Frequently Asked Questions - All FAQs

The SPNConference has a powerful new echo canceller that can handle echo cancellation for multiple incoming signals from the far side. You can have multiple codecs and a phone line coming in and bridge all of them together. You will need to assign a minimum of four signal buses (or mixes) for conferencing. We suggest the following protocol. Setting up the AEC signal routing. Conferencing requires a minimum of four mixes. Two are dedicated to the AEC itself. These are called the AEC Reference mix and the AEC Signal mix. You will need to assign two mix buses in the ASPEN units for these two mixes. We recommend you use mix bus 48 for the AEC Reference and Mix bus 47 for the AEC Signal. The third is the SEND Mix (AEC Out) – you will need to assign a bus for each outbound signal. For example – if you have just a telephone line, you will need one SEND mix for the Tel. If you have one phone and two Codecs, then you will need three Send mixes, one for the telephone and one for each of the codecs. We recommend you use the mix buses 46, 45, 44 etc for these signal mixes.

This will keep all your conferencing mixes close together and separate from your local amplification mixes. Finally, you will need your LOCAL mixes – these are the signals which will be sent to your local amplifiers and may be shared with microphones. How many of these you need is dependent on the number of amplifier channels you have in use. Reference Mix – should carry ONLY the incoming signals from the far side. That would be signals from the telephone, and codecs – the inbound part of any two way communications line. DO NOT put any microphones or local line level sources (such as multimedia inputs) on this mix. AEC Signal mix – should have ONLY the local microphones. No multi-media sources, no line level inputs – microphones only. AEC Out Mix(es) – Will SEND the output of the AEC (which is the echo cancelled microphones), plus any multi-media sources to the fars side. If you want to have a bridged conference system, you will have the codec incoming going to the telephone SEND mix and the telephone going to the codec SEND mix. BE CAREFUL HERE! Make certain that you do not accidentally route the incoming telephone signal BACK on the outgoing telephone SEND mix! Or Codec to codec, etc! Local Mixes – this brings the audio from the far sides and the microphones into the room – note that we are routing the incoming phone and codec signals to the same buses as the local microphones which then feed to the amplifiers.

ASPEN latency is 1.33ms for the SPN812 and 1.43ms for the SPN1624. As you stack additional units, you add only 125 microseconds per additional link (two RU units have 250 microseconds). The 1GB backbone upon which ASPEN Net runs keeps latency extremely low. 125 microseconds is equal to a mere 6 audio samples.

ASPEN is self organizing. You won't do anything. Once your rack is built and the units interconnected with the single Cat6E cable, the ASPEN units will organize themselves into the correct Master/Slave configuration automatically. The top unit will become the master and all the subsequent units will be slaves. If you add another unit to the stack, it will be added automatically in about 90 seconds. Because the 48 mix buses are bidirectional, whatever assignment you have made for the output mixes will remain correct regardless of position n the rack.

The only effect the sequence will have in your rack will be on the control side. You can control an entire multi-unit ASPEN system through a single RS232 port but it MUST be the RS232 on the Master unit. Commands to the slaves will be preceded with a numerical designator surrounded by square brackets so the stack of ASPEN units will know which command is for which unit. Example : [2]run(3) will run macro number 3 in the second unit in the rack. So, the position of the unit will be important when writing control code.

Single RU units such as the SPN16i and SPNConference will give you 15 logic inputs for buttons or 10K linear pots and 8 logic outs for LED"S or contact closure control. The two RU units such as SPN32i, SPN1624, SPN2412 and SPN1612 will give you double that amount (30 in, 16 out). Each logic input can be used for level control (of single or multiple I/O's), to execute macros, call up presets or invoke actions. The macro control language in the ASPEN series is powerful and allows you to build sophisticated control systems with simple, push button interfaces - take a look at the RCWPB8 as an example.

Don't overlook the possibility of using the RS232 port as a control port as well! The send string command in the macro language allows you to build control commands to be sent to other serial controlled devices in your system. With a single button, you can have the ASPEN system change levels, route signals to different speakers, dial the far side connection and lower your projector (and turn it on) - all through a single macro.

It Works...WELL! We built a proven signal path that preserves good gain structure, follows known good audio practices and yet gives you virtually unlimited routing capabilities with 48 mix buses that act independently of each other.

NO LIMITS - Every function is available full time, all the time. No limitations, no over-usage warnings, if you see it, you can use it.

SAVES TIME - Fast setup. No starting from scratch every time! It’s easy to configure and includes a powerful macro control language for everything. No Gas Gauge. No compiling.

EASY - Intuitive GUI takes you from start to finish including an easy one touch macro recorder.

The gain and performance of the two antennas are the same.The ALP 600 is a PC board version of the ALP 700 and uses the same design parameters and the same number of elements. The ALP 600 uses a 4 layer PC board with strip-line matching and balun sections. We eliminated the external coax matching section that you will see on competing \"shark fins\" since we feel the coax solder joint can be broken if the antenna is really mistreated. The ALP 700 conventional antenna is not as rugged as the shark fin. We have, however, modified the ALP 700 to improve the ruggedness, by using a tapered nut for the jam nut on the antenna elements. The old elements would break at the first thread by the nut if the element was bent over. The new nut covers the threads completely and you can bend the element severely and then straighten it without breaking the element. The ALP 600 shark fin is still much more rugged, though. Aside from price, the only time not to use the shark fin, is if you are working in high winds or have the antenna mounted on a moving car or van.

The shark fin looks like a miniature sail. The shark fin is more expensive because the 1/8" thick, four layer board costs us more than all the metal and machining that goes into the ALP 700. I still find it hard to believe

Active antennas sound good to customers but they have many shortcomings:  

1. If the antenna cable is less than 25', an antenna amplifier is not necessary and is actually detrimental to the operation of system.  
2. The antenna amplifier must be relatively wide band since it must handle all the frequencies to which the receiver may tune. If the amplifier is wide band, it will pick up many interfering frequencies.  
3. It is difficult to design a high powered amplifier that is also low noise. The high powered amplifier is necessary to handle all the garbage that can come into a wide band amplifier.

To put it into a few words, antenna amplifiers are never as good as the front end of a well designed receiver. They are a necessary evil and should only be used when necessary. They are necessary under the following conditions:  

1. When the antenna has to be more than 35' feet away from the receiver and the signal to the antenna is also weak such as antenna to performer distances of 100' or more. (If the signal to the antenna is strong then you will have enough signal to compensate for the cable losses.) Keep in mind that the antenna amplifier is NOT solving the weak signal to the antenna problem but just compensating for the cable losses after the amplifier. Again, the amplifier can't compensate for losses before the signal gets to the amplifier; only losses after the amplifier.  
2. If there is a passive splitter after the antenna that introduces loss. This is equivalent to cable loss and the same rule applies; put the amplifier ahead of the splitter. A 2 way splitter has 3 dB of loss, a 4 way has 6 dB of loss and so forth.  
3. When the receiver is a poor design with a noisy front end and the antenna amplifier can boost the signal enough to overcome the receiver self noise.  

Our "active" antenna setup consists of a passive antenna plus an external amplifier, the UHF50. That way you can use the amplifier only when necessary. We use a high power amplifier that is pretty quiet and we also put in a pre-filter that is two blocks wide (50 MHz) in front of the amplifier. It isn't as good a filter as those in our receivers, but it is better than the universal wide open designs. In addition, we make the gain of the amplifier adjustable so that you can match the gain to the losses in the cable or splitter system.  

What is confusing about the whole antenna amplifier issue is that cable loss degrades the sensitivity of the receiver but more gain doesn't improve it. In fact the additional gain leads to overload and intermodulation problems. This is Mother Nature saying not only can you not win, it's hard to break even. Another way to think about it is that our receivers are already about as sensitive as can be. If an additional amplifier could improve sensitivity, then we would have built it in. 

You can tilt the antennas so that they are at 90 degree angles to one another. That is to say, bend one 45 degrees to the left and the other 45 degrees to the right. The tilted antennas are a reasonable way to operate and the best way if the antennas are fairly close together since they couple together much less than if they are both pointed in the same direction (parallel).

The antenna diversity used in our receivers does not select one antenna or the other; it sums the two antennas together and corrects the phase of one antenna so that the antenna signals do not cancel each other out as they might do if they were 180 degrees out of phase. So it does not make too much difference which way the antennas point since the receiver will correct the phase.

Additionally, in any usual environment, the signals coming to the receiver from the transmitter are not in any well defined phase relationship or direction. The signals are reflected from cars, the ground, metal studs, wire in walls, camera equipment and even people, so that the signal that gets to the receiver is pretty well scrambled and impossible to predict. The problem with reception occurs when all the signals from all the reflectors get to the antenna and cancel out. If you use two antennas, then the signals probably will not cancel out at both antennas simultaneously. There is a new problem, though, if you simply add the two signals together. When the signals at each antenna are equal and exactly out of phase they cancel out at the receiver. The phase diversity system that we use on our small receivers detects this condition and simply inverts the phase of one of the antennas. Now the antennas add the signals together for a 3 dB pickup in power. For a good explanation of this, that is more comprehensive than what I can do here, go to this link to our web site.

Dropouts and Noise-ups 1

It is part of our wireless guide. In fact you might want to down load the entire wireless guide because it is pretty good and pretty neutral in its treatment of wireless microphones.

It is surprisingly hard to do. The big problem is the battery terminal voltage is heavily influenced by how the battery has been discharged in the past. If it has been run down slowly with power gradually pulled out say over a 24 hour period, the relationship between remaining battery capacity and terminal voltage is fairly well defined. If the battery has been discharged heavily, say by a Lectro UM250, the relationship is not so clear. Basically the battery bounces back to a high voltage and can look like it is still pretty fresh. Under either a light or heavy load it will run down quite rapidly. The problem is, the battery tester has no way of knowing the past history of that particular battery. 

As a demonstration, if a fresh 9 Volt is accidentally shorted out with a piece of metal for 1 minute, you will get very odd results. The battery will get moderately warm. If the battery then sits unused for 8 hours, the terminal voltage will then measure pretty close to a new battery even under a brief load, but it will only run a transmitter for 5 minutes or so and then die almost instantly. In fact, if you then let it sit for a while again, the voltage will come back up again and die again in a transmitter in just a few minutes. I agree that this is an extreme case but it does demonstrate the problem of prior history.

Even when you know the history you can get bit. Recently we have been running battery tests on different brand batteries and we have found that some alkaline batteries tend to die very rapidly at the end of their life but other brands continue to run with lots of warning before they finally die. What's worse, different batteries from the same manufacturer may act differently. The reason we did the test is that we were getting complaints that the UCR201 was not giving sufficient warning with batteries made by XXXX brand, a major manufacturer. We tested the batteries and found that Evereadys gave 34 minutes of operation after the battery indicator started flashing its warning and the XXXX brand were giving about 3 (!) minutes of warning. We found this to be consistent with XXXX from 3 different parts of the country. Since so many of our dealers sell XXXX, we aren't sure what to do other than recommend Eveready as the standard. The XXXX brand is a perfectly good battery but it has a slightly different chemistry that is optimised for things other than high current drain. 

The safest answer is that a low voltage reading will always indicate that a battery is weak but a normal or high reading may not necessarily mean that a battery is good. This is why so many pros that absolutely depend on their equipment, put in a fresh battery at the beginning of a job or whenever there is the slightest doubt that the battery will "last long enough"

This information was gathered for a question on the RAMPS newsgroup for a UM200 transmitter but should be proportionally the same for other transmitters.

(See also FAQ #009-WIRELESS)

For newer tests on the iPower LiPoly rechargeable 9 Volt See (FAQ #086-WIRELESS)

Here's some more battery information as I promised a few days ago. It took a while to run all the batteries down. Here is what we did: we used the same transmitter, a Lectro UM200 for all the testing. This is a 100 mW UHF belt pack transmitter. This particular unit pulled 75 mA. We ran four different kinds of batteries to a final voltage of both 7.0 and 6.6 Volts. 7.0 Volts is where the LED is pretty dim and where two of our receivers with battery readouts start indicating low battery and 6.6 Volts is the very low battery indication. The transmitter is getting close to completely dying at 6.6 Volts but will usually run to 6.4 Volts or less. The LED goes out totally at 6.8 Volts. I'd put all this in a table but I don't think it would survive the news readers' formatting. So I'll list the type of battery and then the very dim LED point (7.0 Volts) and then the maximum use (6.6 Volts). Your mileage may vary.

• Ultralife Lithium 16.0 hours and 17.2 hours
• Duracell Ultra Alkaline 6.5 hours and 8.25 hours
• Eveready Alkaline 4.75 hours and 6.75 hours
• Varta NiMh rechargeable 2.5 hours and 2.5 (!) hours
• Varta after 2 months of sitting around is the same as above, 2.5 hours.

Here's my conclusions: Assuming that a sound mixer with good common sense would toss a battery when the LED is very dim (or sooner) and using an standard alkaline as a reference, you'll get 3 times the life with a lithium, about 40% more life with an Ultra alkaline and about one half the life with a top quality nickel-metal hydride (NiMh) battery. (Though the Varta NiMh claims only 150 mAh, they start out at more than 180 mAh.) Also, the NiMh batteries don't self discharge as quickly as NiCad batteries since the battery after sitting for two months was still at close to full capacity. 

Disclaimer section: These were fresh, new batteries at room temperature. This was just one test, performed on just one transmitter.

Anti-disclaimer section: Most brands of alkaline batteries are about the same, alkalines and lithiums have a long shelf life, and our transmitters are pretty consistent. We have found the Eveready batteries to give the longest life for a standard alkaline battery. In any case, the ratios of battery life should be good numbers. You guys know what kind of battery life you are getting now, and the ratios should be informative.

Here's the total and long winded story on Lectro low end frequency response. Once upon a time, all the transmitters were set up to be flat (1 dB down)to 50 Hz. Some UM195 users were having problems with low frequency rumble driving the bass compander and causing rumble modulated breathing in the system. We did 3 different fixes in the system, one of which was to roll off the input at 70 to 90 Hz. We picked this number cause that's what Vega was using and they were the Big Dogs at that time and we figured, correctly, that they knew what they were doing. This improved the operation of the systems and all seemed well except that Jerry Bruck (the Schoeps importer) wanted response down to 50 Hz. So for a while we did a Jerry Bruck modification that was basically back to the original input reponse. Jerry knew what he was doing and could handle the room rumble and wind noise in other ways. Then a lazy engineer (me) decided this was too much paperwork to track all his orders and redesigned the UM series transmitters to have a variable rolloff. That way Jerry could have his cake and we didn't have to build bake a special. You can hear the rolloff on voice if the rolloff is set to maximum (185 Hz). If you set it for 100 Hz or lower, it leaves voice alone.

The UH195 and UH200 plug on series were never rolled off since most pro microphones have a low end roll off anyway or have switchable filters. Or they have extended bass reponse and the user has selected the mic for that bass reponse. In any case, we left the UH's alone and they are flat to below 50 Hz.

Here is a reply given to this question on the RAMPS group:   There was a query about whether to send mic level or line level to a camera that always had a mic preamp in the signal chain and used an attenuator to convert line level down to mic level anyway. The answers ranged from it didn't make much difference to the line level would pick up less interference. Both are reasonable answers. In general, it is better to keep signal levels high from the source and if necessary attenuate them at the input to the "load" (in the case described, the camera audio input). If the signal is attenuated to mic level at the source, then any noise picked up in the cable enters the camera at full noise level. If the signal is at line level at the source and then attenuated at the camera, the noise picked up in the cable is attenuated also. This could improve your noise rejection by 30 dB or more.

A case in point: We made a law enforcement wireless system a few years ago that went in the trunk of a police car. Our system provided audio to their VHS video recorder that was tied to a small video camera mounted behind the windshield. The officer wore a transmitter. The distributor buying the wireless systems from us specified mic level to feed the recorders since the ALC (automatic level control) was for that input. In some installations there was severe alternator whine because of ground loops and other clicks as equipment turned on and off. The grounds to the system included the antenna ground to the car chassis at the car whip antenna, the audio ground to the recorder, the power supply ground to the receiver, and a digital ground from our squelch circuit back to their readout box. Those were just the grounds to our receiver and didn't count the grounds to the recorder, camera, etc. The installers in different states and cities all had their own way of hooking things up and varied from good to horrible. An isolation transformer in the audio line usually fixed the problem but was too much money. We finally convinced the distributor to use an attenuator at the input to the recorder and let us send a line level signal. This reduced the ground loop noise by 30 or 40 dB and "solved" the problem. The distances involved were only a few feet and it wasn't really a cable pickup problem. Later on we convinced the distributor that the ALC wasn't a good idea anyway because it confused juries as to what was going on and made it hard to hear. In addition, the ALC was really upset by gunshots, as were the officers of course. Sending line level out of our receiver to the line level in on the recorder made for a bullet proof installation, so to speak.  

So the rule of thumb is high levels at the output and attenuate as necessary at the input, even though it won't make a difference most of the time.  

Here is an adapter for the UCR100 to attenuate its output down to balanced mic level.

This reply was posted to this question on the RAMPS group:

Most wireless systems, even some "pro" systems do not have a limiter-compressor in the transmitter. This forces you to do exactly what you are describing, which is to attenuate the mic input to prevent the occasional overload. All the Lectro transmitters for the last 15 years have a shunt FET limiter before the input preamp. The nice thing about the shunt limiter is that it is out of the audio circuit until a potential overload comes along, then the excess signal is shunted away. The limiter has a range of 25 to 30 dB. At usual gain settings, the transmitter won't overload until after the typical electret lavaliere microphone is already clipping.  

Interestingly, the Vega microphones from years ago had a very effective limiter using an LED/LDR (Light Dependent Resistor). Vega referred to it as a "soft compressor" and it was. Though it wasn't effective when the transmitter gain was set low, for real world use, it was very nice sounding and, in my opinion, one reason Vega was the number one pro wireless. More interestingly is the fact that most current wireless mics have taken a giant step backwards by leaving limiter-compressors out of the bag of design tricks. Check the specs on the data sheets to see if there is an input limiter-compressor. Chances are there isn't one. The COMPANDER used in all current wireless mic systems has nothing to do with the input limiter by the way. The input limiter is in addition to the compander and additionally increases the usable dynamic range.

Here's the answer I posted on RAMPS about 9 Volt battery life:

Here's the last of the 9 Volt battery tests. This is a similar test to what we did in a previous post but with a high power transmitter. (See also FAQ #005-WIRELESS) For newer tests on the iPower LiPoly rechargeable 9 Volt (See FAQ #086-WIRELESS)   Here is what we did this time: we used a 250mW transmitter, a Lectro UM250 in the testing. This is a 250 mW UHF belt pack transmitter that eats 9 Volts like they were potato chips. This particular unit pulled 105 mA. We ran three different kinds of batteries to a final voltage of both 7.0 and 6.6 Volts. 7.0 Volts is where the LED is pretty dim and where two of our receivers with battery readouts start indicating low battery and 6.6 Volts is the very low battery indication. The transmitter is getting close to completely dying at 6.6 Volts but will usually run to 6.4 Volts or less. The LED goes out totally at 6.8 Volts. I'll list the type of battery and then the very dim LED point (7.0 Volts) and then the maximum use (6.6 Volts). Your mileage may vary.

• Ultralife Lithium 5.5 hours and 6.6 hours
• Duracell Ultra Alkaline 2.6 hours and 2.8 hours
• Eveready Alkaline 1.8 hours and 3.2 hours (!)
• Varta NiMh rechargeable 1.0 hours and 1.25 hours

These are interesting results. If you saw the earlier post with a similar table, you will notice that the Ultra alkaline has the same 50% advantage to 7 Volts but when run to 6.6 Volts, has instead, a 13% LOSS. This is not the same as for a 100 mW transmitter. There the Ultra was 50% ahead at either end point voltage. The Ultra fell like a rock when the voltage got to 6.6 Volts. In fairness to the battery manufacturer, these 1/4 watt units are very hard on batteries.   Same disclaimer as before: These were fresh, new batteries at room temperature. This was one test, performed on one transmitter.   And same anti-disclaimer: Most brands of alkaline batteries are about the same, alkalines and lithiums have a long shelf life, and our transmitters are pretty consistent. In any case, the ratios of battery life should be good numbers. You guys and gals know what kind of battery life you are getting now, and the ratios should be informative. We have found Eveready to be the most consistent general purpose alkaline.

On to other projects. I've seen enough battery strip charts for a while.

The 185 series has diodes to the power supply to protect the output from 48 Volt phantom power or other high DC voltages on the output jack. If the voltage on the output line exceeds 12 Volts, the diodes turn on and shunt the excess voltage. Since the diodes are now conducting, they also shunt (kill) the audio. The solution is to remove the 48 Volts. Some years after the first 185 design, we ran into mixing boards (poorly designed IMHO), where the 48 Volt was on all the mics or none of the mics. We redesigned the protection to either include non polar capacitors or shunt resistors to ground as well as the diodes. The newer designs prevent the audio from being shunted. This is what you will find on all receivers after the 185 series. It is possible to change the output on your 185, but I recommend just removing the 48 Phantom power when necessary.

