proof of performance” that they can reliably sense wireless mics and avoid causing interference they must also use a:

Geolocation/Database system

TVBDs must use location sensing in conjunction with a database of registered broadcast license assignments. The database will also include a list of protected areas for wireless microphones such as entertainment venues and sporting events. TVBDs must first access the database to obtain a list of permitted channels in the area before operating. A TVBD that lacks this capability can operate only under the direct control of a TVBD that has it.

Reserved channels

Personal/ Portable devices will be barred from channels from 14 – 20 (470 – 512 MHz). In addition, in 13 major markets where certain channels between 14 and 20 are used for land mobile (municipal and public safety) operations, two channels between 21 and 51 will be reserved and available for wireless microphones. These will be the first open (non-TV) channels above and below channel 37.

This means, at minimum, 16 wireless mic or monitoring systems (8 in each TV channel) can be used simultaneously in any venue. When using our equipment with high linearity (extreme suppression of harmonic distortion known as intermodulation) the number increases to at least 20 systems (10 in each TV channel). Protected areas will be able to operate many more channels.

Multi stage and studio properties can also effectively increase the number of systems in use through:

1) Physical distance and transmitter output power management

This can be augmented by a balance of other techniques such as shifted coordinated frequency sets (same frequency spacing but offset by 100 kHz or more), zone isolation (natural or enhanced shielding between rooms), directional antennas, and filtered distribution systems.

2) Time multiplexing:

Using systems in different rooms at different times

New Approach

There are a couple of techniques that can be used to ensure maximum protection from portable TVBDs. If a city has three consecutive vacant channels (Fig. 2A, see following page), operate your wireless mic in the middle channel. This will force the TVBD to operate on a channel adjacent to an active TV broadcast, which means it will have to operate at its lower 40mW output power (Fig. 2B, see following page).

It is desired to have a portable TVBD that is approaching your production area to sense your wireless audio systems are soon as possible. The effective radiating power of mobile wireless mic transmitters is often diminished by shadowing and body absorption, especially with a body pack transmitter. Conversely, a monitoring system with a stationary transmitter using an antenna fixed in a high position provides a more stable signal. If it is operating at the maximum allowable power of 250 mW (Fig. 2C, see following page), a portable TVBD should sense it from much farther away compared to lower power, mobile wireless mics. This approach is a bit different than what was often recommended in the past: to use separate frequency ranges for wireless mics and monitoring systems. However, this technique allows a monitoring system to act as a beacon, adding a level of protection for wireless mics within the same channel.


The reduction in available spectrum plus forthcoming TVBDs provide new challenges for production professionals. However, through careful planning and adherence to best practices, even large multi-channel wireless audio systems.

By Joe Ciaudelli, consultant for the professional products industry team at Sennheiser.Used by permission.
Published in Broadcast Engineering, and on Sennheiser's website.
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Amplifier Anatomy - Part 2

By: Patrick Quilter,
Chief Technical Officer QSC Audio Products, Inc.

Power amplifiers, those “unsung heroes” of the sound system, have traditionally been the old reliable that consultants and contractors learned to count on as everything else in the sound system became more complex. While the rest of the system demanded more and more attention, the power amps were something you just didn’t have to worry about.

Although power amplifier technology has more to offer and requires more thought, there are some basic principles that every leading amplifier manufacturer follows. Understanding how equipment designers are solving amplifier problems can give you a new appreciation for this basic piece of equipment.

In Part I, we described how an amplifier works, how power supplies affect amplifier function and which amplifier classes work best for various situations. In Part II, we will go into more detail about amplifier circuitry and design.

NEGATIVE FEEDBACK AND DISTORTION In Part I, we mentioned that transistors are not inherentlyperfect magnifiers. Most of the advanced circuit techniques we described involve re-assembly of the audio waveform in various ways. If we had no way of correcting errors, the result would typically be a rather garbled and harsh reproduction.

Fortunately, we have a powerful error-reduction technique called negative feedback.

You probably associate negative feedback with criticism. Actually, this perception is not so farfetched, but in electronics there is nothing bad about “negative” feedback. This technique is basically the same process we use daily when observing one’s progress and making mid-course corrections.

Let’s use driving as an example. Imagine if you tried to steer around a corner with your eyes closed.

Even if you turned the wheel about the right amount at about the right place before you closed your eyes, the car would soon leave the road because of uncorrected small errors. In the real world, you drive with your eyes open. You turn the wheel, observer how the car is turning, and then make small corrections to maintain the desired course. This is a basic example of how we use feedback to correct for errors and produce the desired result.