The +4 dBm designation indicates that a full modulation signal will just reach +4dBm rms max. The line level input level at the transmitter can be attenuated by different factors depending on the pins selected (pin 4 or 5) and the resistor to ground (pin 3 or 4). The gain pot also can vary the applied gain also. To sum up, there is no fixed relationship between the input level to the transmitter and the output level from the receiver other than the +4 dBm setting is the maximum output level you can see. If the transmitter is set up so that you never activate the limiter, then you will never see the +4dBm. If you have 20 dB of headroom between your maximum input level and the transmiter limiter, then you will never see more than -16 dBm. My guess is that you are being pretty conservative on your input levels and could run them substantially higher; this will bring your output levels closer to the max indication and also improve the signal to noise ratios and noise ups.

There are three possibilities: One, batteries do not like to be cold. At low temperatures (32F) battery life can be one third of that at room temperature (72F). Two, some brands of batteries will not deliver the high currents used in our receivers and 100 mW transmitters. We use Eveready as our standard battery. Three, our units will operate to very low battery voltages and you may not be running the battery down far enough. Here's a reply to a UCR201 user that was replacing batteries every 2 hours or so.

At room temperature, the 201 should give you +4 hours of operation. Try an experiment when you have some free time with a fresh battery. Simply run the unit in the battery voltage display mode and see how long it takes to pull the battery down to 6.5 Volts. The system will operate perfectly to below 6.5 Volts since all internal voltages come from several switching power supplies. We have found a lot of variation in XXXX batteries and some batches will not provide the high currents the 201 draws. We have never found problems with Eveready 9 Volts. The XXXX batteries acted so weird I suspected they might be counterfeit. This was on several batches of XXXXXs from different parts of the country. Further testing found that other XXXXXs were almost equivalent to the Eveready's. We remain puzzled. My advice is that if you are getting short life, try the Eveready's as a standard before deciding the unit is defective.

If you are having short battery life in transmitters due to cold weather, keep the transmitter warm as long as possible before you have to use it. Belt pack transmitters can be also be put under the coat so as to be next to the nice warm human being.Alkaline batteries, though very good at room temperature, cannot deliver much current at lower temperatures. Battery life can be as little as one third normal on a cold day and even less if they cold soak for any length of time. Life can be as little as just a few minutes at -20 F.

If you must use disposable batteries (non rechargeable) then lithium batteries are the only good choice. They have shorter life at low temperatures but are still much better than alkalines.

In the AA battery size, NiMh batteries are a good cold weather choice. At low temperatures they have almost as much life as at room temperature and are rechargeable to boot. We recommend the Eveready NiMh batteries and 15 minute charger that we provide with the SM, SMD and SMQ transmitters. One precaution is that the batteries cannot be recharged if they are cold. They can be used without any problem but must be at about room temperature to be recharged. (See FAQ #087-WIRELESS)

In the 9 Volt battery size, NiMh batteries perform as well cold as they do at room temperature but they don't have much battery life (capacity) cold or warm. At one time they were the only choice for very low temperatures but LiPoly rechargeable batteries are now available that have more capacity than alkaline batteries and perform very well at low temperature. They are currently sold under the iPower brand and are available on the internet, from some dealers and from Lectrosonics.  (Also see FAQ #086-WIRELESS)

We make an adapter that converts the UH400 XLR input into a TA5M 5 pin equivalent for 2 or 3 wire lavaliere use. The model number is a MCA5X. You will need to set the UH400 for 5 Volt phantom power but it does have Zener protection if you forget and leave the UH400 on the 48 Volt setting. It's in this catalog: 

You can also do the following, but you MUST have the UH400 set for 5 Volts, not 18 Volts or 48 Volts:
For a 2 wire lavaliere, pin 1 of the XLR is ground and pin 2 is bias (audio). Pin 3 not used for a lavaliere. 

If you have a three wire mic like the Cos-11 it is a little more complicated but not bad. Hook shield to pin 1, a 2 k to 3 k resistor between pin 1 and the Cos-11 white wire (source load) and hook the remaining black wire to pin 2 of the XLR. This will give you a low distortion hookup comparable in gain to the UM series of transmitters. We strongly recommend putting a small 6.8 to 9 Volt Zener diode across pin 1 and pin 2 for voltage protection in case you misset the voltage. The cathode goes to pin 2 and the anode to pin 1.

The MCA5X is certainly an easier solution.

The UH plug-on has the same audio circuity as the UM belt pack with the exception of the variable bass roll-off pot found on the UM. The UH plug-on roll-off is fixed at 70 Hz, which is the best all around compromise, IMHO.

You can insert a "MiniCircuits" brand two way splitter into any of your RF distribution system's antenna outputs with a small loss of 3 dB. This loss will cut your range to 80% of normal and will probably be undetectable in the real world. You will need a total of two 2 way splitters, one each for diversity outputs from the distribution amp. For a diversity Quadbox, for instance, this will give both antennas 5 total outputs, 3 normal and 2 that are 3 dB down. You could change the attenuator that we use at the antenna outputs and reduce the attenuation by 3 dB to make up for the additional split, but I don't think it is worth the trouble and the Quadbox would then be non-standard. You can get the necessary short cables and two way splitters from us. The splitters are also available at the same price directly from MiniCircuits (a top notch company).

Additional Info:

Splitter/Combiners

ARG Series COAX Cables

Usually it won't. Our receivers' front ends are very quiet and within a few dB of the theoretical limits for noise. The range limitations are usually due to other interfering noise sources in the environment. Additional gain before the receiver will not increase range but will lead to increased RF intermodulation and RF overload. The only time you want gain before a well designed receiver is to neutralize cable and/or splitter losses. In this case, the gain must go before the cable or splitter loss and it must be a high quality amplifier such as our IFM50. Putting the gain after the loss is too late since the signal to noise ratio is now poor and gain won't improve it. An analogy for RF is the same as for music; lots of amplification can't overcome poor source material.

This was a general question from the RAMPS group about various wireless transmitters generating low frequency noises when struck.

In general, there is mechanical coupling from the case into the inductors in the main oscillator in the transmitter. A thump on the case moves or bends the inductor, changes the inductance value by a tiny amount and changes the frequency of oscillation. Since a changing frequency is just FM, the FM receiver picks it up as a low frequency thump. There are various ways of reducing the mechanical sensitivity. Most involve very rigid coil assemblies such as inductors wound on ceramic forms. In our case, we use solid quarter wave ceramic resonators. 

The cutest trick I've seen, was a (brand) unit that used a miniature Teflon insulated coaxial line as a resonator. They wound the coax stripped off the outer insulation in a tight cylindrical coil with about 6 turns. The entire shielded coax coil was then soldered on the outside into a solid mass. This made a nice rigid assembly with the center conductor acting as the inductive element since a short coax line with one end shorted looks like an inductor.

The other way to generate a thump is to use a capacitor in the audio circuity that is sensitive to mechanical stress. The wrong kind of ceramic capacitor with DC voltage on it can really generate a lot of voltage when stressed. NPO ceramic capacitor types are as good as most film caps or tantalums but X5R types are bad and Y5Z are horrible. NPO's have the least capacity for a given size and the other types have 5 to 50 times more capacity in a given size and that's why they exist. I tried 50 Volt Y5Z type capacitors in the design of the 48 Volt phantom supply for the UH200C. You could get about as much audio talking into the transmitter PC board as you could using a microphone. Fortunately some small 50 Volt tantalums came on the market that would fit in the same space and saved my bacon. I knew the problem existed, but the severity surprised me.

My advice is to whack the case of a transmitter with both your finger and with a pencil sized object. If you know how a transmitter is going to react to mechanical shock, you can prepare for it.

On the subject of mechanical stress and audio, try the same thing with your electret mic cables. Some are much worse than others. If you tap the cable close to the mic (6") you will get mechanical noise transmitted directly to the mic element. In the middle of the cable, it is due to flexing of the mic cable. Phantom powered are sensitive to this since there is DC voltage on the cable and flexing the cable changes the dimensions and the capacitance of the cable. The pro mic manufactures have taken this into consideration in the choice of cable.

The 100 series does not have a pilot tone in the transmitter so this won't work because the IFB receiver requires a 30 kHz pilot tone to be present before producing audio (unsquelching). If you are trying to run 100 series transmitters on the talent, a 100 receiver feeding a recorder and then monitoring on the IFB receivers simultaneously, I'm afraid you are out of luck. 

The newer LM and UM transmitters can run in the IFB mode and the newer units will have that capability marked on the case. You could run the transmitter in IFB mode and then IFB receivers and 100 receivers would both work. However the sound quality in the 100 receivers would not very good.

The pre-emphasis of the 100 series and the IFB series is about the same. The problem comes from the fact that the 100 series has a dual band compander that treats the bass and treble part of the signal separately. If the audio signal is balanced above and below 1 kHz, the single band compandor in the IFB and the dual band compandor in the 100 will track pretty well. If the bass and treble are not about equal after pre-emphasis, then the weak band will be reduced even lower compared to the stronger band.

The power supply for the electret mic bias turns off immediately rather than delaying till the receiver mutes. It is a design error. All LMs shipped after Jan 2005 have been modified. We have a clean fix for earlier units and will modify your LM at no charge. Contact our service department for help.

This is a long posting made to the RAMPS news group. There is a clearer explanation here: see FAQ 105

Some weeks ago we made measurements of a simulated bag system to see what having a 100 mW transmitter 12 inches (30 cm) away from receivers would do to the receivers' sensitivity. If you read the previous post, you will remember I was somewhat surprised at the performance of the UCR210 receiver in this test. I thought the older, helical resonator front end, single frequency UCR195 would carry the day. In fact, the UCR210 performed as well and in some areas was a bit better. Some users on the group were wondering how the UCR201 would perform in a bag even though this was never our intended use for the 201. This time the numbers are more in line with what I would guess, since the 201 is definitely weaker in this test. The first set of numbers are for the UCR210 and the second set are for the UCR201. I'm not going to try to put them on the same line since I think various news readers will mangle the formatting. To see how the test was run, I included all the text of the previous test below the new numbers. Again, I'll put frequency separation in MHz, then dB of desensing, then resulting percentage of range with a percent and then a comma as a line separator.

Again, the explanation of the test setup and procedure are in the text after this posting. 

(UCR210 from the previous test)
0.5M 20dB 10%,
1.0M 14dB 20%,
1.5M 12dB 25%,
3.0M 10dB 32%,
4.0M 9.8dB 33%
6.0M 3.6dB 66%,
10.M 2.3dB 77%,
20.M 1.8dB 81%


(UCR201 from a new test)
0.5M 89dB 0%,
1.0M 77dB 0%,
1.5M 18dB 12%,
3.0M 23dB 7%,
4.0M 17dB 14%,
6.0M 11dB 28%,
10.M 6dB 50%,
20.M 3dB 71%,
30.M 2dB 80%

As can be seen from comparing the numbers, the UCR201 needs twice the frequency separation before the ranges are comparable. The 3 MHz number looks funny but that's what we measured. The 50% of normal range is reached by the UCR210 at 5 MHz of frequency difference while the UCR201 needs 10 MHz of separation. I would recommend separation of at least one of our blocks (25 MHz) between the 201 receivers and transmitters in the bag. On the other hand, the UCR210 can operate inside the same block with a little care. The difference is due primarily to the tracking front end in the 210 and secondarily due to the higher power level of the first RF transistor in the front end of the 210. Once the signal is past the front end, both receivers are essentially the same.

We ran the same tests with reduced transmitter power (30 mW equivalent) and found a one to one reduction in desensing. The 5 dB reduction in power (100 mW to 30 mW) reduced the desensing by 5 dB. So you could take the above numbers and reduce the desensing by 5 dB which would be equivalent to a 30 mW system at 12 inches or a 100 mW system at 18 inches (45 cm). Note that the levels drop faster than the square of the distance at very close distances. A 10 mW transmitter would improve the desensing numbers by 10 dB. The reason I mention 10 and 30 mW is those are the power levels of some other popular transmitters on the market. I don't think the matching receivers will be quite as tough as the UCR201 but with the lower transmitter power, the upshot is the 201 numbers probably will describe what other systems would do. The one exception, to the best of my knowledge, is the small Sennheiser receiver that does not have an RF amplifier but takes the antenna input directly to the mixer. It has a very good third order intercept but does not have the sensitivity of other Sennheiser receivers. It would be interesting to measure it and see if the resistance to desensing would counter balance the lack of an RF stage. I'm sure it would. A good universal rule of thumb is that 10 MHz of separation with reasonable quality systems will give you 50% of your range. You will need at least 5 MHz even with the UCR210.

Larry Fisher
Lectrosonics



[[[ Below is the text of the original measurements and a complete description of the setup and parameters. The new measurements above were made in the same manner. We tried to make sure we were comparing apples to apples.]]]


We did some interesting RF measurements on a simulated two way bag system to see how much the bag transmitters would affect the bag receivers' sensitivity. A two way bag system will at least consist of multiple receivers to receive audio signals from the talent, a portable mixer to mix the audio and one or more transmitters to retransmit mixed audio to the video cameras. The immediate question is "If the receivers and transmitters are on different frequencies why should the transmitter reduce the sensitivity of the receiver?" One obvious answer is that the RF front end of the receiver is not a perfect filter and can let strong, nearby frequencies pass through and overload the first amplifier. In addition, transmitters do not produce a single sharp frequency but have some noise 5 Mhz or more from the carrier. The levels are very low but bag systems have antennas that are very close together. In the same way, the local oscillator in the receiver produces some noise many MHz away from the desired frequency and acts the same as having noise in the transmitter. Instead of trying to calculate all this stuff it is simpler to just measure a simulated system.

We put a transmitter 12" (30cm) away from an antenna mounted on a power meter and measured an average signal of -5dBm (.5mW) from a transmitter with 20 dBm output (100 mW). This is a very strong signal to bleed into a receiver but will be very typical of a bag system with 12" of separation. We used this level for the interfering transmitter for all the sensitivity tests. We then checked the receiver sensitivity with the transmitter off and then on and measured the reduction in receiver sensitivity for different frequency offsets between the transmitter and receiver. So to simulate a bag system where the talent's transmitter is on 540 MHz and the bag is re-transmitting mixed audio to the camera on 550 MHz, we would inject a 550 MHz signal at -5dBm into a UCR210 receiver set at 540 MHz and see how much that affected the receiver's ability to pick up the desired 540 MHz signal. We attenuated a block 21 UM200C transmitter set at 550 MHz down to -5 dBm and combined it with a weak 540 MHz signal from a signal generator, set the receiver to 540 MHz and checked the sensitivity with the transmitter off and then on. With the transmitter off, the receiver had a normal sensitivity of -107 dBm for 30 dB SINAD.(Same as "signal to noise ratio" at these values) With the transmitter on, the sensitivity fell to -104.7 dBm for a decrease in sensitivity of 2.3 dB. Or the receiver was desensed by 2.3 dB. This means that in the real world, a 10 MHz offset in the two systems' frequencies with the antennas 12" apart, the usable range from the talent to the bag would have been reduced by 23%. This is pretty small and surprised me. I thought it would be much worse. (There is no reduction in the distance from the bag to the camera since the receiver at the camera is not near a transmitter.) At 0.5 MHz separation with the talent transmitter and bag receiver still at 540 MHz and the bag transmitter at 540.500 MHz, the desensing was much worse at 20 dB. This would reduce the talent to bag range to 10% of normal and is a good reason to never operate with only 0.5 MHz frequency separation. Here's some more values for a UM200 UCR210 system and I hope the news readers don't totally mangle the formatting. I'll put frequency, then dB of desensing, then resulting range with a percent and then a comma as a line separator. Something should get through.

0.5M 20dB 10%,
1.0M 14dB 20%,
1.5M 12dB 25%,
3.0M 10dB 32%,
4.0M 9.8dB 33%
6.0M 3.6dB 66%,
10.M 2.3dB 77%,
20.M 1.8dB 81%

The above numbers are for a system with a tracking front end with relatively high power RF amplifiers both of which help. The relatively high power transmitter of course hurts. If you didn't need much range, bag to camera, you could attenuate the transmitter. The better solution would be to move the transmitter away from the receiver. On the other hand, 5 Mhz of frequency separation would leave you with 50% of your range and that would still be a gracious plenty in 99% of the cases. This is all a good reason to have the transmitters and receivers in a bag system separated by a frequency block (25 MHz). But if you have to operate in one TV channel, 6 MHz, off set the frequencies as much as reasonable.

I had postulated before we tried this experiment that the helical resonators in the fixed frequency UCR195 receiver might work better in a 2 way bag system than the higher power RF amps and wider tracking filters in the UCR210. Here's the numbers for a UCR195D at 536.250 MHz and a block 21 UM200C. The measurements were made the same as above,frequency, then dB of desensing, then resulting range and a comma.

0.5M 30dB 3%,
1.0M 28dB 4%,
1.5M 26dB 5%,
3.0M 25dB 8%,
4.0M 7.8dB 41% 
6.0M 4.4dB 60%,
10.M 3.1dB 70%,
20.M 2.8dB 72%

The numbers are much worse for separations less than 4 MHz. Basically the dual gate GasFet is wiped out in the UCR195D until the helical resonators can start filtering at 4 MHz off frequency. Note the rapid change in the numbers between 3 and 4 MHz. This is where the helicals really begin to filter. Overall, the higher power RF amps in the UCR210 carry the day, though the helicals are roughly equivalent at 4 MHz and higher. If I can say something in self defense here, that is why the newer Lectros are rightly accused of eating batteries; they burn it in the front ends.

Having been major wrong about the UCR195 and the helicals, I thought we'd test out my hypothesis that the "rock" transmitters (crystal controlled UM195) would have less noise off frequency than the synthesized transmitters such as the UM200. Here we used a UM195 on 536.250 and varied the frequency of a block 21 UCR210 to check sensitivity versus separation. Note the frequency offsets are a little different.

1.5M 28dB 4%,
2.0M 25dB 8%,
4.0M 17dB 14%,
6.0M 7.0dB 45%,
10.M 4.2dB 62%

Notice that the range is much more affected by the UM195 transmitter. The receiver is the same UCR210 used in the first measurements yet the receiver is now much more affected. The only explanation is that the UM195 is noisier off frequency than the UM200C which is not what I would have expected. By the way, this "noise" would only show up under these bag conditions. Twelve inch separations from transmitters to receivers are not usual. We will measure some different UM195's but at the moment the synthesized radios seem to perform easily as well as the fixed frequency units in bag system conditions. Now where's the crow seasoning.


Larry Fisher
Lectrosonics

RAMPS is an acronym for "rec.arts.movies.production.sound". This is a news group for sound mixers in the film and video industry as well as others that have similar interests in field recording of sound. There are a number of very helpful pros that support the group and there is a lot for anyone to learn. You will need a news reader to access the group. You can use the link below to access RAMPS using Google Groups. Go to RAMPS group

If you are using alkaline batteries, you will have very short battery life. The MM400 and the SM transmitters use a single AA battery to reduce size and weight. Since they are 100 mW transmitters, the load on the single battery is unusually high, about 450 mAh. Alkaline batteries are not designed for that high a current and will last less than 2 hours at room temperature. In cold weather, the run time is much less and can be on the order of minutes.

Lithium batteries can provide higher currents and will run slightly more than 6 hours at room temperature. If run a few hours at a time, they can provide a maximum of 8 hours of life. However at freezing temperatures they have shorter run times. (See FAQ #087-WIRELESS)

NiMh batteries are now available with capacity ratings of 2500 mAh and will operate the MM400 or SM transmitters for slightly over 4 hours. The above times were made with very fresh batteries; as the ads say "Your mileage may vary". 

With 15 minute chargers available from several large and reliable battery manufacturers, they are a very viable alternative. Some of these chargers (RayOVac and Eveready) will operate from a 12 VDC source. Finally, the NiMh batteries are not nearly as sensitive to low temperatures as other battery types.

This was an email question that I re-posted on the R.A.M.P.S news group since we see quite a few units that are destroyed by water immersion each year. Below is the reply which also talks about salt water immersion. (VERY BAD. Get the battery out immediately)

Hi David,
Some of the following doesn't apply to you now, since it has been a while since the unit fell in the water. What you have done so far is OK.

Get the battery out of the unit as quickly as possible. Turning the unit off is not enough. Wash the unit with clean or distilled water. If it fell in sea water, wash it with any water that has less salt in it than the sea water.(If there is nothing better available, you can even use Diet Coke or a dry martini, shaken not stirred). Vodka or other alcohol that doesn't have lots of additives in it can be used in a pinch also. In fact, an alcohol final rinse is a good bet anyway, both to promote faster drying and to clean off any dirt or oils that may have been in the water. Tilt the boards so that the alcohol runs off and doesn't puddle around the tiny adjustment pots. After the final rinse, shake the boards to remove the alcohol or water. Then warm the boards to drive off the residual water left from the water or alcohol final rinse. Hair dryers, light bulbs, warm oven, engine block, sunlight, heater vents, etc. As far as the maximum temperature, as long as you can firmly touch the components and not quickly feel pain (140 F) you are OK. After several hours or more and the unit is bone dry, put in a battery and try the system.

In fresh water with the battery in the unit, you have a few minutes before serious damage results. In salt water, with the battery in the unit, it is a matter of seconds. Get the battery out of the unit as quickly as possible. If it fell in sea water, don't bother opening the unit to wash it. Plunge the entire unit with the battery door open into water or alcohol. Or pour water or alcohol into the unit and slosh it around. Then do it a second time with a fresh batch of water or alcohol. Then you can open the unit and do a third rinse. Inspect the circuit traces for corrosion. If there is any corrosion, try to rinse it off and/or brush the corrosion away using a stiff brush moistened in alcohol such as a cut down acid brush or a toothbrush. If it has corrosion, it may need to come back to the factory. We usually end up replacing the circuit boards or exchanging the unit completely but there is no harm in cleaning it, drying it and firing it up. Best of Luck

A BIAST is used with the UFM50 antenna amplifier. It provides DC phantom power on the coaxial line to the UFM50 so a customer does not have to run a separate cable to power the UFM50. Some Lectrosonics receivers already have phantom power built in to the antenna connectors and don't need a bias T. 

Using the bias T at the receiver with a CH-20 will put 12 VDC at up to 150 mA to power the UFM50. The internal polyfuse is a 300mA unit and should allow 150 mA under any temperature conditions. The UFM50 pulls 85 mA at 12 VDC. Good quality UHF cable, should allow several miles of length before resistance drops would reduce the available voltage. Since the amplifier can only compensate for 400 feet of Belden 9913, the lowest loss medium sized cable, this will never be a problem.

The bias T consists of a feed inductor to apply DC to the BNC that goes to UFM50 (the antenna side) and a blocking capacitor to keep DC off the receiver BNC side. The RF is connected directly from one BNC to the other with only the blocking cap in series. A locking power jack connects to the CH-20, the recommended power supply. A series diode and the polyfuse are in the circuit for protection against reverse voltage and shorts.

The Bias T is effective from 60MHz to 950 MHz with less than 1 dB of loss. From 150 MHz to 850 MHz the loss is less than .5 dB. The unit is labled for customer ease of operation.

See also FAQ #089-WIRELESS for wiring a COS-11 for more normal sound levels than the 114 dB SPL assumed in this FAQ #025.

We recommend a 2k to 4k source resistor. Wire the white lead to the resistor and the other end of the resistor to ground. The black drain lead is wired to the center pin. We do not recommend just wiring the source lead (white) to ground. Below is a long post to the RAMPS group that explains why:

After my post about the COS-11 wiring, I was asked by a dealer if the lower output COS-11 red dot wasn't a simpler solution than building a resistor into the connector to simulate a three wire hookup on a MM400, which is a two wire system. So we went back and made more measurements. Lectro isn't set up to make precision sound level measurements but we faked it fairly well. I used some B&W 602's (7" Kevlar cone) to produce a surprisingly low distortion 400 Hz audio signal (less than 0.3%) and a Radio Shack sound level meter for level measurements. The Radio Shack was calibrated to a B&K sound level meter and is actually pretty decent. As further proof of the pudding, some lavaliere mics that were spec'ed as overloading at 118 dB, indeed did overload at those levels. We placed the mic about 3 inches from the loudspeaker cone so we could easily get levels of 125 dB at the microphone.

With 114 dB into the mic and using a two wire mic setup (signal taken from the black wire or drain), we set a mixer level of 0 dBm for a COS-11 grey dot(?) with a shorted-to-ground source, and measured -8.8 dBm for the same mic with a 2.2k source resistor. A COS-11 red dot measured -9.1 dBm shorted to ground and -18 dBm with a 2.2k resistor. Distortion levels in the same sequence were grey dot shorted 9%, grey dot with 2.2k 1%, red dot shorted 2%, and red dot with 2.2k 0.44%. These results are pretty much what would be expected; the source resistor reduces gain and provides negative feedback to the FET, both of which reduce the distortion. In addition, the red dot is a lower gain mic and therefore has less distortion at high sound pressure levels. What is most interesting here is that a grey dot with a source resistor has less distortion than the red dot shorted at the same sound pressure level and at the same output level (gain).

We now increased the sound pressure level to 124 dB and ran the same sequence. Distortion levels were grey dot shorted (unusable)%, grey dot with 2.2k is 1.75%, red dot shorted is 8%, and red dot with 2.2k, is 0.7%. Again, these results are pretty much what would be expected; again the source resistor reduces gain and provides negative feedback to the FET, both of which reduce the distortion. Again,the red dot is a lower gain mic and therefore has less distortion at high sound pressure levels. This is a repeat of the previous findings and once again a grey dot with a source resistor has less distortion than the red dot shorted at the same sound pressure level and at the same output level (gain). 

These numbers jive fairly well with info from Sanken's site. So my recommended hookup for a MM400 is to use a 2k or 4k source resistor with either of the Sanken microphones. The MM400a transmitter will be driven about 6 dB into compression when set for minimum gain before the microphone starts to clip and this will be at sound pressure levels of 120 dB and 130 dB respectively. Without the source resistor, the levels will be more than 10 dB lower. For the UM200 or UM400 transmitters, always wire the microphones as three wire microphones which is shield to pin 1, black to pin 2, white to pin 3, and pin 4 grounded back to pin 1.

We had available two B6 microphones and I assume (that word again) they are the standard 6 mV/Pa units since the overload point that we found was close to the specified values. We wired the two mics with a 10k resistor from tip to sleeve of the MM400 connector, which is the wiring recommended by Countryman. We found that the B6's overloaded at between 114 and 117 dB spl and that value is essentially the same as their spec sheet value of 118 dB spl. Our sound pressure arrangement was more practical than exact. Since the microphones swing their entire bias supply of 500 uA at the 114 to 118 dB spl input, they are fairly hot mics. This 10k wiring will develop 1 Volt pk at the normal input of a MM400a. The maximum level that the MM400a will handle before limiting is 0.42 Volts peak. I want to emphasize that this is limiting and not clipping. The signal will still be clean just compressed. So a B6 at maximum signal swing can drive the MM400a into 7 dB of compression. However the B6 itself will start to clip before the transmitter audio circuits begin to clip.

To reduce the gain of the standard B6 4 dB, put a 4k resistor in place of the 10k across the B6 leads in the connector, i.e., from the center pin of the MM400 connector to ground. Three dB is due to a reduced load and a dB or so due to reduced current. This will get the max output of the B6 close to the max non limiting input of the MM400a. If this is still too hot and you are uncomfortable with running the MM400a at minimum gain in loud situations, then I recommend the reduced gain B6 which is down 10 dB from the standard unit.

Click here for more information on Countryman microphones

The big change was the way we indicated the onset of limiting in the transmitter. The LED was changed to green to warn our users that this was a different system than in the past. The previous red LED on the B version came on just before limiting and users were setting the gain too low on the transmitter and having noise artifacts from both the compander and RF link. On the C version, the green LED is turned on by the same matched FET as is used in the limiter itself so the LED comes on when you are in limiting and not before. Too many users were being "scared" by the previous limit LED even though the limiter is pretty smooth and IMHO, does little damage to the signal.

In scientific tests performed on anybody we could drag into engineering, we found that the gain on the C version is set about 10 dB higher by the average user than on the B version. What is really important though, is the questions and complaints from end users about low level noise, artifacts, etc. have dropped off by a factor of ten. In reality, if the gains are set the same on the B and C versions they will perform the same.

The biggest unseen change to the C transmitters is the addition of a circular isolator or circulator to the output stage of the transmitter. This is a magnetically polarized non linear ferrite device that has three ports, any of which can be used as input or output. What's black magic about this device, is power applied to port A goes to port B, but power applied to port B goes only to port C, and power applied to port C goes only to port A. What this does for a transmitter is this: the output stage is connected to port A of the circulator and the power is delivered to port B which is connected to the antenna. The transmitter acts just like a regular transmitter so far. However, if the antenna tied to port B picks up power from another transmitter such as a two way radio or more commonly another wireless , this power doesn't get back into the output stage that is connected to port A but gets transferred to port C. Port C is tied to a 50 Ohm resistor and the incoming RF is simply dissipated as heat. The circulator reduces intermodulation between transmitters by 30 to 40 dB. Intermodulation between transmitters is probably as common as intermodulation in receiver front ends and some interference attributed to the receiver may be really due to transmitter intermod.

Frequency diversity is the sending of the same information using two different frequencies of transmission or two different transmitters set to different frequencies. The idea is that the receiver(s) will choose which frequency has the better signal at any given moment and use that as its preferred signal. This is a method that is independent of spatial diversity (two spaced antennas). It is in fact possible to use both spatial and frequency diversity and gain benefits from both. It does require two transmitters and two microphones but it gives a substantial increase in redundancy and immunity to drop outs. See page 19 of the Venue receiver manual for more information.

The Venue receiver will do both antenna and frequency diversity simultaneously. The reduction in drop outs and noise ups should be just as dramatic as the reduction going from single antenna reception to regular diversity reception. Definitely use two antennas. An antenna port is a terrible thing to waste.

Here's what I think I know about antenna orientation. If you are outdoors with the transmitter in the normal position with the transmitter antenna vertical, then you will get the best antenna strength ON THE AVERAGE with the dipole antenna arms also vertical. Inside where you have lots of reflections and the antenna signals are coming from all over, then vertical or horizontal polarizations are about the same. So vertical polarization is still, the safest orientation. There are two exceptions to this. If the transmitter antenna is horizontal, then the dipole arms should also be horizontal. Or if you can't separate the two dipole antennas by ideally at least a 1/2 wavelength (8 inches) to get good space diversity, then you can go to polarization diversity by rotating one antenna 90 degrees to the other. My advice is, if you can get the antennas 8 inches apart or more, then go with the vertical polarization.

The 195 and 200 series receivers had a dynamic noise reduction circuit that looked at RF level, audio level and audio frequency. Those three variables went through an analog multiplier and the result moved a variable high frequency filter up the frequency spectrum. Higher RF, higher audio level and higher audio frequency moved the high pass filter roll off, higher in frequency. The trick was to get the filter out of the way before the listerner could detect the roll off, so the attack time of the filter is less than 0.5 ms. In addition, for any significant high frequencies, the filter is 2 octaves above those high frequencies. There is a little different description in all the manuals for the receivers and is described as a trimode filter. 

The filter helped remove compander breathing and "halo" effects around the audio in weak audio conditions. Some designs accomplish this with substantial transmitter pre-emphasis and receiver de-emphasis, such as our 187, 190 and IFB series. This can lead to other problems at high audio levels with high frequencies, most easily shown with the dreaded key test. The down side to our filter method, is that at very low audio levels with very little high frequency content, the 200 system's high frequencies are rolled off. These audio levels are generally so low that they typically show up only when listening to room noise or microphone self noise. The 200 systems also made lavalier microphones sound quieter than they really were. The other drawback, was that it was possible to hear the filter working if the transmitter gain was turned way up in a moderately quiet room, so the self noise of the lavalier and the room noise was high enough to cause the filter to move up and down in response to the random high frequency noise. This was audible as a moderate roughness to the noise.

Just for the record, all the 195 and 200 series have had a basic flat response to 20 kHz or more. The system roll off at 20kHz is not what testers are hearing. ( I base this statement on the fact that some systems with highly regarded audio, roll off before 20 kHz and in one case before 16 kHz and I'm not aware of complaints that these systems are muffled.) At high audio frequencies, the tri-mode filter itself is out past 40 kHz. The reason for the roll off above 20 kHz in the basic system is to prevent supersonics from messing up the compander circuits.

The 400 series does not use a compander or pre-emphasis and so the tri-mode filter was left out. We figured this would help is in competitive comparisons at very low audio levels by making the system sound more open and capable of accurately reproducing small sounds and room noises. However, when some of the beta testers compared 200 and 400 series systems side by side, using a common microphone, they were bothered by the fact that the 200 sounded quieter. (Heart attacks at Lectrosonics.) Further trials by the testers, suggested by a very worried engineer, convinced them that the noise was indeed self noise in the microphone and not noise in the 400 system. This is not to imply that the 400's are as quiet as a wire.

Here came the suprise to the recovering engineers; fully half the beta testers wanted the noise reduction left in, in order to reduce the self noise of their favorite lavalier. Since we don't want to lose low level competitive comparisons because of of lack of "air" or transparency, we have compromised with a menu selectable tri-mode filter. Since users have been "hearing" this noise reduction system for 15 years in all our wideband systems, we have decided to implement it with the same parameters in the 400. The noise reduction is menu selectable at three "strengths".

The 400 series digital hybrid series transmiters and receivers will emulate other analog systems such as the Lectro IFB, 100, 195 and 200 series. Basically, we turn off the high level hybrid processing of the signal and just run the DSP as a dual band compander. We can also emulate any single band compander since that is a much simpler process but will choose those brands that have enough units in the field to make economic sense for us. At the moment we will only emulate two other brands and frankly we aren't going to heavily advertise the fact that we can do it. I don't want to get into a shoving match about whether the emulation is "correct" or not. Our dealers can simply make a recommendation of "try it and see". Interestingly, the single band emulation has most of the problems of the real unit in the areas of breathing, noise modulation and the dreaded "key test". The emulation gives only small improvements in these areas. We were hoping for much more since we could detect overload much more accurately but.... To be fair, the 200 series dual band emulation is no better or worse than a real 200 transmitter. One additional downside to the emulation is that you will have 1.5 ms of processing delay that you won't have with a real full analog system.

The emulation was created for three reasons: one is to sell a few more units by being compatible with a competitor's system, to have a reasonable upgrade path for users that have invested in 200 series systems and don't want to sell the farm to move to the 400 series and finally sometimes it is very handy on a movie set to be able to make a system imitate another when you are just one item short.

There are quite a few you can do with just your ears and some others that require a minimum of audio gear. I'll list some tests you can do, a simple explanation of what that test can show you and then a link to a longer explanation of how to do the test and how to interpret it. The best way to do the tests is simultaneously as a comparison in performance between several systems. It is easy to forget what a given system sounds like if there is a day or so between tests. Sometimes the differences are so dramatic you could remember them years later though.

1. The Dreaded Key test. Some people complain that this test only shows how well a system reproduces keys. Though true, it also indicates how well a wireless system will handle sibilants. Doing poorly on this test will generally correspond to roughness or spitting in sibilant reproduction. This due to gross overload in the audio circuits due to large amounts of pre-emphasis in the transmitter. Frankly most listeners are not critical of sibilants since if a performer sounds an "s" all they are looking for is a corresponding hiss out the sound system. However, if you are a critical listener, once heard it is hard to ignore. (See Dreaded Key Test FAQ#034-WIRELESS)
2. The Bump Test. This test will reveal the inherent signal to noise ratio of the wireless system and also how well the compandor handles low frequency audio signals. The “inherent signal to noise ratio” is the signal to noise ratio before companding. Poor results in this test will indicate a system that has what is commonly refered to as either "breathing" or a "halo" around the sound. (See Bump Test FAQ#035-WIRELESS)
3. The Input Limiter Test. This test will check to see if the transmitter has an audio input limiter (most don't) and if it does have one, how well the limiter performs. A good limiter lets you operate closer to full modulation, reduces overload distortion and improves the noise and interference performance. Screaming into the microphone is not the best method of checking this feature. (See Input Limiter Test FAQ#036-WIRELESS)
4. The Classic Walk Test. As the name implies, this is a test where one person takes a walk while talking into the transmitter,and the other person listens to the receiver output. The classic walk test is to see how far away you can get with the transmitter before dropouts are bad enough to make the system unusable. You can walk until a count of 8 to 10 dropouts occur, for example, and define that as the limit of the range. Or, walk until the dropouts or hiss buildup is objectionable according to your own assessment. (See Classic Walk Test FAQ#038-WIRELESS)
5. The Short Range Walk Test. A “short range” walk test checks to see how well the receiver handles deep multi-path nulls that occur at a close operating range with a generally strong RF signal. This tests how well the squelch and the diversity system works. This test corresponds well with real world use where the Classic Walk Test is a test of range at distances that are rarely encountered. (See Short Range Walk Test FAQ#039-WIRELESS)
6. The Hard Wired A-B Test. This requires a simple mixer two identical microphones, one connected with an audio cable and the other with a wireless system, to perform a listening test. Better than two mics would be to split one audio or mic signal so that one part goes through the wireless system and the other is direct. (See Hard Wired A-B Test FAQ#042-WIRELESS)
   

This simple test reveals how well a wireless mic system can handle high frequency audio transients and, in fact, the quality of the entire audio processing chain in the system. Set up the wireless system with a pair of headphones or a sound system at a fairly high level without feedback. It is best to be able to listen to the audio output of the receiver away from the acoustic sound that the keys themselves generate. Set the input gain on the transmitter for a normal level with an average speaking voice.

Gently shake the key ring loosely near the microphone so that the keys jingle and rattle. Shake the keys within a foot or so of the microphone, then move them gradually away from the microphone while you shake them until they are as much as 8 to 10 feet away from the mic. Listen to the audio that comes out of the receiver. Does it sound like car keys, or a bag of potato chips being crushed?
Next, have someone talk into the wireless system while the keys are shaken as in the previous paragraph. Listen for distortion of the talker’s voice while the keys rattle. Move the keys from a foot or so from the microphone and then away from the microphone to as much as 8 to 10 feet and listen to the effect on the talker’s voice.

This is a tough test for anything other than a hard-wired microphone. The results you hear will tell you, without argument, how well the input limiter, and compandor attack and decay times work in the design, and give you a clear idea of the audio quality you can expect from the system in real life. A loosely shaken set of metallic car keys on a key ring produces large quantities of high frequency transients. A wireless system that fails this test miserably, and a lot do, will also distort sibilants in the human voice. Often listeners don’t notice this high frequency transient distortion because sibilants don’t have a specific frequency but are more like random noise. Distorted random noise still sounds like noise. On a system that fails the key test, however, strong sibilants won’t have a clear, open quality but will instead have a muffled sound as if someone’s EVALUATING WIRELESS MICROPHONE SYSTEMS hand has been put between the mouth and the mic. The key test will warn you to listen closely for the effect. The key test will also reveal audio circuits that are upset by supersonics. The peak energy of jangling keys is actually around 30 kHz, well above human hearing. If the circuits in the transmitter don’t filter out the supersonics, the compandor will respond grossly. This is a valid test since sibilants in the human voice also contain supersonics. Supersonic overload will cause sibilants to sound ragged as the level is driven up and down by sounds you can’t hear.

This test will reveal the inherent signal to noise ratio of the wireless system and how well the compandor handles low frequency audio signals. The “inherent signal to noise ratio” is the signal to noise ratio before companding. This test requires listening to the system in a very quiet environment with minimal background noise. Place the transmitter and microphone in a different room from the receiver, or use high isolation headphones to monitor the audio output of the receiver. In either case, there must be minimal background noise near the microphone. Background noise at a high enough level will negate the test.

Set up the system for normal voice levels, then place the transmitter and microphone on a table or counter. Make a fist with your hand and gently bump the table with the meaty part of you hand (not your knuckle). The idea is to generate a low level, low frequency “bump” near the microphone at just enough level to open the compandor on the wireless system. Try varying how hard you bump the table with your fist to find a low level that just opens the compandor and listen to the results. When you“bump” the table, listen for background noise that sounds like a “whoosh” or “swish” that accompanies the sound of the bump.

The idea is to listen to how much background noise is released through the wireless system when the “bump” occurs, and also to whether or not the “bump” heard through the wireless sounds the same as in real life.
This is an excellent test of the difference between a single-band compandor and a dual-band compandor with DNR filtering, as well as a test of the signal to noise ratio of the wireless system. With the transmitter gain set for a normal voice level during this test, the results you hear will be what the system will actually do in real use.

It is also interesting, although not a valid test, to set the transmitter gain at minimum, then turn the receiver output up to maximum, and do the bump test again. The only reason to do this is to help understand just how much noise is actually suppressed by the system in normal use, and to emphasize the importance of proper transmitter gain adjustment.

A wireless mic system design that uses a large amount of preemphasis/ de-emphasis as noise reduction will likely do fairly well in the “bump test,” however, it may also fail miserably in the previous “car key test.” (See Dreaded Key Test FAQ#034-WIRELESS)

In this test, you will need to make some loud noises at the microphone, but be able monitor the output of the receiver in a fairly quiet environment. It’s best done with two people. The purpose of this test is to listen to how well the transmitter input limiter can handle audio peaks well above the average level. 

Set up the wireless system for an average level so that the system indicates brief peaks at full modulation with a normal voice, with the microphone at a distance of 2 feet from the talker’s mouth. While the talker speaks at a constant level, bring the microphone closer and closer to their mouth. Make sure breath pops don’t get into the microphone when it gets close to the mouth by keeping the microphone to the side of their mouth. If the transmitter has a poor limiter, or no limiter at all, the signal will get louder and then begin to distort as the loudness increases. In a system with a good limiter, the sound will get louder up to the beginning of limiting, and then will remain at a fairly level volume even as the mic is moved closer to the mouth. The character of the sound may change due to the different distances as the mic is moved closer to the talker’s mouth, but the system should be able to handle the higher levels without distortion. 

You can also test a limiter by shouting into a microphone, but keep in mind that the character of the talker’s will change as they go from a speaking voice to a shout. This makes this method harder to judge. Some wireless system designs try to prevent overload by having low microphone gain available to the user. This compromise will result in a poor signal to noise ratio when the RF signal gets weak

Here's a long reply posted to the RAMPS news group. It has a variety of wirings depending on the output level you need.