We can also use feedback in amplifiers. The output of real-world circuitry is distorted: The output is higher or lower than desired. We usually don’t know just what the errors are, or we could correct for them. We can correct for unknown errors by comparing the output to the input and telling the circuit to increase or decrease the output until they match.

To describe the actual process, let’s assume we want an amplifier with a gain of 10. The actual, imperfect amplifier has a gain that varies unpredictably from 8 to 12, resulting in up to 20% of output error. We attenuate, or reduce, the actual output of the amplifier by a factor of 10. If the amplifier were error-free, this reduced feedback signal would be a perfect match to the input signal, but the actual feedback signal varies around the input signal by 20%.

We have an accurate picture of just the error because we have compensated for the desired gain by the 10-to-1 attenuation. Now, here’s the tricky part. If we magnify the apparent error by amplifying the mis-match between the input signal and the feedback signal, then combine this information with the original input signal, we can get the main amplifier to reduce its own errors automatically.

We have a crucial choice, however. The error signal can be added or subtracted from the input. If we add the error signal, it just makes the error worse. This is called positive feedback, and it turns transistors into oscillators. It turns sound systems into oscillators, also. In an amplifier it leads to wild, runaway operation. If, on the other hand, we subtract the error signal, the error is always diminished. Thus, we have used negative feedback.

In practice, we reduce error by putting much more gain than we need into the amplifier. It we increase the gain of the amplifier by a factor of 10, the gain, even with errors, will range from 80 to 120. When we “close the feedback loop,” using the 10-to-1 attenuation, the error signals always a large positive value. The amplifier quickly reduces its own output until the feedback and input signals match up.

In effect, because the amplifier has extra gain, it is in a constant state of “holding back,” which makes it easier to hit the desired target. With enough extra gain, it is ultimately the accuracy of the feedback circuit itself, not the amplifier, that determines the accuracy of the final output.

There is a limit, however. All circuitry has a slight lag between input and output. If you increase the gain of the circuit too much, it will become too sensitive, and the combination of lag and feedback will cause hunting or oscillation around the desired value. This problem, called instability, limits the amount of feedback that can be used.

The only fundamental cure is to reduce the circuit lag by using high-speed components. There has been a lot of progress in this area in the last 20 years, and today’s transistors are about 10 times faster. This progress probably explains how solid-state amplifiers have gradually eliminated the harshness that some listeners heard in early amplifiers, which used relatively slow power transistors.

Feedback can be applied around all of the cascaded elements in the amplifier. This process, called global feedback, is popular because it corrects all internal errors in one swoop. Some designers prefer a process in which they sub-divide the amplifier into several cascaded feedback loops, called local feedback. Certain forms of instability are easier to eliminate with the local feedback technique, but internal signal levels must be higher when one feedback loop feeds the next. QSC amplifiers use global feedback because it is less expensive and is easier to assure accuracy with only one set of critical feedback components to worry about.

One notable effect of high feedback is on amplifier clipping characteristics. Without feedback, transistors approach saturation gradually, giving the effect of cushioning the impact. This type of clipping is called soft clipping. With feedback, the output signal is forced to stay on track until the last possible moment, resulting in an abrupt impact, called hard clipping, which sounds more fuzzy than soft clipping. Some amplifiers feature low-feedback designs to smooth out the clipping.

Because it is so hard to reduce distortion without high feedback, the trade-off is often between clean amplifiers with harder clipping and slightly mushier amplifiers with smooth clipping. The choice depends on personal preference and on how much the amplifier will be overdriven.

In any case, it is important to avoid sticky clipping. Poor feedback circuit design can make the amplifier track its input too far, and then snap back and ring without damping. The amplifier should enter and leave clipping cleanly with no snapping or chattering.

PROTECTION CIRCUITRY The lower the impedance of the load, the greater the current drawn from the amplifier, and the greater the heat generated in the output transistors. If too many loudspeakers are connected to the amplifier, or if the ends of the loudspeaker wire touch together by accident, the load impedance goes very low, and the current flow becomes dangerously high. If the flow is not limited, the output transistors will burn out. Therefore, amplifiers need some kind of short-circuit protection. There are many ways to protect against short circuit, but the trick is that you can’t prematurely limit performance into normal loads. QSC amplifiers continuously monitor the load impedance. Loads above 2 ohms draw safe currents, and only normal voltage clipping occurs. Below 2 ohms, the maximum amplifier current is reduced if it exceeds the safe limit for more than a fraction of a second. This way, short peaks are permitted even into marginal loads, yet the amplifier is still protected against gross, sustained overloads.

Other common protective circuits include turn-on and turn-off muting, shut-down or muting in case of excessive temperature, protection against radio pickup (RFI), and dc fault protection, which shuts down the amplifier is a transistor loses control. The design of these circuits is just as important as the actual audio circuitry because they can make the difference between surviving an accident or winding up with burnt rubble.