To the Group:
To make the measurements on the B6 we used the same setup as in the previous post, "Lectrosonics MM400a and COS11 red dot wiring" (See COS-11 test setup in FAQ #025). We had available two B6 microphones and I assume (that word again) they are the standard 6 mV/Pa units since the overload point that we found was close to the specified values. One unit was one that Carl at Countryman kindly loaned us several years ago and the other was from a customer who was having difficulties matching to 400 series transmitters. We found several factors that could cause possible problems with the B6 and a UM400a. The bias resistor for the UM400 series is 4k. This is higher than what we have used on the UM200 by a factor of 4. We chose the higher value because of the improved noise performance of the 400 series. In an effort to increase input signal levels to get past self noise, we increased the bias resistor value between our pins 3 and 4 to 4k. We had also run into problems getting enough output out of some big name low current microphones that wanted to look into a 20k (!) load.

The B6 microphones that we measured here were pulling 750uA and 950uA at 3 Volts which is 2 to 3 times higher than the B6 spec sheet. At first I thought this might be the problem since this much current would pull the operating point well below 3 Volts.

I called Carl at Countryman and learned more about B6's than the average person should know. 

What Carl said specific to the B6 is that the ideal voltage at the B6 mic terminals is 1.5 Volts at which point the mic will draw 500uA. This was a lower voltage than I expected and changes what we would recommend for biasing. The 500 uA does differ some from the Countryman web site values but products always change the most right after the moment you publish "firm" specs. 

Countryman's original recommended UM400 wiring inside the TA5F, from their website was:

• pin 1 shield
• pin 4 white (center conductor)
• pins 2 and 3 install a 2.8k resistor between them.
• This will give slightly more than a 3 dB reduction in signal, compared to our Lectro wiring recommendation.

Carl now prefers another configuration, which is to ignore our internal resistors entirely and wire a 1.5 k resistor from pin 2 to pin 3 and and a 3.3 k resistor from pin 3 to the the hot lead of the B6. The reason for the new recommendation is to reduce the high B6's sensitivity and get it closer to other commonly used mics.With our bias impedance and the resistors this will drop the signal 6 dB total below our original Lectro wring and bias the mic at 2.1 Volts. So the wiring is:

• pin 1 shield
• pins 2 and 3 install a 1.5k resistor between them.
• pin 3 3.3k resistor in series with the mic's white lead (center conductor)
• i.e., a 3.3k resistor between pin 3 and the mic's hot lead (white).

Another wiring, for more attenuation, will change our 5 Volt bias to 1.7 Volts on the B6 and drop the level 14 dB below our original wiring is as follows:

• pin 1 shield
• pin 3 white (center conductor)
• pins 2 and 3 install a 1.5k resistor between them.
• pins 1 and 2 install a 1.8k resistor between them.

So here are three wirings which will drop input levels 3, 6 and 14 dB below the Lectro recommended wiring. I agree with Carl that the 6 dB wiring is the best all around. However, there is a bit more to the dynamics of the situation than just limiting and clipping levels. The 3 dB wiring will let the B6 drive the UM400 into 10 db of limiting (compression not distortion), even with the gain at a minimum. The 6 dB wiring will be 7 dB into limiting. The 14 dB wiring will not drive the transmitter into limiting before the mic itself clips. The downside is that input noise levels will come up by the same amounts which might be a problem in very quiet environments. 

The standard B6 is spec'd at a maximum input sound level of 118 dB. We measured gentle clipping at 114 dB on the high current mic and 117 dB on the lower current mic. These measurements are probably not as precise as Countryman's since they were made at higher voltage and current levels (more gain) but still are certainly comparable to Countryman's spec sheet. Considering that these are higher current mics than other electrets and at clipping, the mic is swinging the entire 500 uA bias supply, the std B6's are hot mics indeed. The gain reduction wiring above does nothing as far as increasing the sound level limit of the microphone itself to more than 118 dB spl. Therefore, the lower gain B6 (-10dB) version may be a good spare mic choice, certainly for loud situations since it would handle 128 dB Spl.

The long and short of it is that Carl wishes wireless mic manufacturers would standardize the input circuits and if not that, then at least not change the inputs willy-nilly. I agree with Carl and certainly we are guilty of changing the input values when we went from the 200 series to the 400 series. I would add that it would be great for the wireless manufacturers, if the all various mics had similar output levels and similar bias currents. What makes it tough, is that the bias currents between manufacturers vary by 15 to 1, the output levels by 25 dB or more and recommended loads from 1k to 20k.

Carl made a very interesting proposal which was to just provide a bias voltage (say 5 Volts or 3 Volts), a DC blocked audio input and a ground and let the mic manufacturers recommend the resistor values for the drain and/or source loads and build them into the mic connector. As Carl pointed out there is lots of room inside a Switchcraft TA5F connector. 

Best Regards,
Larry Fisher
Lectrosonics

The classic walk test is to see how far away you can get with the transmitter before dropouts are bad enough to make the system unusable. You can walk until a count of 8 to 10 dropouts occur, for example, and define that as the limit of the range. Or, walk until the dropouts or hiss buildup is objectionable according to your own assessment. When comparing two or more different wireless systems, it is very important to repeat the same exact path for each walk test, position the receivers and the transmitters on the body in the same location with the same interconnections, and apply the same criteria to define the limit of the range, or it will not be a valid comparison. Even if the maximum range of the system is well beyond what you would normally need, this test will demonstrate the sensitivity of the receiver and how well the system handles weak signal conditions in general.

Before conducting these tests, the wireless mic system should be set up exactly the way it will be used. The microphone and transmitter must be in the exact postition on the talker’s body where they will be used, and the receiver must be connected to whatever equipment it will feed, with power and antennas connected and positioned as in actual use. Unless the wireless system is set up this way, the results of the walk tests will not be realistic. Do not remove antennas on the transmitter or receiver to try to simulate extreme operating range, as this will alter the way some receivers work, such as Lectrosonics models that use SmartSquelchTM and SmartDiversityTM circuitry.

If you have a frequency selectable system, try the walk test using at least 3 different frequencies since even tiny amounts of interference can radically change the results. If you are comparing two systems, try to select identical frequencies of operation thereby comparing apples to apples. If the receivers have scanning functions, check test frequencies that are free of interference As little as 1 uV of interference can reduce a good systems range by one half.

A “short range” walk test checks to see how well the receiver handles deep multi-path nulls that occur at a close operating range with a generally strong RF signal. This tests how well the squelch and the diversity system works. This test corresponds well with real world use where the Classic Walk Test is a test of range at distances that are rarely encountered. Do not remove the antennas on the transmitter or receiver to worsen the conditions, as this will negate the validity of the test.

Before conducting these tests, the wireless mic system should be set up exactly the way it will be used. The microphone and transmitter must be in the exact postition on the talker’s body where they will be used, and the receiver must be connected to whatever equipment it will feed, with power and antennas connected and positioned as in actual use. Unless the wireless system is set up this way, the results of the walk tests will not be realistic. Do not remove antennas on the transmitter or receiver to try to simulate extreme operating range, as this will alter the way some receivers work, such as Lectrosonics models that use SmartSquelchTM and SmartDiversityTM circuitry. 

If you have a frequency selectable system, try the walk test using at least 3 different frequencies since even tiny amounts of interference can radically change the results. If you are comparing two systems, try to select identical frequencies of operation thereby comparing apples to apples. If the receivers have scanning functions, check test frequencies that are free of interference As little as 1 uV of interference can reduce a good systems range by one half.

Find a location where multi-path reflections will be abundant, such as an area with lots of metal file cabinets or lockers, a medium to small metal building, a metal trailer, etc. Place the receiver antenna/s within a couple of feet or so of a metal surface to exaggerate multi-path cancellations at the antenna. The antennas on a diversity receiver need to be at least a 1/2 wavelength apart to achieve the maximum benefit of the diversity technique. If the receiver cannot be configured this way in actual use, then position the antennas as they will be used.

Walk around the area with the transmitter while speaking and try to find a location where a dropout or squelch (audio mute) occurs. Moving the transmitter around within a couple feet of a metal surface may help to generate a multi-path condition. The idea in this test is to see how prone the system is to producing dropouts, and to look for loud noise bursts that occur during a dropout if and when one does occur. An effective diversity system will make it difficult to find a dropout, which will tell you something about the effectiveness of the diversity circuitry. 

If and when a dropout does occur with a strong average RF level at the receiver, the receiver should simply mute the audio during the dropout and not allow any noise or noise burst to occur. An aggressive squelch system in the receiver is best in a close range situation, as it will eliminate noise bursts created by dropouts, however, it will also limit the maximum operating range as in the previous test. A less aggressive squelch allows maximum operating range, but will generally allow noise bursts to occur during dropouts at close range.

The two walk tests "Classic" and "Short Range" illustrate the dilemma of a conventional squelch system in having to choose between either close range or distant operating range, and also illustrates the benefit of an adaptive squelch system like the Lectrosonics SmartSquelchTM which automatically configures itself for close range or long distance operation as the system is being used. The tests are also a good proving ground for Lectrosonics SmartDiversityTM. 

After conducting both types of walk tests, you will have a good idea of what to expect in actual use. Some systems may provide excellent maximum range characteristics, but prove to be noisy in short range, multi-path conditions. Other systems may be great at the short range test, but be poor performers in the maximum range test. Of course, the ideal wireless system would do well in both tests.

The UH400TM has an extended low-frequency response (-3dB at 35Hz vs. -3dB at 70Hz) when compared to the UH400A. This allows for sound system measurement when testing low frequency system response.

The IM has an extended low-frequency response (-3dB at 35Hz vs. -3dB at 70Hz) when compared to the LM bodypack transmitter. Otherwise, they are the same.

This test is a matter of setting up two identical microphones, into a mixer, one connected with an audio cable and the other with a wireless system, to perform a listening test. An even better way to perform the test is to split a single mic signal so that one path is through the wireless mic system to the mixer and the other is hardwired to the mixer. Though seemingly simple, there are several mistakes that can really mess up the results:

1. The listening levels of both paths must be EXACTLY the same. Even a very slight difference in level will “fool” the ears into hearing differences that may not actually exist.
2. The test absolutely requires two people. One person speaking and the other person listening. A single person hears the sound coming from their own mouth and the sound coming from the sound system and the sounds combine at their ears to give very misleading results. This particularly true with systems that have audio delays (digital wireless systems or mixers). Even having the two people in the same room is enough to mess up the results. The only way to do the test with one person is to use a very high quality, pre-recorded signal as the source.
3. Keep in mind that the truer system is not the one that sounds brighter or warmer or bassier but the one that sounds most like the hardwired signal. You'd be surprised how many people ignore this fact.
4. If you have to use two microphones, it is always a good idea to swap the microphones and listen to them a second time to see if there are slight differences in the microphones themselves that may have been detected in the first comparison.
5. If you have to use two microphones make sure they are positioned exactly the same from a sound source or someone’s mouth so the the same audio signal enters both microphones.
6. Slight amounts of background hiss will make the overall sound of that signal seem brighter. It is a known pre-sensitization effect in the ear. Make sure that the gain of the wireless system is set properly so that background hiss is minimized.
7. If at all possible, use headphones. Again, it's not whether you like the sound it is how close is the wireless system path to the direct wired path. Headphones are much more analytical and make it many times easier to hear small errors in sound reproduction.

Switch back and forth between the cabled and the wireless setup as the listener compares the sound of each setup. This, of course, is best done in a “blindfold” manner where the listener has no way of telling which setup is being monitored, and by writing down a few notes about the results.

The previous limit LED indicator on the UM200B was wrong (misleading) and led to many users setting the transmitter gain wrong (too low). After 5 years of confused users calling the sales crew, the service troops and yours truly, I decided to change The Damned Thing. The reason I think it was the right thing to do, is that the number of calls about level problems has declined dramatically. 

I didn't do the change casually for the very reason some users have brought up; mixing B's and C's is confusing. If you set the LED's the same, the modulation of the UM200C will be about 10 dB (!) hotter than the B. 

We told users in the UM200B manual to set the limit LED so it was on 10% of the time. The users didn't want to do that since the red LED was worrying them so they were setting it for no limit LED or occasional flashing. The new recommendation on the UM200C is for occasional flashing which is the way most users wanted to set it.

Here's what can be done:
Just use the receiver monitoring. The receiver metering has not changed and the basic audio in the transmitter didn't change either, just the transmitter readout. If you use the metering on the receivers, then the B or C transmitters will be set the same. Also, just remember to set the B's for more LED flashing than the C's.

(See FAQ#027-WIRELESS for UM200B and C Differences)

Antenna gain is specified in some different ways that are confusing. The first is Gain referenced to an isotropic radiator (G subscript i) which is an antenna that radiates omni directionally and equally in all directions. A dipole antenna in this specification has a gain of about 3 dB Gi. However,an isotropic radiator doesn't exist in the real world. Any efficient antenna has more gain than an isotropic radiator. Even a simple 1/4 wave whip antenna has a gain of 3dB Gi. Since it radiates its power only above the ground plane, or into half free space it has a Gi of 3 dB. 

The other way of specifying the gain of an antenna is referenced to a dipole (or G sub d ). Obviously a dipole has a gain of 0 dB or unity referenced to Gd. The dipole is just referenced to a dipole. 

As an example, our ALP600 log periodic antenna has a gain of 4 dB Gd or 4 dB better than a dipole. If we wanted bigger, more impressive numbers, we would just rate it at 7db Gi. 

The ALP600 is more directional than a dipole of course, in fact 4 dB more directional. Gain is always proportional to directionality. Most of the directionality is in the vertical plane or up and down. The horizontal plane, left to right, which concerns you, is +- 60 degrees or 120 degrees total for a gain equal or better than a dipole.

The RG-59 will work for even longer runs than 3 feet. Assume operation at 600 MHz and 70% velocity factor for the RG-59. You do get some mismatch, but you will get it at any point beyond a 1/8 wavelength which is only 1.7 inches (!). At a 1/4 wavelength, 3.4", you get the worst mismatch but that will still give you 88% of your power. At a 1/2 wavelength, the 75 Ohm cable will look like a perfect 50 Ohm match (!), since whatever impedance is at one end is transformed to the exact same impedance at the other end. At 3/4 wavelength it looks like the 1/4 wavelength case, 88%, and at 1 wavelength it looks like the 1/2 wavelength case or a perfect match again. This just repeats for each additional 1/4 or half wavelength of cable. Added to this small mismatch, you will have the cable loss which is actually lower for RG-59 than for RG-58, since RG-59 is physically larger. You can use RG-59 for up to 15 feet or so where I wouldn't recommend RG-58 beyond 10 feet. 

You will see some mismatch to input or output filters but it isn't severe. Basically under the worst mismatch 1/4 wavelength, the 50 Ohm source or load is transformed into 100 Ohms. Antennas and filters will shift some but it isn't horrible. (RF is weird stuff.)

UCR201 receivers built between the beginning of 2004 and January of 2005 have a voltage divider problem that we became aware of in early February of 2005. External supply voltages greater than 12 Volts overload the analog inputs of the microprocessor and cause all analog readings such as audio level and RF level to be incorrect. This also causes the scan function to work incorrectly and display "interference" in frequency bands where there actually is no interference. 

The quick solution is to run the 201 from 9 Volt batteries or power it from voltages below 12 Volts. The permanent and better solution is to let us modify the unit at no charge, in or out of warranty. This can be done by us at the factory, by one of our servicing dealers or by you if you are moderately handy with a soldering pencil. We will send the surface mount diode, layout drawing and "fix it" instructions to anyone who needs the fix. Check with service to get the package. We include an extra diode because if you drop one on the rug, you'll never find it.

All units shipped after 9 Feb 2005 will have the same fix as described below and the next board rev will incorporate the exact same fix. The fix consists of a small SOT23 surface mount diode from the offending voltage divider to the 5 Volt supply at the micro-processor. The 3 terminal diode is soldered to a copper trace and to two vias on the board. It is necessary to scrape a little bit of solder mask (green screen) from the trace so the solder can bond. This fix, or attempt at a fix, does not affect the warranty. Using a Black Beauty soldering iron and blow torch will, however, result in nasty comments from the repair crew.

• Here's the S/N for the affected 201's.
• 4895 to 5649 need to be modified.
• 4795 to 4894 are in a grey area and MAY need to be modified. This can be checked by operating from a 9 Volt battery, finding a clear RF frequency on the display and then plugging in the CH20 supply and seeing if the RF level goes up 3 or four steps. If the CH20 level is different than the 9 Volt level, the unit needs the modification.
• Units before 4795 are OK.
• Units after 5649 are OK

The following units were in our finished gods and have been modified already:
5269, 5304, 5321, 5389-5394, 5417-5419, 5458-5459, 5463, 5465-5469, 5471-5479, 5484-5504, 5508-5515, 5517. Dashes indicate anything between these number are OK.

Another rule of thumb. Anything shipped from Lectro after 9 Feb 05 is modified and any units in for repair after that date will be modified.

Yes and no. If you charge the battery just until the charger indicates that it is done with the fast charge, you will have charged for 15 minutes or less. A reasonable person that hadn't read the manual, (like you and me), would think that it is completely done. In fact, it is only changing from the high charge rate to a trickle charge. If you stop charging at this point, you will get about 90% of full performance out of the battery. If that is all you need, then there is no reason not to just do a 15 minute charge. To get the last bit of charge into the battery, however, you will have to let it remain on the charger for another hour or a little more. It is still pretty remarkable that you can get a nearly fully charged battery in only 15 minutes.

The reason for the slight undercharge is that the battery gets hot from having charge crammed into it in such a short period of time. The charging current is over 8 Amps. That's why the 15 minute chargers actually blow cooling air over the battery and the charger's electronics.

Hot NiMh batteries do not hold as much charge as a room temperature battery. So it is necessary to allow the battery to cool down a bit before the last bit of charge can be put into the battery using a trickle charge of a fraction of an Amp.

The battery drain increases by a third if you use a high current mic at 48 Volts. Fortunately, the most common professional mics are relatively low current (such as Sennheisers). Approximate battery life will go from 4.5 hours at no phantom, to 3.7 hours with a low current mic, and to 3 hours with a high current mic. Battery life can be improved for some low voltage, high current Schoeps by running them in the 18 Volt position. A number of popular performing microphones are just as happy at 11 Volts as they are at 48 Volts. There is absolutely no advantage to running them at 48 Volts; it is just wasted battery power. Switch the UH transmitter to 18 Volts with these mics and reduction in battery life will be reduced by more than half. 

So the overall answer here is, check the specs of the microphone to see what minimum voltage it really requires and then set the UH to 18 Volts if possible. If you do that you should lose even less than a half hour of battery life. If the specs are not at hand, try running the mic at 18 Volts and see if you are happy with the results. In fact, have someone else switch the voltage and see if you can tell the difference, no matter what the specs say. You may be able to save some battery money.

The UH series has always had an unbalanced input. Since a mic is usually plugged right into the unit, a balanced input was unnecessary. Recently we have gone to a pseudo-balanced input to make some professional mics happier with our input. The current arrangement for the UH400 has a 500 Ohm input load on pin 2 and a 500 Ohm resistor load to a 20 uF capacitor to ground. As far as what the mic sees it is 500 Ohms to ground on both pins 2 and 3 or equivalent to a center tapped 1000 Ohm load. If the mic has emitter follower outputs for pins 2 and 3, this keeps the UH400 from "shorting" the audio on pin 3 and raising the distortion level of the mic. Mics with a floating and balanced output (a transformer or equivalent) will work into this input with the additional benefit of standard common mode rejection. The advantage to us is backwards compatibility with older transmitters and our unbalanced lavaliere mic adapter.

See MCA5X Adapter

This wiring also makes it possible to use a two wire lavaliere mic with the UH400. Simply set the phantom power to 5 Volts and wire the two wire lavaliere to pin 1 shield and pin 2 hot.

It is a gentle statement to say that our phantom wiring is merely unconventional. The UH series has always had an unbalanced input. Mics used with the plug-on units usually go right into the unit and a balanced input was unnecessary. Given that, a balanced phantom feed would be a waste of effort. So pin 3 has phantom power on it but audio is tied to ground with a series 500 Ohm resistor in series with a 20 uF capacitor. Pin 2 is audio in (hot) and is tied to the phantom voltage with a 1k resistor and goes to an amplifier with a 1k input impedance. (See FAQ#049-WIRELESS for full impedance details.)

If that isn't enough, the 48 Volts is really a regulated 42 Volts to compensate for our lower impedance feed resistors. Forty two Volts is an approximation of what the usual 2 mA, 48 Volt mic "sees" from a 48 Volt supply due to voltage drop in the DIN specified 6.8k feed resistors. One of our goals was to be able to operate higher current pro mics. With the 1k feed resistors and 42 Volts , we can provide 7 mA with reasonable voltage drop. Just don't think we cheated you on Volts if you measure the phantom voltage with a meter without a mic load. 

There is yet another facet to our madness. The lower voltage and reduced resistor loss means less power has to be supplied by that overworked 9 Volt battery that you have to buy. By using constant current diodes instead of larger lossy resistors the supply noise is still well filtered but we don't have big power losses. This arrangement also meant we didn't have to use huge 63 Volt capacitors to filter out supply noise on a 48 Volt supply. We can use 50 Volt parts on the 42 Volt lines. The inside of the UH400 is really packed and figuring out how to get around those large caps is one of the things that kept us from doing phantom power years earlier.

Finally, by using lower value feed resistors, we can accommodate some older pro mics that want 12 to 15 Volts at about 10 mA . Doing this much current with 6.8k resistors would waste a lot of valuable battery power. In fact we make a T-power adapter that takes advantage of our capability of delivering relatively large currents at low voltage

Yes, you will still have delay. The 400 series units have the same delay no matter what emulation mode they are in. The system delay is equally split between the transmitter and the receiver. If you are using a UH400 in 200 mode with a UCR201 receiver, you will still have 1.6 ms of delay in the UH400 and none in the UCR201. If you used a UH195 (fixed frequency analog)with a UCR411, you would have 1.6 ms delay in the UCR411.

The delay has little to do with what emulation is running in the DSP but is primarily due to the conversion from analog to digital and digital to anlog in the units. That conversion is exactly the same in all emulation modes.

The 3 ms delay is small and is equivalent to the time it takes sound to travel 3 feet.

The MM400B has a better seal between the battery compartment and the electronics compartment. You will notice a new line of screws in the middle of the cover. We were getting occasional leakage between compartments. The battery door now has a spring battery contact because the old solid door was putting too much pressure on the battery. The new spring door controls that pressure. The chain that was used to retain the battery door on most of the MM400A's was recently replaced with a stainless wire lanyard that is much stronger. That lanyard is carried over to the MM400B also.

The biggest change is a new off-on switch. The only way to turn off the older MM400A was to unscrew the battery door a turn or two. This could introduce moisture under wet conditions when all the user wanted to do was to shut the unit off. We added an easy to find, magnetic on-off switch on the side of the unit. The magnet activates a tiny reed switch on the inside of the case. Since this does not penetrate the case, the unit is just as weather proof as always. If you don't like the on-off feature the switch can be removed and the unit will stay on at all times. It can also be reprogrammed to become a mute switch. If this mode is programmed, the unit can still be turned off with the switch. You simply activate the switch three times to turn the unit off.

The MM400C replaced the SMA connected antenna with a permanet antenna for much better water sealing. About the same time as the C changeover, we went to a darker finish that tarnishes less.

The UCR201, UCR211 and UCR411A (but not the UCR411 -- no room for the code) are equipped with a Pong game. To activate, hold down the UP button while powering on.

The Venue has the Whack-A-Mic game. To activate that, hold down button 1 while powering on.

The R400 doesn't have a game yet but we are thinking Asteroids, because all it requires is a rotation control and a fire button. Stay tuned.

The blocks are a holdover from our early UM200 systems before we put in microprocessors. The frequency switches directly controlled a parallel input PLL (frequency chip). Since we had 256 possible switch settings using two 16 position switches and we were making 100kHz steps, we defined a block as 25.6 Mhz in size. Block 0 would have gone from 0Hz to 25.5 Mhz, Block 1 from 25.6 to 51.2 MHz and so forth. So the beginning frequency in any of these blocks would have been the block number times 25.6 MHz and the last frequency in that block would 25.5 MHz above that.

So, in a sense our blocks were pretty arbitrary and have little to do with the world outside Lectrosonics. As far as the outside world, you can call the sales department at Lectrosonics (800 821-1121), give them your location(s), what other wireless gear you have and they will make a good recommendation. If they mess up and you are using our recommended gear, we then have the responsibility to get you out of trouble. Our dealers are pretty good at frequency selection since they live in the neighborhood.

The other valuable resource is the RAMPS group since you can probably find someone that is operating in your area and can tell you what works. (See info on RAMPS in FAQ#021-WIRELESS)

I assumed that you were speaking about our gear since you mentioned blocks. If you are interested in other brand equipment, call their offices and get advice for your area. The other manufacturers are generally quite helpful and if they aren't then try some one else.

Note: "vertically downward" (e.g., a beltpack upside-down)

When indoors, antenna orientation is not as critical as when the situation is line of sight outdoors. Indoors, the RF signal is bounced around so much that orientation is not as critical. In all cases though, you are safest using vertical transmitter antennas with vertical receiver antennas. 

As far as the question of whether the antenna should be above the transmitter or below it, the only rule I'd follow is the higher the antenna the better. Higher, upside down is better than lower, right side up. Usually, right side up does get the antenna a little higher.

Yes and no. If the 1/4 wave ground plane antenna is properly built and has the necessary good ground plane, it has exactly the same gain or range as a dipole antenna. The 1/4 wave whip antennas on receivers and transmitters are just mounted to a connector for convenience and do not have the large ground plane that they should have. They are not good examples of a properly built 1/4 wave ground plane antenna.

However, the dipole antenna does not need a ground plane since the two arms of the dipole balance each other perfectly and no ground plane is needed. Generally a good dipole will work much better than the typical receiver whip antenna. The drawback to a dipole antenna is that it is twice as big as the small 1/4 wave whip antenna and correspondingly much more difficult to place on a person. In addition, the dipole antenna not only doesn't need a ground plane, it shouldn't be near a ground plane such as a transmitter case, a receiver housing or a camera body.

If you can find the room for the dipole it is generally a better choice; not because it is a better antenna but because the usual whip antennas don't have good ground planes due to space restrictions.

As long as the bends are moderate, the bends don't affect the antenna much at all. The overall length and distance to the ground plane establishes the capacitance of the antenna and the length establishes the inductance. Neither of these vary much with small bends in the wire. The capacitance and inductance establish the resonate frequency of the antenna which is the most critical factor. If the wire gets very close to metal (or the ground plane) then the capacitance to ground increases greatly. If the wire makes a loop or folds back on itself then the inductance will also increase a lot. Either of these would lower the resonance below the "cut" frequency and the antenna would not work as well.

The cable in our antennas can be dekinked by bending it with your fingers and you can do this hundreds if not thousands of times without breaking the wire.

Coiling coax or laying it on metal has little or no effect on loss. Even if the coax is terminated poorly, all fields are internally contained in the coax. The only time you will have currents on the coax is from the antenna at the end radiating back to the shield. Coax is a truly balanced cable in that the currents on the center conductor and on the shield are exactly the same and in opposite directions. This is true at least to the limits of shielding and copper resistance.

Since the magnetic and electric fields are self contained, the outside world doesn't "see" any current flow and coiling the cable or even wrapping it around a piece of iron makes little or no difference. The prohibition against coiling coax has been around for a long time. I know that, because many years ago I read an article that disabused "yours truly" of that firm belief.

The worst thing about a coiled cable is that the signal has to go through a longer piece of cable and has a little more loss than a straight shot. A one foot diameter coil is harmless for the coax that we all use.

However, coiling cables in a very small coil deforms or even damages the cable, and will mess up the cable impedance.

Yes, in its simplest form it is just a wire that is twice the length of a 1/4 wave whip. However, it doesn't work very well with a standard 50 Ohm antenna output or input. A half wave antenna has a very high input impedance (5000 Ohms or so) and requires an additional matching circuit to work well. Basically, a 1/2 wave antenna is a dipole antenna that is end fed rather than the more common center feed. 

Our SNA600 dipole antenna is a good example of a center fed dipole. It is a 1/2 dipole that has two 1/4 wavelength arms with a center feed where voltages are low and currents are high. The impedance of a center fed dipole is close to 50 Ohms and easily matched with striplines in the antenna body. 

Since the two dipole antennas are equivalent, there is no advantage of one over the other as far as gain goes. Neither antenna requires a ground plane.

In comparison to a 1/4 wave antenna, the gain is also exactly the same. For an explanation (See FAQ#056-WIRELESS). So in specific answer to your question, a wire twice as long as our normal antennas would not work well at all.

The M152 uses the same element as the M150 microphone. The difference is in the wiring inside the 5 pin connector. The M150 is wired so that it uses the pin 4 source resistor built into Lectrosonics belt pack transmitters such as the UM100, UM200, UM400 and LM. The M150 is wired as a three wire microphone with the bias being fed to the drain of the internal FET and the audio coming from the source of that FET.

The M152 uses a 3.3k source resistor built into the 5 pin female connector. The audio is picked up from the drain of the internal FET. This arrangement gives much better results with the SM transmitter and is still fully compatible with other Lectrosonics transmitters. The reasons for this new input arrangement on the SM transmitter are given in the link at the end of this FAQ.

The M152 is exactly like a two wire biased microphone and the M150 is a three wire biased microphone. Like all mics that can be wired either 2 or 3 wire, the M152 has opposite polarity compared to the M150. If you are using the M150 with the M152 you should reverse the phase using the receiver menu on one of the systems.

The sensitivity of the M152 is about 7 dB lower on a UM100 or UM200 compared to a M150. The sensitivity is almost exactly the same on a UM400. The sensitivity of the M152 is about 6 dB lower on a SM again compared to a M150. The sensitivity of the M152 is about 3 dB lower on a LM transmitter. These variations are easily adjusted with the transmitter gain controls.

There are a variety of professional lavaliere microphones that are well liked. Every user has their own opinion about which ones are the best for different situations. The problem is that these mics are radically different in their output levels, bias currents and in some cases the voltages that they will tolerate. In addition some are wired as three wire microphones (bias + audio+ ground) but others are two wire microphones with bias and audio on one lead plus a ground lead. Variations in output levels from different manufacturers can be more than 30 dB and bias currents can range from 20 uA to 800 uA. In the movie industry, the mic may be required to pick up a whisper in one scene and a scream in the next. It is no wonder that microphone and transmitter design is always a series of compromises. The input to the SM transmitter tries to overcome these compromises.

The bias voltage in the SM input is set by a servo loop that regulates the DC voltage at the microphone to a user selectable choice of 2 or 4 Volts. This is in contrast to the typical 5 Volts plus series resistor bias circuit that can result in a mic voltage that can vary from 1 Volt to almost 5 Volts. The lower voltage range can result in reduced headroom and the higher voltage can result in internal Zenering (overload) in some microphones. The SM input can handle mic bias loads from 1uA to 2000uA while still maintaining full bias voltage regulation. The servo loop also incorporates a filter that causes it to servo out frequencies below 20 Hz and rolls off the response of the lavaliere itself to wind noise, thumps and breath pops. These low frequency excursions are stopped right at the mic FET and then do not overload early audio stages in the transmitter.

At audio frequencies, the servo bias looks like an extremely high impedance resistor (constant current source) so that none of the output of the microphone is wasted in a 1k to 4k bias resistor. To prevent large voltage swings, the input to the first amplifier is a virtual ground input. This input is very low impedance so that the current developed by the mic FET is used entirely to drive the virtual ground input. Since the virtual ground input sees a high impedance source made of the mic FET's drain and the servo bias, the virtual ground input has very little loop gain noise. Since the mic's FET is operating into a virtual ground, there is very little voltage swing on the FET drain which reduces distortion on the FET compared to a conventional input.

The new input has the advantages of low noise since the noise is determined by the noise of the mic's FET and not by a bias resistor. It has the advantage of a well defined bias voltage that is not dependent on a compromise choice of transmitter bias resistors and mic current drain, i.e., two different manufacturers trying to guess what the other one is going to do. The input also has the advantage of very low voltage modulation on the FET drain reducing distortion. Finally, the input does not run out of voltage or current headroom since the bias voltage is well defined, DC current is supplied by the servo loop and AC current is "supplied" by the virtual ground amplifier. At minimum gain, the input will handle 240 uA of peak input current without engaging the limiter.

The most important advantage has to do with the limiter circuit that we have in all our transmitters since we can make it work better in the SM. Our standard limiter is a shunt circuit that shunts excess audio signal to ground when input levels are too high. In the past we have had to buffer this low impedance limiter circuit from the relatively high impedance input circuit for the mic bias supply. The amplifier that we had to have between the mic input and the shunt limiter was subject to overload at high input levels. Generally, the lavaliere mic overloaded before the buffer amp but not in all cases. Some high current mics could overload the buffer. The buffer amp also had to have unity gain so its output didn't overload and this meant this low gain amp added at least 3 dB of noise. With the new input circuit, the shunt limiter can be right at the input. No buffer amplifier is needed. This is because the virtual ground input circuit is very low impedance and is just what the shunt limiter is looking for. The advantage is that the limiter range is at least 30 dB no matter what the transmitter gain setting or input level from the lavaliere mic. There is no other transmitter that has anywhere near this limiting range for high input levels.

Some careful design went into this circuit and it is compatible with almost all of our previous mic wiring recommendations including line level inputs. Some microphones can benefit from a slightly different wiring scheme and that is noted in the SM manual. Old wiring, new wiring and compatible wiring is listed. About the only thing that doesn't work is the 40 dB attenuator wiring for very high signal level line level inputs. This can still be accomplished by putting a single 25k resistor in series with pin 5 of the TA5F input connector.

Some phantom powered mics have a balanced and floating output and some have both outputs balanced but referenced to ground. Most outputs are electronic but you can think of the two cases as being a floating transformer winding or a center tapped transformer winding. Either way works fine into a balanced (mixer) input. The fully floating output does have some common mode noise advantages when operating into a less than perfect balanced system.

In the case of the Lectro transmitter, the input is unbalanced and you have to unbalance the mic output. (As far I know, the universal box is an innocent bystander here.) The problem is that the two different balanced systems require different wiring and what is right for one is very wrong for the other.

Fortunately, you can try one way and then the other and pick the one that gives the best results. By high and low, I am referring to plus and minus polarity from the mic. By best results, I mean loudest and clearest. Generally the differences will be dramatic.

For the fully floating balanced output, ground the shield at the TA5F pin 1, the low side wire (from XLR pin 3) at the TA5F pin 1 and the high wire (from XLR pin 2) to TA5F pin 3. This grounds the shield at the transmitter where the RF is the highest and ground references the mic low side at the transmitter.

For the balanced but ground referenced mic output (center tapped) everything is the same but the low side (XLR pin 3) is not connected to anything. If pin 3 were to be grounded, in this case, half the transformer winding is shorted to ground since the winding is grounded both at one end and the center. If it is an electronic output referenced to ground (Schoeps and some others) then that output is shorted and distortion will rise on the other output.

There are several ways of making a universal setup that will work with both types of mics. The first is to use a balanced to unbalanced transformer. The drawback here is that you need a very good transformer. The second way is to put 200 to 500 Ohm resistors in series with the low side signal (pin 3 XLR signal) and tie it to ground. This will not short a ground referenced output to hard ground but to a 200 to 500 Ohm load. A floating output will have one end of the signal referenced to ground through the same 200 to 500 Ohm resistor. The down side is that you have a resistor in the mic line and will lose a little signal but that usually isn't a problem.

After all of this was written, a customer sent in his wiring solution. Though it doesn't have the shield grounded at the transmitter, which bothers me a little, he has had good luck with the following wiring. 

" I believe all's well now. I power the mics with a stand alone 48VDC power supply. I have two of them. A Neumann and Sennheiser. Both lost in effect about 10dB or so when the cable adapter to the input of the UM400 transmitter unbalanced the signal by grounding pin 3 to 1 as suggested in your wiring scheme for self powered mic level sources. Lifting pin 3 from ground brought the signal level back to normal. The wiring I am using now is this: The XLR pin 1 is tied to the shield but the shield is open at the transmitter TA5 end. Pin 2 (high) of the XLR goes to the transmitter input (TA5F pin 3) and XLR pin 3 (low) goes to pin 1 of the TA5F. The levels now are fine."

As long as the 48 Volt box ground doesn't get tied to the transmitter ground through something like a common power supply, this should work fine with either variety of mic, floating or ground referenced.

A similar problem in the UH400a transmitter was fixed in the following way.

(See FAQ#049-WIRELESS for UH400a fix)

We have a lanyard solution that is much stronger than the ball chain. It is a free part and is available by calling or emailing service@lectrosonics.com and letting them know how many kits you need and what model of MM400 you have (MM400, MM400A or MM400B).

Click here for TechNote #1029 on Bead Chain/Lanyard Replacement

The VRS is the (S)tandard receiver module for the Venue receiver system. The VRT is the more complex (T)racking front end receiver module for the Venue system. There are no other differences between the modules in either the RF sections or audio sections. If you are familiar with our other receivers, the VRS front end is similar to the UCR201 and the VRT front end is similar to the UCR411. For an explanation of what a tracking front end is see the following FAQ.

What is a tracking front end? See FAQ#065-WIRELESS

This is cribbed from the UDR200C manual and it was written years ago but is still very true:

"A number of years ago, the problem posed to the design staff was to retain the RF reliability of the Lectrosonics’ fixed frequency designs but add the frequency flexibility of a frequency agile design. The universal (but not best) way to build frequency agile systems is to design a wide open front end that will pass any frequency within the tuning range of the system. This leads to compromised RF performance in the front end with the possibility of interference, forcing the user to switch frequencies in an attempt to sidestep the interference. This makes frequency agile receivers a self fulfilling system; you have to use the frequency agility to get away from the problems caused by the frequency agile design compromises. The problem of frequency agility is further compounded when you realize that frequency changes “on the fly” cannot be made on any type of wireless system. For example, if there is suddenly an interference problem with a system in use, on stage for instance, a frequency change cannot be made without interrupting the program. Basically, the show must go on. In multichannel applications, changing the frequency of one system will usually produce all kinds of new intermodulation problems with the other systems operating in the same location. Frequency agility is not the universal panacea for interference problems. It is only another tool and a limited tool at that. The first line of defense must be the system’s basic immunity to interference. That required a new look at frequency agile receiver design.

"FREQUENCY TRACKING FRONT-END
Our solution to the wide open front end problem was to design a selective front end that can be tuned to the frequency in use. Since we wanted this front end to be equivalent to our fixed frequency front ends, this was a daunting task. Lectrosonics has always used front ends with more sections and much more selectivity than any other wireless manufacturer. The final design consisted of a total of 12 transmission line resonators with variable capacitance applied to each resonator by a microprocessor. This allows each resonator to be individually tuned by the microprocessor for any user selected frequency in a 25 MHz band. This sophistication produced a front end that was as selective as fixed frequency designs, yet could cover the entire 25 MHz range.

"HIGH CURRENT LOW NOISE AMPLIFIERS
The gain stages in the front end use some rather special transistors in a feedback regulated high current circuit that combine three parameters that are generally at odds with one another. These are: low noise, low gain and relatively high power. It is easy to understand the advantages of low noise and high power capability but why is low gain desirable? The answer is that in a receiver, low gain allows the front end to handle stronger RF signals without output overload, which is “increased headroom,” so to speak. The result of a design that takes all three of these parameters into consideration at once, is a low noise RF amplifier with a sensitivity rating equal or better than the best conventional design with a hundred times less susceptibility to intermodulation interference. Combining the high power gain stages with the tracking front end produces a receiver that is unusually immune to single and multiple interfering signals close to the operating frequency and in addition strongly rejects signals that are much farther away."

Here's an internal email from DT (David Thomas) that describes this situatiuon and its fix

I just got a Venue master that apparently wouldn't power up for a dealer. On examining it, I found it to be intact, hardware-wise, but it had corrupted firmware, such as might happen during a botched upgrade attempt.

The wonderful thing is: THIS PROBLEM IS 100% FIELD RECOVERABLE!

It is easy to think that a unit that "won't power on" certainly isn't failing due to firmware, and even if it were, how can you upgrade if it "won't power on"? Well, actually, you can and I just did!

The way the Venue's power supply works, the micro always gets power, and it is in charge of turning the rest of the circuits on or off. If an upgrade attempt fails, it is possible that the program firmware won't work correctly, which can mean that the unit doesn't power on when the power button is pressed. Nonetheless, the micro has power.

The bootloader portion of the firmware is code protected at the factory, so that in theory, it will always be possible to recover from a botched upgrade. This case was no exception. I held down the two buttons to the left of the LCD and applied power, and the display lit up happily, displaying the word UPDATE. I was then able to load the correct firmware in from the PC, and the unit is now working!

So, word to the wise (and perhaps for the troubleshooting guide and FAQ list): a Venue that appears to have power supply problems may in fact just have bad firmware loaded. The way to check is to attempt a firmware update.

David

Switchcraft makes the R3F connector which allows for adjustment of the barrel in 45 degree increments.

Switchcraft R3F Connector

If you are using VRpanel.EXE to control and monitor one or more Venue frames (VRM), you can take advantage of the software's "nickname" feature. This feature allows a name of your choosing to be associated with each receiver channel or diversity pair. The new name can be viewed or changed in the "Set Up VR dialog" and is displayed in the "Main Window" next to the faux LCD corresponding to the applicable channel or diversity pair.

VR Dialog Screenshot

If no nicknames are assigned, the default names "Rx 1" through "Rx 6" are used for single receivers, and "Rx 1 & Rx 2" through "Rx 5 & Rx 6" are used for diversity pairs.

The original reply made 13 Jun 05 is below. It is long but does discuss why the problem exists. In short, the problem is due to a ground loop between the common ground between the audio cable shield, the power supply cable to the transmitter and the power supply ground to the mixer. On 15 Nov 06 we released a new product which solves the problem by isolating the ground to the transmitter. ISO9VOLT battery eliminator

This was an email from a customer: 
We are unable to use the Lectrosonic 400 series wireless mics as camera links between our mixers and the Panasonic Varicams. While the units work fine between talent and mixer, there is a significant signal to noise problem when used from the mixer to the cameras. We have traced the problem to the use of the external power modules for the transmitters and/or the battery distribution box (Hawk Woods) we are using. Comparison with a 400 transmitter powered by a disposable 9 volt battery demonstrates the desired performance. 

(Lectrosonic makes a unit called the battery eliminator which allows their transmitters to be externally powered and therefore eliminates the accidental loss of audio between mixer and camera due to an undetected battery run down. Our sound packages, mixer, receivers and transmitters are powered from a rechargeable NP 13 battery via a power distribution tap made by Hawk Woods with a variety of cables [4 pin to single and dual coaxial ] and the aforementioned battery eliminators.) 