SWITCHING AMPLIFIERS Remember those classes D, E and F that we promised we would talk about in Part I? As you have seen, heat in the output transistors is a major problem, and it is inherent in the way linear amplifiers operate. Whenever the amplifier is delivering only part of its power to the load, waste heat is created.

There is another way of converting dc power into audio power that reduces the inherent heat losses.

Because the losses occur when the output transistors are partially on, we avoid this state. We turn them fully on and send all of the dc power to the load, or we leave them fully off so that no power flows. In both cases, little or no power is wasted in the transistor.

To get the desired average power in the load, we rapidly switch the transistors on and off for the desired percentage of the time, with the time the transistors are on varying from 0% to 100%. The switching must occur much faster than the highest audio frequency if the averaging is to work correctly. This is the switching amplifier or class-D amplifier. (Classes E and F are special cases, as is class C, that apply mainly to non-audio uses.) How does on-off switching drive a loud-speaker? The magnetic field in the voice coil does not collapse instantly when the amplifier switches off. The loudspeaker continues interpolating a fraction of the waveform while the amplifier is switching.

Class-D amplifiers are only now becoming practical. The speed requirements for the switching transistors are 50 to 100 times greater than for linear audio amplifiers. The high-frequency switching causes radio interference, and many practical problems must be solved to attain the same audio fidelity that we expect with linear amplifiers. When the switching causes radio interference, and many practical problems must be solved to attain the same audio fidelity that we expect with linear amplifiers. When the switching is perfected, we can expect the heat sinks to be about one-fourth the size, reducing amplifier size and cooling requirements. When combined with the switching power supply, we will have a size and weight breakthrough comparable to the transition from tubes to solid-state. It will take a great deal of experience to overcome the reliability problems associated with complex new circuitry. This, too, should be an active area for development in the 1990’s.

MECHANICAL DESIGN We have treated amplifier design as an electronic problem, but the physical size, location and ruggedness of the parts is at least as important. You can inspect these features by eye at trade shows or on other occasions when the insides are on display.

The power transformer is the single heaviest element and must be mounted securely. Also, look for adequate clearance around it to allow for cooling and shifting in case the amplifier is dropped.

The power transformer is the single heaviest element and must be mounted securely. Also, look for adequate clearance around it to allow for cooling and shifting in case the amplifier is dropped.

The height of the chassis is another important variable. Low-profile chassis allow you to mount a lot more amplifier power in a given amount of rack space, but such a design pots much more strain on the rack ears. The rack ears should be well-connected to the chassis in such a way that overstress does not damage or loosen some other critical element, such as the faceplate. Rear support is strongly recommended. Standard-height chassis provide a greater mounting surface and allow for more internal space around components, such as the transformer.

Fan cooling creates noise in the chassis, but it dramatically reduces the size of the heatsink. Because of the increased power ratings of modern amplifiers, it is harder to find convection-cooled units. Class-G and -H amplifiers reduce heatsink requirements (as will class-D designs eventually) and may help bring back convection-cooled amplifiers. Meanwhile, high-quality, variable-speed fans can minimize the cooling noise. You should always check fan noise carefully if people will be sitting near the amplifier.

There are a number of details you should check. For example, external connectors, controls and displays should be recessed to protect them from external damage, and lockout covers are sometimes available for extra security. All connectors should be high-quality. Make sure input and output connectors are firmly mounted and will resist the strain of somebody tripping over the cabling.

Internal connectors also need to be rugged.

Sub-assemblies and circuit boards might shift as the amplifier bounces around, so excessive rigidity can be a problem. Card-edge connectors are frequently troublesome. Cable-type or shock-mounted connectors with some “give” are preferred.

Gold plating is frequently preferred on input connectors, but it might wear through after a few insertions.

The quality of the underlying plating is probably more important.

Internal connectors should also have corrosion-proof plating. Gold plating is best for small signal connections, but it can burn through at high currents. Heavy-duty connectors should use preciousmetal alloys or conductive oxide platings in which current flow actually improves the integrity of the contact.

You’ll find these principles used in QSC amplifiers and in amplifiers from other leading audio companies.

Various engineers have their own opinions on how each problem is best solved, but, in many cases, it’s basically a question of continuously refining a chosen approach. The basic principles here should help users appreciate what goes into modern amplifiers, often the unsung heroes of sound systems.

Courtesy QSC Audio. Used by permission.
Article in Sound & Video Contractor Feb. 20, 1993

David McLain | Loudspeaker Guy! | CCI SOLUTIONS
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