This s/n hiss is introduced at the transmitter and is related to a ground potential between the 400 transmitters and any other ground in the package. Dc voltage from 411 transmitters to ground of mixer measures 4.5mV. Resistance, which should be 0, measures 4 or 5 ohms. This results in audible hiss between our mixers and the cameras. Changing the power cables, audio cables or battery eliminators does not solve the problem. The more equipment added to the Hawk Woods power distribution, the louder the hiss becomes. 

This is the same power scheme we have used with the rented Lectrosonic 200 series transmitters without any problem. We are shocked that this problem is unknown by Lectro and the wider sound community.

Our First reply:
The hiss problem is probably caused by a ground loop between the common battery feed to the mixer and transmitter and the audio ground to the transmitter. The switching power supply in the UM400 is noisier than the linear regulator in the UM200. This is invariably true of switching power supplies and is a trade off for their greatly improved efficiency. The reason the ground loop problem is showing up on the transmitter is that this is probably the lowest level audio in the system. When the UM400 is run from a 9 Volt internal battery, the ground loop is broken and every thing is normal.

One solution would be to star ground every thing with separate lines at the transmitter input since it is the most sensitive point in the system. The other would be to use an isolation transformer in the audio feed to the transmitter. Neither of these is very easy. The easiest solution is to use the balanced output of the 442 mixer to accomplish the same thing.

The balanced input would be wired so that pin 1 of the 442 output XLR goes to the cable shield and pin 1 of the UM400. Pin 2 of the XLR goes to pin 5 of the transmitter (line level input). Pin 4 of the UM400 goes to pin 1 of UM400 to form the line level pad. Then pin 3 of the 442 XLR goes to pin 1 of the UM400 (along with the shield of the cable). This way the balanced output of the 442 is referenced to pin 1 (local ground) of the transmitter. This will require a 2 conductor plus shield cable of course.

We will duplicate the problem here if possible with our 442 mixer and then apply the "cure". We should get this done in the next few days and will let you know how well it works.

One other thing that could be adding to the noise problem is if the UM400 is not receiving a line level signal. If the output from the mixer is fed directly to pin three of the UM400, the sensitivity to ground loop noise will be 20 dB worse than if the signal is fed at line level to pin 5 with pin 4 tied to pin 1.

The SM input is a radically different input system compared to our previous microphone inputs. It is so superior to the old way of doing things that we will eventually introduce this input system on all our UHF transmitters. We realize this causes some confusion for our customers but the advantages are very real. The improvements are audible and make the transmitters easier to use and much harder to overload. It is no longer necessary on some mics to introduce pads to prevent overload of the input stage, divide the bias voltage down for some low voltage mics, or reduce the limiter range at minimum gain settings. For a more detailed technical discussion of the improvements in the SM servo input stage, see FAQ#061-WIRELESS. We have spent many, many hours trying to make the change from the old system to the new system as painless as possible.

For 90% of the microphones in common use, no changes are necessary to the wiring of the 5 pin connector. For some microphones the wiring can be simplified. For line level inputs, our custom musical instrument cables, adapter cables and so forth we have managed to keep the 5 pin wiring the same for old and new transmitters. You can find complete wiring diagrams for the SM transmitter on our web site. The exceptions to this compatibility are all three wire microphones (including our own M150) and a few odd wirings such as the 40 dB attenuator wiring for line level inputs. All this can be found on our web site under the Support tab, but I will list some of the more popular mics here after I discuss some of the headings on the diagrams on the web site.

The first section discusses what each pin of the 5 pin connector does. The most radical change is that pin 4 is now a voltage selector pin. You can skip this technical section if you just want to know how to wire your mic.

The next section is is boxed and labeled "Works with SM only". These wirings are specific to the SM transmitter and make wiring a Countryman B6 or E6 or a three wire microphone such as a COS-11 very quick and easy. However, these wirings won't work with older Lectro transmitters such as the UM400, UM200, etc. If you need the two wire Country B6 or any three wire mic to work with both older transmitters as well as with the SM go to the last section below labeled, "Compatible with SM and other Lectrosonics Transmitters". 

Countryman B6 and E6 are shown in the first diagram of the section labeled "Compatible with SM and other Lectrosonics Transmitters". TheB6 and E6 are two wire mics but still need special wiring because they are unhappy if run from more than about 3 Volts. The added 1.5k and 3.3k resistors shown in the diagram make the microphones compatible with any Lectro transmitters. This wiring bypasses the servo section and runs the Countryman from the 5 Volt bias supply directly. If you can use the easy wiring above in the "Works with SM only" section for the B6, it gives a little better control of sub sonics and voltage drift with humidity; otherwise there is no difference in audio response. If you have a Countryman B6 or E6 already wired for attenuation for use with a UM200 or UM400, it should still work fine with the SM. font-family: 'Times New Roman'; font-size: medium; line-height: normal;"

Sanken Cos-11 microphones, the Lectrosonics M-150 and other three wire microphones to be used with the SM will all require new wiring. If the wiring is not changed, they will have much higher output than usual and extra distortion at high levels. The reason is that the source follower wiring used with the UM200 and UM400 series is not compatible with the SM virtual ground input. The second diagram in the "Compatible with SM and other Lectrosonics Transmitters" section shows a compatible wiring that will work with all 5 pin Lectro transmitters. This wiring converts the three wire microphone to a two wire system with no changes in audio quality. The microphone polarity will be reversed so you may want to enable the phase switch on the Lectrosonics receiver. This wiring is electrically equivalent to the easy wiring in the "Works with SM only" section above.

All two wire mics (except the Countryman B6 and E6 as described above) such as the MKE-2 and the Lectro M-152 will work with the SM with no changes. The two wire setup is shown in the third diagram in the "Compatible with SM and other Lectrosonics Transmitters" section. 

The fourth diagram and fifth diagrams in the "Compatible with SM and other Lectrosonics Transmitters" for unbalanced and balanced line level inputs are the same as for previous transmitters.

The sixth diagram at the lower right for low z dynamic microphones is changed compared to previous transmitter wirings and has the addition of a jumper wire from pin 4 to pin 1. This tells the servo bias supply to shut down and set the pin 3 input voltage to 0 Volts. This additional jumper will reduce the mic output by less than a decibel when used with older transmitters.

The finish on the MM is a nickle -Teflon coating that has some very desirable qualities. It is conductive, very scratch resistant and very protective of the aluminum body in the presence of salt water or sweat. It does form a thin layer of nickle tarnish over time and exposure to corrosive elements. The tarnish is not harmful but is a cosmetic fault. We tried other none tarnishing finishes such as smooth nickle, chrome over nickle and gold over nickle and found that they looked great but a small pinhole in the finish allowed major destruction of the aluminum under the finish. A small pinhole rapidly grew into a large blister in the presence of salt water. The Teflon nickle finish kept the damage isolated to the original pinhole. 

We tried 6 tarnish removers and found one that works very well, is readily available, and has been around for more than a hundred years. Other polishes worked but not as easily or well. The tarnish remover is Wright's Silver Polish and can be found in almost any grocery. It's also available on the web.

Wright's site

Simply follow the directions on the container.

The Sanken CUB-01 boundary mic does not seem to have the usual FET output stage and also seems to have a large capacitor across the power supply lead (bias lead). This means that it can't be wired as the usual three wire microphone with the SM. The wiring below seems to work well and is fully compatible with our other transmitters.

• Our pin 1 to Sanken shield (ground).
• Our pin 2 to Sanken black wire (5 Volt power).
• Our pin 3 to a 511 Ohm resistor and the other end to Sanken white wire (audio). This matches well to our 300 Ohm input input while providing a satisfactory 811 Ohms to the Sanken mic.
• Our pins 4 and 5 no connections.

The color of the antenna caps or bands correspond to the last digit of the block number of the system. The universal color code used for resistors was picked for the code.

Score the insulation of a 50 Ohm coax such as RG58 (RG174 is OK if it is less than 5 feet) one quarter wavelength from the end of the cable. Do it very gently and don't score or even scratch the stranded shield wires since they will easily break off. Remove the outer insulation. Pull the shield down so that it has some slack in it with a bulge near the remaining outer insulation of the coax. Poke an opening in the shield with a blunt tool near the remaining insulation. The hole should large enough to pull the center conductor with the inner insulation on it through the opening. Straighten the shield so that you have a quarter wavelength shield wire and a quarter wavelength center conductor. These are the two arms of your dipole. Fold the shield back along the outer insulation of the coax cable. The center conductor is left pointing in its original direction. Now cover the center conductor and the shield with a half wavelength or more of shrink tubing to make it look nice. This antenna works best if the center conductor and the folded back shield are both in the open air since both the folded back shield and the center conductor radiate.

The times given are for turn on till shutdown (failure) of the respective transmitter. Times on alkaline AA's will drop greatly if the batteries are below room temperature. These times are typical and not guaranteed. Also note that the batteries were fresh and in excellent shape. 

The SM or MM transmitter will operate for slightly less than 2 hours on one Eveready or Panasonic alkaline AA battery, for 4 hours and 5 minutes on an Eveready 2200 mAh NiMh battery and for 6 hours and 30 minutes on an Eveready lithium battery.

The SMd dual battery 100 mW transmitter will operate for slightly less than 6 hours on two Eveready or Panasonic alkaline AA batteries, for 8 hours and 30 minutes on two Eveready 2200 mAh NiMh batteries and for more than 14 hours on two Eveready lithium batteries.

The SMq dual battery 250 mW transmitter will operate for slightly less than 2 hours on two Eveready or Panasonic alkaline AA batteries, for 5 hours on two Eveready 2200 mAh NiMh batteries and for 7 1/2 hour on two Eveready lithiums.

The variable power SMv transmitter will operate at 50 mW for 2 hours on an alkaline AA, for 4:45 on a 2200mAh NiMh AA, and for 7:20 on an Eveready lithium AA.

The variable power SMqv dual battery transmitter will operate at 50 mW for 5:50 hours on two alkaline AA's, for 9 hours on two 2200mAh NiMh AA's, and for 14:40 on two Eveready lithium AA's.

Your mileage may vary, See FAQ#087-WIRELESS.

An Eveready brand alkaline battery will power the LM for a little over 6 hours at room temperature. This time is from turn on until the transmitter shuts off. Lower temperatures, stale batteries or different brands will affect the operating time.

Since this was posted on 2006-03-02, we have come up with a third solution for interrupting the ground loop. See Isolating Battery Eliminator

What's happening is that the switching power supplies inside the UM400 transmitters create noise on the ground plane. This noise then has two paths because of the dual ground paths: one through the DC power and one through the audio ground connected by the mixer. Although mixers often have a transformer balanced output, some transformers have capacitance which can couple pins 1 and 2 to ground at high frequencies (such as hiss from DC switching supplies).

The solutions to this are as follows:

• Use external isolating transformers at your mixer outputs.
• Run a wire between the antenna connectors on the UM400 transmitters. One way to do this is to take a length of 18 ga. wire and crimp hoop lugs on each end. Then, using the SMA connectors for the whip (or external) antennas, lock the lugs onto the transmitters.

The range is not really limited by how loud the RM is or how the gain is set on the SM though these do have some affect. The limit on range is reflections from other surfaces. This effectively limits the range to 8 feet or less with maximum output on the RM and maximum audio gain on the SM. Even outdoors this short range is true since generally both the talent and the operator are standing on the ground and this is a reflective surface. The reflections cause the SM to receive echos of the transmitted tones so that the SM is trying to decode the proper sequence of tones and a delayed version of the tones. Amplifying the tones does no good at all since it is reflections that cause the failure of the decoding.

There is a positive aspect to this in that it is difficult for one RM to activate two transmitters at once which could lead to unintended results. More importantly, some prankster can't play the tones through the PA system and cause havoc. It is possible to to play the tones through a 2 way radio and allow very remote control. Cell phones don't work very well in this application due to the large amount of signal distortion (destruction?) that goes on.

The link shown below has loss numbers for just about every kind of coaxial cable that there is. The loss is per 100 feet of cable so if you have only 50 feet of cable you will have half the loss and so forth. Note that the loss figures are different at different frequencies.The frequencies are across the top going from 1 Mhz to 5000 MHz. The types of cable are grouped together in families vertically on the left side.

Click here for chart on cable losses

Some digital recorders have high levels of radiated RF and some have moderate levels of radiated RF but all radiate some RF. In any case, if the RF is the same as the operating frequency of the receiver, the range will always be affected. In most cases that particular frequency will be unusable. Some older digital recorders were so bad and radiated on so many frequencies simultaneously that any selected frequency was unusable. The only reasonable solution was to put the recorder in a trash compactor and then engage the crush mode, preferably multiple times. Modern digital recorders are much better but again, all radiate some RF.

The only current solution is to get the receiver antennas as far away from the recorder as possible. Increasing the separation by even a few additional inches may really pay off. We have some co-ax antennas that can be mounted on the straps of a bag system that will really help also. Things are easier on a cart setup since the antennas can be flown overhead to dramatically increase separation.

Here's a link to the Sound Devices site that discusses the problem in a nicely neutral manner.

Sound Devices Tech Note on RF

We have had 3 Venue systems in the field that either don't want to power up or if they do power up, run for a few seconds to several hours and then shut down. Since replacing the external power supply with a different type "fixed" the problem, we suspected the external power supply. However, we were never able to get a "bad" supply" to fail. We overloaded them by 50% and heated them to 150 degrees F ambient and they merrily continued working. The answer turned out to be much simpler than the maligned power supplies.

On Venues with a full compliment (six) of the new VRS or VRT modules, the electrical current requirements are too close to the electrical current rating of the input fuse on the main board (VRM). This is currently a 1.5 Amp polyfuse. A fully loaded Venue with the newest mainframe and newest modules pulls 16.7 Watts. This is 1.4 Amps at 12 Volts or 1.1 Amps at 15 Volts. Both these number are getting too close to what the polyfuse is guaranteed to handle, particularly at higher temperatures. We have 2.5 Amp fuses in stock that are exactly the same physical size as the 1.5 Amp fuse. We changed out all the 1.5 Amp fuse in units at Lectro and are using the 2.5 Amp fuse in all future units. We will send free fuses to customers upon request (800 821-1121 or service.repair@lectrosonics.com). We will also send new fuses to our servicing dealers. We will also upgrade the fuses at Lectro at no charge. I recommend changing the polyfuse in all Venues that use or will use the newer modules. Older modules and older mainframes use less power and will be OK but it is still recommended to change out the fuse when it is easy to do so.

The polyfuse is under the main cover and is about 1 inch away from the red power switch on the front panel. We have put pictures and instructions on our web site. Here's a link: Venue polyfuse. Two soldering iron tips heating the opposite pads of the polyfuse will remove it in seconds. If you screw it up, we'll still fix it at no charge.

The standard external power supply (DCR15/1A2U) for the Venue is an 18+ Watt unit and is OK to run the new Venues. The maximum power the Venue pulls with six VRS's or VRT's is 16.5 Watts. In testing, we have overloaded the standard Venue power supply to 25 Watts for 2 hours and it was fine. We will load the supply to even higher levels and also run it at high loads in the oven to just be sure that it isn't a part of the problem. (The tests after this was first written show the supply delivering 37 Watts at room temperature and 24 Watts at both 150 degrees F. and at 90 Volts line voltage input.)

This fuse overload would explain why checking the external supply always shows the supply to be good. The reason substitute supplies "fixed" the problem is that if a substitute supply has slightly higher voltage under load, the current demand from the Venue will be less and the fuse would then be OK. Remember, the fuse is right on the edge of working or not.

Best Regards,
Larry Fisher
Lectrosonics

Yep, we do. We've had so many customers having problems when using other brand wide band RF amplifiers in their antenna systems that I caved in and set up some wide band versions of our distribution amps and amplifiers. They have a 230 MHz bandwidth and cover all our standard blocks. At least they still have a very high intercept point with low noise figures. The prices are the same as the narrower, 2 block wide units. The web site may not be changed yet but the sales crew have the details. The wide band version of the UHFM-50 is the UHFM-230. The UMC16A is the 50 MHz version and the UMC16B is the 230 MHz wide band version.

This is one of those Zen "It depends" questions and answers. If you are in a high noise environment you will attenuate the signal and the noise the same amount and the signal to noise ratio at the input to the receiver will remain about the same. The performance of the system in the high noise environment will not change due to the 7 dB of attenuation.

If you are in a quiet environment, (the middle of Montana) then you will attenuate noise and signal the same 7 dB. The external noise is now attenuated to a lower value than the front end noise of the receiver. The front end noise of the receiver is now the determining noise floor and you've reduced the signal 7 dB so you get 7 dB less range than you would have gotten in the wilds of Montana.

Since RF is hard to see directly (magic), let's put it into audio terms that is maybe more familiar. If you have a good mic in a very noisy environment and the talent is screaming into it (rock venue), you can attenuate the mic signal 7 dB (pad), then turn up the gain 7 dB to compensate and the system signal to noise ratio will remain the same. 

If you insert the same 7 dB attenuation pad in the mic line while recording a very weak signal in a quiet room (one hand clapping in a Zen temple) then the input noise of the mic pre amp is the dominant noise source and the overall signal to noise ratio is decreased 7 dB as you have turn up the board gain to compensate for the 7 dB mic pad. The pre amp hiss is now the problem.

So as I said, it depends. In general it's not good to throw away 7 dB of signal from the antenna, but sometimes it doesn't hurt.

Here is a memo we sent to our dealers and service people about batteries. Please realize it is therefore a little blunter than our usual FAQ's:

If you are using alkaline batteries, we recommend the use of Eveready alkaline batteries particularly in our synthesized (frequency agile) equipment. Other brand batteries may work perfectly well, but Eveready is our standard for testing and analysis. In different purchases of Duracell batteries from different locations, we have found the battery life to vary by more than 2 to 1. On the other hand, we have found very consistent results from Eveready batteries. Also, in all tests we have done, a standard Eveready Energizer always gives us longer battery life than standard Duracell Procells. If customers call complaining of battery life we should ask them to try an Eveready Energizer before returning equipment for repair. Assure our customer that this does not mean Duracell is a bad battery , but that the Duracell battery was not intended or designed for the heavy current draw our equipment requires.

Always try to get the customer to try the Eveready batteries. If it is necessary to convince the customer to try Eveready's in a test, we can send them free batteries. That is cheaper than us wasting our time and theirs testing a perfectly good unit and finding no problem. Try to talk to the actual user of the Lectro product rather than someone who has been tasked with returning the unit and has no idea of what is going on. Remember these points.

1. Cold alkaline batteries have short lives. To get full battery run times, the batteries must be at room temperature or higher.
2. Standard Duracell batteries have 90% of the life of standard Eveready batteries in our equipment.
3. Some Duracell batteries that we purchased at retail were inconsistent in our equipment and occasionally "failed" after as little as 1.5 hours of operation. They may have been counterfeits.
4. The worst offenders for battery life are the UCR201, UM400, UH400, UM250, UM450, and UT700.
5. Using 48 Volt phantom power on the UH plug-ons, will drain the battery even faster.
6. Our battery life numbers are based on complete discharge of the battery, i.e., the unit is run until the battery is completely discharged and the unit shuts down. If the customer discards the battery at the first indication of falling battery voltage, then they will only get about one half of our specified battery life numbers. The customer needs to read the manual to understand what the battery indicators are really telling him. There is probably no diplomatic way to tell a customer that simple fact.
7. It is very rare that a unit is drawing too much current or has maladjusted battery status indicators. Ninety nine percent of the time, we find that there is nothing wrong with the Lectro unit. I repeat, 99% of the time, it ain't us.
8. In general, our transmitters pull more power from the battery than other brands and thus require more care in choosing and using batteries.
9. There are a number of FAQs on our web site about batteries and tests we have done.
10. Lithium disposable 9 Volt batteries have problems with our high current draw and if they are not stored before use in an airtight pouch they will fail very quickly.
11. Rechargeable Li Ion batteries such as iPower are much more economical and give slightly longer performance than standard alkaline batteries. Customers have seen iPower batteries shut down at turn on in the UH400 due to sensitive protection circuity in the iPower battery.

Thanks,
LEF

All rechargeable batteries wear out. Your car battery is a well known example. Every time a rechargeable battery is used, it loses a small amount of capacity. Whether that becomes a problem or not depends on how much capacity you need. Here's an example: let's say you have an SM transmitter and the NiMh AA rechargeable batteries that we provide, currently 4 each 2200mAh Eveready batteries. These batteries happen to run your SM transmitter for 4 hours and 5 minutes (4:05) and that is just fine with you because you only have to operate the transmitter for exactly four hours every day. In few weeks, you are going to be very disappointed, because after 20 charges, the battery will only power the transmitter for 3:59. The reason is that every time you charge the batteries you will lose a tiny amount of capacity and you don't have any spare capacity to lose. In fact after a year of use, the batteries will probably only run the transmitter for 3 hours. There is nothing wrong with the batteries or the transmitter. Rechargeable batteries just slowly wear out.

The solution is to start out with a higher capacity NiMh AA battery such as an Eveready 2500 mAh cell or a Sanyo 2700 mAh cell. You will now get almost 5 hours of battery life initially and you won't be down to 4 hours of life until 200 recharges or 6 months later. This is even after assuming that the higher capacity batteries wear out twice as fast. (Eveready quotes the 2200 mAh battery as being rechargeable 1000 times and the 2500 mAh being rechargeable 500 times.)

This is an extreme example caused by your run time requirements being so close to the limits of the smaller capacity battery. If you only needed 2 hours of battery operating time, the smaller battery would be fine and would actually operate for many years before falling below your requirements. If you look at the economics of the battery for the SM, the rechargeable batteries really make sense. Top quality alkaline batteries will only run the SM transmitter for less than two hours and cost about 40 cents apiece. Two hours is rarely long enough. Lithium AA batteries will run the SM for 6 hours but they are $2.00 each. High capacity NiMh AA batteries are $3.00 each but the cost per use is a tiny 1.5 cents. Compared to a lithium AA battery, just two uses of a rechargeable AA pays for the battery.

This discussion is also applicable to the iPower 9 Volt lithium ion polymer rechargeable battery. Exactly the same arguments can be made about both saving money and gradual wear out of the battery with recharging. (See FAQ#089-WIRELESS)

Q: So do rechargeable batteries wear out? 
A: Yes

Q: How many times can I recharge them? 
A: It depends on your run time requirements.

Q: How do i know when they are worn out? 
A: When they no longer meet your requirements.

Here's a long post that appeared on the news group RAMPS. What is RAMPS? (See FAQ021-WIRELESS)

To the Group:
Here's the results of the new version of the iPower 500 mA rechargeable Li Ion Polymer battery tests. We also ran tests on the Ultralife 9 Volt single use lithiums and standard Eveready alkaline 9 Volts for reference. The batteries were run down multiple times in a UM450 because it is a power hog, 125 mA at 8 Volts or 1 Watt (!). We then ran the batteries down in a UM400 (a power piglet) just for further reference. In all cases the transmitters were run continuously until shut down of the transmitter. The batteries were then put on the charger and recharged. It made little difference if the battery was charged for one hour (green indicator just came on) or overnight. The runs were at 72 F ambient. Here's the results:

The first iPower battery in UM450:
3:18
3:17
3:17
3:09
3:17
3:18
3:16

The second iPower battery in UM450:
3:23
3:21
3:17
3:17
3:14
3:10 (double charged. See end of post.)
3:09

Brand new from the factory Ultralife Lithium in UM450:
1:57

Few months old Eveready Alkaline in UM450:
2:08

The Ultralife does not like the high current demands of the UM450 and does worse than an Eveready alkaline. The 500 mAh iPower has almost exactly 50% more capacity than either, at this high current load. If you remember the previously posted cold tests, the iPower also did quite well, though that was on a 400 mAh battery. Below are tests in a UM400, a more normal load:

The first iPower battery in UM400:
5:32

The second iPower battery in UM400:
5:09

Two brand new from the factory Ultralife Lithiums in UM400:
7:11
7:03

Two (few months old) Eveready Alkalines in UM400:
4:27
4:19

Here the Ultralife Lithiums come into their own, since the current drains are more reasonable and show a 62% increase in battery life compared to the Eveready alkalines as standards. The iPowers still show a good 20% increase over the alkalines.

This UM400 was a block 29 and seemed to pull a little more current than the average UM400. The ratios of battery life should still be valid. If you are getting a particular battery life on alkaline Evereadys, simply add the proper percentage. If you are using ProCells (Mallory) the iPower improvement will be an additional 10% more (30% instead of 20% for example).

Final notes: 
Doing a 100% discharge on Li Ion Poly batteries (iPowers) is worst case for usage.
The iPowers work well in the cold.
We tried a "double" charge on the iPower with no big effect. We charged it completely, removed the battery, and started the charge cycle again. It took about an hour for the charging light to go from red to green again. I don't know what that means, but it didn't look good. However, the capacity of the battery went down only 4 minutes. I don't think it was significant, but....
We will ship the iPower system with the UM450. 

I can't stress enough that "YMMV".

Best Regards,
Larry Fisher
Lectrosonics

Alkaline batteries, though very good at room temperature, cannot deliver much current at lower temperatures. Battery life can be as little as one third normal on a cold day and even less if they cold soak for any length of time. Life can be as little as just a few minutes at -20 F.

If you must use disposable batteries (non rechargeable) then lithium batteries are the only good choice. They have shorter life at low temperatures but are still much better than alkalines.

In the AA battery size, lithium and NiMh batteries are the best cold weather choice. At low temperatures NiMh have almost as much life as at room temperature and are rechargeable to boot. Our tests indicate that the NiMh AA batteries when used in an MM400 or SM transmitter have 75% of normal capacity at minus 15C (-15C) or +5F. We recommend the Eveready NiMh batteries and 15 minute charger that we provide with the SM, SMD and SMQ transmitters. One precaution is that the batteries cannot be recharged if they are cold. They can be used cold without any problem but must be at about room temperature to be recharged. The lithium batteries are also good at low temperatures but they don't like being cold for long periods of time. If cold soaked at 5 deg F for an hour, they still have 95% of their normal capacity. Allowed to cold soak overnight they only have 50% of their capacity. Here are some run times in a cold SM transmitter at different temperatures:

At room temperature:

• NiMh 4:02
• Lithium 6:01

At 5 def F cold soaked for one hour:

• NiMh 2:52
• Lithium 5:45

At 5 deg F cold soaked for 16 hours:

• NiMh 2:46
• Lithium 3:19

In the 9 Volt battery size, NiMh batteries perform as well cold as they do at room temperature but they don't have much battery life (capacity) cold or warm. At one time they were the only choice for very low temperatures but LiPoly rechargeable batteries are now available that have more capacity than alkaline batteries and perform very well at low temperature. They are currently sold under the iPower brand and are available on the internet, from some dealers and from Lectrosonics. (See FAQ#086-WIRELESS)

Here is schematic for muting a two wire microphone. The in's and out's can be swapped. You can treat the arrangement as if it is just a normal two wire mic. That means that it will have to be hooked up to a transmitter or other bias supply in the normal manner. There will be a faint click due to RF in area from the transmitter but it won’t be objectionable for your application. This only requires a single pole, single throw switch. When the switch (or push button) is closed the capacitor is connected to the two wires and shorts out the audio. There will be a little audio bleed-through, mostly at low frequencies. The larger the capacitor, the less bleed there will be. The resistor across the switch is necessary to keep the capacitor lightly connected to the bias supply so there won’t be a large pop when the switch is closed. A better but much more expensive solution is to use a unit designed to be silently muted by a DC control signal.

See MUTE

See also FAQ #25 for an earlier post with some measurements. The resistor recommendations are a little different than in this FAQ #94 since the levels in FAQ #25 were at 114 dB SPL. 

(See FAQ#025-WIRELESS)

The recommendations below are for more "normal" usage.

We recommend wiring the standard level COS-11 in a 2 wire configuration with a 1k resistor in series with the white wire (source wire) to ground. This reduces the output of the standard mic by 6 dB which will prevent overload of the input buffer on the UM400. This isn't necessary for the SM series since it cannot be overloaded by the COS-11 under any circumstances. It does make the microphone compatible with both transmitter series. 

The Red Dot Cos-11's are designed to have 9 to 10 dB less output than the standard COS-11's and can therefore be wired with the source lead directly to ground. 

Adding a 1k resistor to the source lead of either microphone will drop the 2 wire output by 6 dB compared to the source being wired directly to ground. 

Using a 3k source resistor will drop the output of either mic an additonal 6dB compared to using the 1k resistor or 12 dB total. This much gain reduction should not be necessary and is for information only.

We also recommend wiring our pin 2 to pin 4 to make the wiring fully compatible with our UM400 and older Lectro transmitters. A schematic is below.

The early iPower batteries were larger than a standard 9 Volt battery. Shortly after their initial release they made a smaller case for the batteries.

The new smaller 500 mAh units are marked the same as the older, larger 500 mAh units. The only difference I see in the case is that the older unit has a small recessed circular area only around the small positive terminal. The new smaller 500 mAh unit has a large rectangular recessed area that encloses almost the entire top of the battery. It's recessed about 0.030" deep. The new batteries are actually smaller than standard 9 Volts and should never be a problem as far as sticking. 

We were told in early 2006 that it would be several weeks before the first new batteries were available. We only stock the smaller cells because we don't want them stuck in our units. If you order from some one else, I think I'd do it verbally to confirm what you want. If you have some of the larger batteries you might contact iPower and see if you can return them. Do a search on the RAMPS newsgroup for contact information.

Here is the way to correct the problem of the XLR insert sleeve coming loose and causing the battery door to "flop" around as shown below.

You will need a .050 inch allen wrench (Lectrosonics part number 35700). We will mail you one at no charge if you ask our service department at 800 821-1121 or service.repair@lectrosonics.com.

On the UCR401, push the XLR and sleeve until the sleeve face is flush with the battery door as shown below.Then tighten the allen screw. Don't overtighten.

The sleeves on the UCR201, UCR211, UCR411, and UCR411A are not flush. Push the XLR sleeve until the sleeve face sticks out 1/16 of an inch (approximately) as shown below and then tighten the allen screw. Don't overtighten.

Thanks to our service manager, Dean Slotness, for the pictures and explanation.

The SM weighs 60.5 grams (2.13 ounces). A lithium battery weighs 15.5 grams, an alkaline battery weighs 22.5 grams, and a NiMh battery wighs 28.8 grams. The SM with a lithium battery is a formidable lightweight with 6.5 hours battery life, 100 mW RF output and only weighing 76 grams (2.7 ounces).

On the side of the newer units, on the blue metal strip, you should see a +10 dB marking indicating that these units have an increased gain compared to your older units. Here's what we said on RAMPS (See FAQ#021-WIRELESS).Though a little long it explains our rational to making the change and what some users were seeing as far as typical gain settings on the SM transmitter. We didn't make this change lightly because we knew it would cause some confusion. Below is the slightly edited version of the RAMPS thread. The actual thread is at Link to RAMPS thread. We kept the display at 0 to 44 due to memory limitations.

Larry Fisher Sep 6 2006
To the Group: 
After some 6 months or so of field experience with the SM transmitter, we are strongly considering increasing the overall gain of the SM and SM series transmitters by 10 dB. This would increase all input configurations including line level input. The display would now show 10 to 54 dB of gain rather than the current 0 to 44 dB. The same displayed numbers for gain would represent the same gain on either version, i.e., a setting of 30 dB would be the same on either version. Equivalent input noise, response, limiting and so forth would not change. 

From what I've read on RAMPS and various emails, indicates we are a little light on gain. Also, I think some users don't like running the gain at maximum levels even if that is the best setting in a given case. Any and all feedback from the real world would be appreciated. Best Regards, 
Larry Fisher 
Lectrosonics

Scott Farr Sep 6 2006, 9:27 am 
I run my SM transmitters at 38 on the gain setting never any higher and I am using TRAM TR50's 
Scott

Larry Fisher Sep 6 2006, 10:13 am 
Thanks Scott, 
You are within 6 dB of full current gain (44). You would still set the transmitter to 38. What is the lowest you remember setting the transmitter? 
LarryF 
Lectro

Scott Farr Sep 6 2006, 11:10 am 
I think around 32

wyatt Sep 6 2006, 12:05 pm 
I am using 8 SM transmitters on my current production. I am using an assortment of lavs, and my gain settings varry accordingly. The lowest gain I have used to date would be 24 using a cos-11, outdoor scene, times square. the highest would be 44, indoors, using either a b6 (grey band) or a TR-50 (i dont quite recall which at the moment). My normal operating gain for a cos-11 has been between 29 and 31. I think a gain increase would a beneficial improvement, especially considering that you can get most of the lavs that we, as soud mixers, employ daily with attenuated output levels. I have yet go below 24db on my transmitter gain.
Wyatt Tuzo

Larry Fisher Sep 6 2006, 2:29 pm 
Thanks Wyatt, 
More good info. 
LarryF 
Lectro

Scott Smiith Sep 6 2006, 8:20 pm 
Larry, 
This is something I have noticed as well (although the output of the mike used definitely plays into the equation). I would say at least 6 db would be helpful, and there is certainly nothing wrong with 10 db, as long as it doesn't risk overloading the front end preamp. Scott D. Smith C.A.S.

Larry Fisher Sep 7 2006, 12:13 pm 
Hi Scott, 
When the gain is set to minimum, even with the additional gain, the SM will still require 250 uA of peak current from the mic before the limiter even begins to engage. This would require a lavaliere that pulled 500uA of bias. This is higher than any pro mics we have measured. Another way of saying this is that at minimum gain, any known lavalliere will run out of bias (clip) before the limiter starts to limit let alone overdrive the preamp. Further, even at minimum gain there is still more than 30 dB of limiting before the limiter gives up and allows the preamp to overload. As I said earlier, I think we were being way too conservative on gain. Field experience seems to be bearing this out. 
Best Regards, 
LarryF 
Lectro

LaFayette Sep 8 2006, 3:20 am 
Greetings Larry,
Would this be an upgrade mod on existing units or only for new versions? 

I've set mine from 30 w/ Sankens outdoors to 38 w/ Sonotrims indoors. It has been suggested to go w/ red dot Sankens since the regular ones tend to be hot w/ SM's. Although, I also hear that the Sankens wired specifically for the SM sound better anyway, more low end. Should I save my money on new Sankens and just buy the upgrade on the SM? Cheers, S.L.

Larry Fisher Sep 8 2006, 4:42 am 
Hi, 
There is no downside to running the current SM at maximum gain or close to it. Performance of the current units at 44dB gain (wide open) will be exactly the same as the new units at 44 dB gain (10 dB left). So if you have sufficient gain now to handle the mics you have, then it is not necessary to change the units. This is not a necessary "upgrade" and won't be automatically done to units in for other repairs unless the customer requests it, unlike mandatory upgrades that we always make to repairs. Mandatory upgrades are due to a mistake we made and are done at no charge. 

Older units can be changed to the new gain structure and corresponding firmware. There will be a $70 charge to open, test and reseal the unit. This is the same as for the RM addition mentioned earlier on RAMPS. On the other hand, you can get both done at once for the same charge. Depending on the age of the SM you may also pickup a new emulation also.There is no charge for the firmware and new value resistor themselves; just the work involved in opening, closing, sealing and testing the unit. If we are already inside a unit for other reasons, there will be no additional charge for the gain change. As I said above, the customer must request the change since if they are happy with the current gain structure, we are very satisfied to leave it alone. 
Best Regards, 
LarryF 
Lectro

morantzsound (Steve Morantz, C.A.S.) Sep 9 2006, 11:35 pm 
I was told when I bought my SM's anywhere form 24 to 30 was average. I usually stay around 27 or 28 and make adjustments at the board or Deva. It has worked pretty well for me, but will start experimenting with higher outputs. 
Steve Morantz C.A.S.

You can add a RF choke to each lead of the XLR in the Schoeps mic. We have helped several customers with an RF bead placed around all three wires at the XLR connector inside the mic. The Mouser part number is 623-2643000301. The Fair-Rite brand part number is 2643000301. Placing short leaded 100 pF capacitors between all the XLR pins and one to the ground lug might help also, but the chokes are most effective. Schoeps has a factory modification that is more elaborate, that works very well.

--Clearing up the Issue of Higher Transmitter RF Power and IM

For many years, Lectrosonics has built wireless transmitters that are higher-powered than those from other vendors. In addition, when we say a typical power of 100mW, we don’t mean that we had one engineering sample reach that power with a westerly wind. We center our production on 100mW with small manufacturing variations both up and down. We have built 100mW units for what we feel are four good reasons:

1. Increased range
2. Fewer dropouts
3. Reduction of interference
4. Better signal-to-noise ratio

There have been discussions about problems with higher powered units like ours, but the only real negative of higher power when properly implemented is a slightly increased drain on transmitter batteries. Since most of the battery power is used supporting digital processing in the Lectrosonics’ transmitters, the increased RF power is only a minor consideration. Another way of saying it is, if you have made the choice of digital processing in the transmitter, you might as well have increased RF power too, since it doesn’t change the battery life that much. The new generation of rechargeable batteries can reduce battery costs for all wireless systems and battery usage isn’t quite as important a consideration as it was several years ago.

--Design Considerations

Careful design has removed the two other problems that are sometimes discussed when higher power is considered. The possible problem of increased intermodulation (IM) in the output stage of the transmitters at higher power has been solved by using an output isolator in the antenna circuit. This prevents two transmitters that are physically close together from creating IM products. This isolator is unique to Lectrosonics’ transmitters and is in all transmitters except the LM series. Possible IM in the receivers has likewise been solved, because Lectrosonics’ receivers have always had higher power RF stages for substantially better IM rejection than competing receivers, so the increased transmitter power is no problem whatsoever. 

The technology used in the Lectrosonics products is all fine and good but the real question revolves around what happens in actual use. What about a real world situation with lots of transmitters on a stage? We’ve been hearing stories that 100mW transmitters are absolutely unusable in a stage environment and that 100mW systems will wipe out the wireless operations of theatres over city blocks, if not the entire Eastern Seaboard.

--Real World Numbers

Let’s first apply a little common sense and then run some numbers, first looking at general reception issues and then those specific to transmitters:

If intermodulation was really a consistent problem with 100mW transmitters then it would be only slightly less of a problem with 50mW transmitters, all things being equal. The only way to change from a supposedly “real” problem to a rare occurrence would be to make a radical change in transmitter power such as down to 5mW or less. A 3dB difference in power doesn’t mean much when typical signal levels on a stage are making 50dB swings as people move around. Just as an increase in power from 50mW to 100mW doesn’t solve all dropout problems and only extends effective range by 30%, reducing power from 100mW to 50mW won’t solve all IM and transmitter interference problems if there were any to begin with. In the same way, even a 3 to 1 change from 100mW to 30mW is insignificant when signal levels at the receiver can vary by 100,000 to 1, i.e., the previously mentioned 50dB.

To show why there shouldn’t be IM problems at either power level, let’s do the numbers. We will analyze the signals that would be present at a receiver from a 100mW transmitter in the worst possible situation and then in a real world situation. Even better, we will use published numbers from other manufacturers rather than our own measurements (even though they are about the same). We will do a third order analysis since it is accepted as being the worst case. Second and fourth order products are not a problem because they are totally removed by the receiver’s RF filters and fifth order and higher are at much lower levels than third order.

NOTE: The Firmware listing & information has moved to Wireless Firmware Lookup on the support site.

Click here to redirect to Firmware History and to lookup product firmware history

This posting about Lectro firmware has taken a while and has engendered a lot of discussion here at Lectro. The firmware version and update issue is rather confusing. Unlike computer software, where the latest version is always better, "it ain't necessarily so" when it comes to our products. The simple reason is this: the firmware is very secondary to the hardware. Most firmware "updates" are made because of hardware changes, some of which are forced on us by outside suppliers. Here is just one example of many. For the UM400 we have 3 branches of the firmware. The 2.x branch is for units using a Philips 7026 phase locked loop, the 3.x branch is for units with the Texas Instruments TI2050 PLL and the 4.x branch is for units with the National LMX2353 branch. Obviously the 3.x branch is not better than the 2.x branch and the 4.x branch is not twice as good as the 2.x branch. Almost all of our firmware revisions have to do with hardware changes and not improvements in the product. We have to do revisions when sometimes the only change is that a company has discarded one IC package for a different one that has more or fewer pins and more or fewer functions. Just as often, we are informed that a part is being dropped from production by an IC manufacturer because the 100 thousand a year that they sell to 4 or 5 companies is not enough to keep the part in production. So we find a similar part, change the PC board and revise the firmware to handle a new command set for the new part. We also do many revisions to make our manufacturing and testing simpler or simply better. It may be easier to put a correction factor for modulation at different frequencies into the firmware than to select varactors and resonators to make deviation uniform across a block. That doesn't mean that older firmware with select parts is better for the customer than standard parts and newer firmware with a correction factor. All this is to say, we've pared down the hundreds of firmware revisions to those that truly affect the end user. Some of these listed changes are the fixing of firmware bugs and some are added features. If the bug has never affected your system and/or if you never need the feature then it may not be worth your effort to "upgrade" the firmware. Further, adding new features may also require hardware changes that may not be possible or may be more expensive than it is worth.

To summarize, a higher firmware number by itself is meaningless. It does not mean a better product. We have listed changes by date of manufacture since serial numbers are not reliable indicators of firmware version. This is because our products are manufactured on many frequency blocks and serial numbers are assigned to units months before actual shipping. Our service department can help you with specific questions about your unit. Please have your serial numbers available so they can check the date of manufacture.

This is the same text as that engraved on the back of the UCR401 and UCR411a receivers.

FOR SPECTRUM ANALYZER
Press all 3 keys simultaneously to either enter or exit the spectrum analyzer. MENU key to stop, zoom or start scan. Zoom is indicated by < > icons. In zoom, since  most data is off the screen, the cursor is  centered and the data scrolls. Use the UP  and DOWN keys to scroll. To save cursor  frequency, press all three simultaneously  and then select "use new". To clear  spectrum, turn power OFF briefly.

FOR PILOT BYPASS
Step the menu key to the MAIN window.  Press the MENU and UP keys together for  b bypassed or p normal plot.

FOR THE 1 kHz TEST TONE
Step menu key to SETUP/EXIT window.  Step SEL UP key to SETUP/TONE  window. Press TONE (MENU) key. Press  TONE (UP) key. Step LVL (UP/DOWN)  keys to set tone level. Press MENU key to  stop and EXIT tone.

TO LOCK AND UNLOCK
Press and hold the MENU  key for 5 seconds.

TO RESET BATTERY TIMER
Press and hold MENU and DOWN key together  for one second.

The best way is to use three 2-way splitters on 3 of the outputs from your 4 way multicoupler and run the last output as usual for a total of 7 outputs. You will have a 3 dB+ loss on the split lines. If you are working at relatively close distances (under 100 feet) and/or in a RF noisy environment, you will not notice any change in performance. The reason is that if you are close you have plenty of signal and a small loss won't be noticeable. If you are in an RF noisy environment, the splitter will drop the signal and the noise both by 3 dB and the signal to noise ratio at the input remain the same. This is true until the attenuated external noise is less than the front end input noise of the receiver. I would try the passive splitters and save the money. Plus, splitters are always handy things to have around even if you eventually get a UMC16B.

Here's a reply to a problem with range on RAMPS. The test described can be done with any of our UHF receivers.

Vin,
I didn't get into the thread before, because I wanted to make some distance versus RF display measurements before I started mouthing off. What we wanted to determine was a setup such that a user could make a simple, repeatable measurement to check our equipment for proper operation. This "test" should show the proper operation (or not) of a Lectro transmitter and a Venue receiver. Here's the setup:

1. Two vertical right angle whip antennas, type A500RA, are on the back of the Venue receiver.
2. The Venue receiver is about 1 meter above the ground with no large metal objects close to the antennas other than a metal or plastic cart surface under the Venue and the metal case of the Venue itself.
3. The UM or SM transmitter is held by the metal case at arm's length away from the body with the antenna vertical.
4. Line of sight between the transmitter and the Venue.
5. All done outdoors.

We are not trying to duplicate a real use case here but we are trying to eliminate all variables such as body and clothing absorption (15 dB), antenna gain factors (0 to 5 dB), defective antenna amplifiers (30 dB), bad cabling (60 dB), reinforcement from room walls (6 dB), etc. Under our simplified but repeatable conditions, whether you have RF interference or not, the Venue RF display for a good system will be full scale at a separation of 100 meters. You may get dips due to multipath but moving a foot or more in any direction should get you out of the multipath. Again, the maximum reading is the correct one for this test since multipath will rarely increase the signal more than 6 dB but can decrease the signal by 30 dB.

We did this test in our parking lot with a clear line of sight between a UM400 100 mW transmitter and a block 26 Venue receiver. To double check the real world against theory we did a path loss calculation. Here's the path loss formula:
The formula for path loss between two 0dBd antennas given a separation D and a wavelength y with y=0.461 meters at 650 MHz is
Path Loss in dB = 22 + 20 Log (D/y).

For 650 MHz at 100 meters separation the Path Loss = 69 dB. Since a 100 mW transmitter power level is is 20 dBm then the signal at the Venue antenna is 20 dBm - 69 dB = -49 dBm. Full scale on the Venue is 1000 uV or -47 dBm. (Remember 0 dBm at 50 Ohms is 0.224 Volts not .775 Volts as in a 600 Ohm audio system.) In any case, the -47 dBm at full scale is scarily close to the theoretical path loss result of -49 dBm. Ground bounce reflections can add 6 dB to the numbers and diversity antenna addition can add 3 dB. In any case, with some hand waving, the actual measurements seem very valid.

As long you have line of sight between the transmitter and Venue, it doesn't make any difference what kind of ground surface you are on. Interference, even at high levels will only increase your RF readings on the Venue scale. The European 50 mW units will shorten the 100 meter readings to 70 meters. The LM would be about 70 to 90 meters depending on the particular LM. The SMq, UM250 and UM450 250 mW units will increase the full scale range to 160 meters.

This test only checks the power level of the transmitter and the RF operation of the receiver; it does not address any added factors such as interference in the area or the rest of your setup. It does give you a starting point for diagnosing problems but it is only valid under the 5 conditions above. If you perform the above test and the results are good, then you can start adding antennas and cabling to the system. The remote antennas, cables, amplifiers, etc., should give at least the same distance results as this right angle whip test or there is something wrong. Before I get buried in replies that say full scale at 100 meters is impossible, reread the 5 strict requirements above; this is a very special test setup.

If you get the correct readings for RF level, then the next most probable cause of short range is interference. Then the Venue scanning function should find the problem.

Best Regards,
Larry Fisher
Lectrosonics

This was a posting to RAMPS about "Calculating Intermod Frequencies".

Here's a chance for a 40 page dissertation that I'm not going to take. In the interest of keeping it simple and easy to remember, I'll make some general statements that are 99% true, i.e., errors are 40 dB down and good enough for sound mixers. ;-)

Intermod is calculated in exactly the same way by all the programs and has little or nothing to do with i.f. frequencies.

Odd order intermod (3rd, 5th, etc.) is much more of a problem than even order intermod (2nd, 4th, etc.) because with odd orders it is possible to generate interference that is very close to the receiver frequency. This interference can therefore pass straight through the receiver front end filters. Of the odd orders, third order intermod is of greatest concern because the interference is always at a much higher level than 5th or 7th.

Second and 4th order are of lesser concern because the carriers that generate them cannot be close to the receiver frequency and will be filtered out by the front end. (Our IFB receiver is one of the few receivers for which this statement is not true. The IF is so low, 70 kHz (!!), that the image at 140 kHz from the operating frequency can be generated by 2nd order intermod.)

Intermod due to wireless transmitters getting into the receiver is not a factor if the transmitters are all, repeat ALL, 20 feet or more away from a good quality receiver. Intermod between a transmitter and a TV station does not follow the 20 foot rule for the TV station component obviously, or for two TV stations.

Intermod generation between transmitters is more of a problem in most situations particularly if the transmitters have standard output stages and are closer than 5 feet apart. The intermod frequencies are exactly the same as in receivers and any intermod program will catch them. I have seen fairly strong 5th and 7th order when transmitters are only a few feet apart.

A quick discussion of image frequencies is appropriate here since most programs also calculate images and this is where different brands do have differences. The knowledge of the i.f. frequency is critical in determining where the image frequencies will lie. A first low i.f frequency says the image frequency will be a small distance from the receiver operating frequency. Most modern receivers have a high first i.f. such as 244 MHz. This puts the image at 2 x i.f = 488 MHz away from the tuned frequency. Almost 500 MHz away means the front end filters can strongly suppress it. Generally, with modern i.f.'s, other wireless transmitters don't cause image problems. It's from other high power transmitters in the environment.

If the above is the Reader's Digest version of intermod stuff, here's the Cliff's notes:

1. Third order intermod is the biggie.
2. Any program will calculate third order.
3. Third and higher order between transmitters is usually the main problem.
4. You only need to worry about intermod if the transmitters are close to the receivers (less than 20 feet) or each other (less than 5 feet) or in other words, always.

Since the original question was "Mathematical formula in finding Intermodulation free frequencies" here it is:
1.Find the difference between two transmitter frequencies. 
2.Subtract the difference from the lower frequency and add it to the higher frequency. Those will be the two interfering frequencies, so don't have a receiver on those frequencies.

Example: Transmitter A=525.000 MHz, transmitter B=550 MHz. Difference is 25 MHz. Don't have a system on 500.000 (525-25MHz) or on 575.000 (550+25MHz). Note that you can't screw up the math. Doing it wrong such as 525 +25 and 550-25 gives you the original starting frequencies. Also note that two systems don't have intermod problems. It takes two transmitters to generate garbage on a third frequency and you have to have a third system in order to have something to interfere with. TV stations do count as a transmitter, however, if the signal is strong at the receivers, i.e., equivalent to a wireless transmitter at 20 feet or less.

Well at least I managed to keep it to only one full page (if you use a small font).
Best Regards,
Larry Fisher
Lectrosonics

In general, wireless receivers have a local oscillator that mixes with the incoming RF signal to produce a lower frequency signal at the Intermediate Frequency (IF) that is then processed in the rest of the receiver. For instance, in a 180 MHz, CR187 receiver, the IF is at 21.4 MHz. This is much easier to filter and amplify than the transmitter's 180 MHz carrier. To produce this signal, the incoming 180 MHz is mixed with a signal at 158.6 MHz to produce a difference signal at 21.4 MHz.

The mixer can also mix a signal at 137.2 MHz to 21.4 MHz since the difference between the local oscillator at 158.6 and 137.2 MHz is also 21.4 MHz. Therefore there are two signals that can easily produce 21.4 MHz by mixing with 158.6 MHz: the desired 180 MHz and the "image" of 137.2 MHz. The mixer is equally sensitive to either signal and without a front end, a receiver is just as sensitive at the image frequency as it is at the desired frequency. The RF front end, which is tuned to pass 180 MHz but stop the image of 137.2 MHz prevents any response to the image. This is why the 187 series receivers have multiple helical resonators in the front end. The front end has to smack the image down 100 dB or more. This image frequency should be taken into consideration when doing frequency co-ordination, though with modern receivers, the image is almost completely rejected.

Here is a RAMPS posting.

To the Group:

This post is in reply to an earlier post about "hot" SM transmitters. The SM puts out no more heat than a UM400; it just does it in a smaller volume. Since, despite the non stick finish, I'd never had any luck cooking omelets with an SM, I thought I'd run some crude tests.

I tested the temperature rise of an SMa transmitter, lying with its backside on a pad of paper with the temperature probe between the unit and the pad of paper. The air was fairly still with only the usual office air conditioning running. The pad blocked some air circulation to the backside and seemed to be equivalent to a unit on a belt. After 3 hours the temperature rise was 16 degrees, from 76 to 92 F. The battery was an Eveready lithium disposable, though the battery should make no difference. An SMQa (250 mW) under the same test rose 22 degrees from Seventy-four to Ninety-six degrees F.

I then firmly taped an SM transmitter to my leg with summer dress pants between the back of the unit and my leg. The probe was positioned between the pants material and the transmitter back. The unit was then covered in 6 layers of a Lectrosonics' jacket (polyester fleece) spread out over a 12 inch by 16 inch area around my leg. I think it is safe to assume that all the heat went into my leg. After the temperature stabilized with the transmitter off at 94 degrees F, I turned it on and at 45 minutes it was up to 104.5 degrees. At 66 minutes it was at 104.7 degrees and was effectively stable. The temperature rise (10 F) was smaller than the "belt" example (16 F) even though there was no heat loss to the air since heat was carried off by the circulatory system.

The 8 hr metal contact standard for Europe is 43 C or 109.4 F. and this correlates well with the fact that I never detected the smell of burning Larry. Another web tidbit was that oxygen gas sensors for neonatal units use a small metal plate warmed to 45 C (113 F) to increase the gas exchange rate from the skin of a baby to the detecting unit. These can be operated for up to 8 hours.

The point here is that if the SM is not touching the skin, the heat rise exists but is moderate. If the surrounding air temperatures are very high or the unit is in the sun, the unit might be uncomfortable to touch but that is only if it is not in contact with a cooling device (human being) and if it isn't in contact then there isn't a problem. If it is in long term contact with a person, then the person will cool the unit so that there is only a small temperature increase. The skin on a healthy person is going to be less than 98 degrees to begin with or they've got much worse problems than the 0.75 Watt heat source of an SM.

In spite of all this, if the talent doesn't like the warmth, then a larger transmitter that spreads the heat out (UM400a) is one solution or a pouch or some thin foam between the transmitter and the talent.

As far as potential burns, the safe touch temperature for unpainted metal for 10 seconds is 132 F. Ten seconds is more than long enough to remove the offending object to a safe place. If they have a good arm, it can be removed 50 feet or so.

Best Regards,
Larry Fisher

A customer sent us a SONOSAX SX-BD1 and indeed we found excess noise with this unit when used with a UH400a. We are 99% sure this is due to common mode noise on the Sonosax outputs particularly when the 40 dB output attenuator is selected. (We didn't have a schematic and we didn't want to tear up the customer's unit). When used with a good mixer, the common mode noise was mostly canceled out but that was not true with the UH400a.

We found that grounding the XLR pin 3 output of the SONOSAX cable to pin 1 of the same XLR removed the common mode noise by forcing the output into an unbalanced mode that matched the input of the UH400a. This can be done either in the SONOSAX male XLR itself or inside a barrel adapter available from Lectrosonics P/N 21750. This adapter is normally used to reverse the XLR polarity and is wired as pin 2 to pin 3 and pin 3 to pin 2. You can undo this by switching the wires at just one end of the XLR. To fix the SONOSAX SX-BD1 problem, simply wire pin 3 to pin 1 at the male end.

To wire a positive ground lavaliere mic (some older TRAM's and Sony's) to the new servo input used on the SM series and the new UM400a, LMa and UM450 transmitters, use the following wiring arrangements. (Positive ground lavalieres are also known as negative bias lavalieres.)

This is the simpler "servo only" wiring and is not compatible with older Lectro transmitters (UM200, UM400, etc.)

• Pin 1 of the 5 pin TA5F is not connected.
• Pin 2 goes to the shield or ground wire of the mic.
• Pin 3 goes to the bias (audio) wire of the mic.
• Pin 4 is not connected.
• Pin 5 is connected to Pin 3 of the 5 pin TA5F.

The following is the compatible wiring and requires an external resistor but this wiring can be used with all Lectrosonics transmitters, old and new.

• Pin 1 of the 5 pin TA5F is connected to one end of a 2.7k resistor.
• Pin 2 goes to the shield or ground wire of the mic.
• Pin 3 goes to the bias (audio) wire of the mic and to the other end of the 2.7k resistor.
• Pin 4 is not connected.
• Pin 5 is not connected.

Other value resistors can be used in a pinch from 2k to 4k, including the 3.32k resistor that we provide in the 5 pin wiring kit.

We did some interesting RF measurements on a simulated two way bag system to see how much the bag transmitters would affect the bag receivers' sensitivity. A two way bag system will typically consist of multiple receivers to receive audio signals from the talent, a portable mixer to mix the audio and one or more transmitters to retransmit mixed audio to the video cameras. The immediate question is "If the receivers and transmitters are on different frequencies why should the transmitter reduce the sensitivity of the receiver?" One obvious answer is that the RF front end of the receiver is not a perfect filter and can let strong, nearby frequencies pass through and overload the first amplifier. In addition, transmitters do not produce a single sharp frequency but have some noise 5 Mhz or more from the carrier. The levels are very low but bag systems have antennas that are very close together. In the same way, the local oscillator in the receiver produces some noise many MHz away from the desired frequency and acts the same as having noise in the transmitter. Instead of trying to calculate all this stuff it is simpler to just measure a simulated system. Though these measurements were made on a UM200 transmitter and UCR201 and UCR210 receivers, the numbers should be comparable for the current UM400 or SM transmitters and the corresponding UCR401 and UCR411 receivers.

To see what kind of interfering levels would exist in a bag, we put a transmitter 12" (30cm) away from an antenna mounted on a power meter and measured an average signal of -5dBm (.5mW) from a transmitter with 20 dBm output (100 mW). This is a very strong signal to bleed into a receiver but will be very typical of a bag system with 12" of antenna separation. We used this level for the interfering transmitter for all the sensitivity tests. We then checked the receiver sensitivity with the transmitter off and then on and measured the reduction in receiver sensitivity. We then repeated the measurements for different frequency offsets between the transmitter and receiver. To simulate a bag system where the talent's transmitter is on 540 MHz and the bag is re-transmitting mixed audio to the camera on 550 MHz, we would inject a 550 MHz signal at -5dBm into a UCR210 receiver set at 540 MHz and see how much that affected the receiver's ability to pick up the desired 540 MHz signal. We attenuated a block 21 UM200C transmitter set at 550 MHz down to -5 dBm and combined it with a weak 540 MHz signal from a signal generator, set the receiver to 540 MHz and checked the sensitivity with the transmitter off and then on. With the transmitter off, the receiver had a normal sensitivity of -107 dBm for 30 dB SINAD. (Same as "signal to noise ratio" at these values) With the transmitter on, the sensitivity fell to -104.7 dBm for a decrease in sensitivity of 2.3 dB. The receiver was desensed by 2.3 dB. This means that with a real bag system having a 10 MHz offset in the two systems' frequencies and with the antennas 12" apart, the usable range from the talent to the bag would have been reduced to 77% of normal range. This is a pretty small reduction and surprised me. I thought it would be much worse. (There is no reduction in the distance from the bag to the camera since the receiver at the camera is not near a transmitter.) To simulate a worst case situation, we reduced the frequency separation to only 0.5 MHz with the talent transmitter and bag receiver still at 540 MHz and the bag transmitter now at 540.500 MHz. The desensing was now much worse at 20 dB. This would reduce the talent to bag range to 10% of normal and is a good reason to never operate with only 0.5 MHz frequency separation. Here's some more measured values for a UM200 UCR210 system. I'll put frequency and then resulting range as a percent and also in actual feet, assuming 300 feet for a normal system.

Here are the results of UM200 and UCR210 at 12 inches apart:

• 0.5 MHz separation results in 10% of normal range or 30 feet
• 1.0 MHz separation results in 20% of normal range or 60 feet
• 1.5 MHz separation results in 25% of normal range or 75 feet
• 3.0 MHz separation results in 32% of normal range or 96 feet
• 4.0 MHz separation results in 33% of normal range or 100 feet
• 6.0 MHz separation results in 66% of normal range or 200 feet
• 10 MHz separation results in 77% of normal range or 231 feet
• 20 MHz separation results in 81% of normal range or 243 feet

Some users have wondered how the UCR201 would perform in a bag even though this was not our intended use for the 201. This time the numbers are more in line with what I would guess, since the 201 is definitely weaker in this test.

Here are the results of UM200 and UCR201 at 12 inches apart:

• 0.5 MHz separation results in 0% of normal range or 0 feet
• 1.0 MHz separation results in 0% of normal range or 0 feet
• 1.5 MHz separation results in 12% of normal range or 36 feet
• 3.0 MHz separation results in 7% of normal range or 21 feet
• 4.0 MHz separation results in 14% of normal range or 42 feet
• 6.0 MHz separation results in 28% of normal range or 84 feet
• 10. MHz separation results in 50% of normal range or 150 feet
• 20. MHz separation results in 71% of normal range or 213 feet

As can be seen from comparing the numbers, the UCR201 needs twice the frequency separation before the ranges are comparable. The 3 MHz number looks funny but that's what we measured. The 50% of normal range is reached by the UCR210 at 5 MHz of frequency difference while the UCR201 needs 10 MHz of separation. I would recommend separation of at least one of our blocks (25 MHz) between the 201 receivers and transmitters in the bag. On the other hand, the UCR210 can operate inside the same block with a little care. The difference is due primarily to the tracking front end in the 210 and secondarily due to the higher power level of the first RF transistor in the front end of the 210. Once the signal is past the front end, both receivers are essentially the same.

A quick measurement with the antennas between the bag transmitter and receiver at 18 inches instead of 12 inches as above, showed a reduction in interference power of 5 dB. This is a huge change, is faster than the usual square of the distance rule and would allow you to more than double the range for some smaller frequency separations.

The results for all of this are:

1. Use a UCR211 or UCR411 for a bag system if possible rather than a 201 or 401.
2. Try to separate the antennas of the transmitters and receivers by 18 inches if possible.
3. Separate the frequencies by 5 MHz on a 211 or 411 system and by 10 MHz on a 201 or 401 system if possible.

More separation is better, particularly physical distance.

Here's the Lectro line on the SR receiver for bag use:

1. We realize that our customers are going to use the SR in their bags no matter what we say, just like they did with the UCR401 and frankly, they have had pretty good success with the UCR401.
2. The SR has a better front end than the 401 though not as strong as the 411. The input stage is more resistant to overload than the UCR401. However, the SR does not have front end tracking like the UCR411 (after all, which transmitter do you track?).
3. The customer should take the same care with the SR as with the UCR401. For instance, don't operate in the same block as the bag transmitters, try to keep 25 MHz or more of frequency separation between the transmitters in the bag and the SR receiver frequencies and try to have as much physical separation as possible. Inches can make a difference.
4. We are already planning to make a fourth bottom adapter for the SR that has two 6 foot audio cables and a 6 foot power cable that customers can cut to bag length and fit with their own custom connectors. Any of the different bottom adapters can be swapped in just a minute or so.
5. As with all our digital hybrid receivers, the audio performance is absolutely equal to the UCR411.

With a little care, the SR should make a fine bag system receiver

The mic level input on the UM400a has the new servo input with much more gain for low impedance signals than the UM400 with the older input. The increased gain improves performance with dynamic microphones. However, switching back and forth between the UM400 and UM400a transmitters will require constant readjustment of the gain setting if you are using the MC40 cable. To make life simpler we have a MC41 cable that will work with either transmitter type and matches the gain to within a few decibels. If you want to build your own, the wiring of the TA5F is as follows:

• pin 1 is ground
• pin 2 N/C
• pin 3 to a 3k resistor in series with the mic level wire.
• pin 4 jumped to pin 1 (sets the servo bias to zero)
• pin 5 N/C

This places resistance in the audio line and reduces the input to the UM400 by only a few decibels but reduces the UM400a by 20 dB and matches them very closely. The MC41 can be used with all the servo bias mics such as the SMa, SMQa, SMDa, UM400a, UM450, and LMa. It will also work well with any of the older style transmitters.

Here are some things we've found that will strengthen and protect the connection:

1. If the mic cable is much smaller than the strain relief boot, use a very flexible sleeve to increase the effective size of the cable. Allow about an inch of this sleeve to stick out of the TA5F boot. What you are looking for is a gentle curve in the cable when it is pulled at right angles to the TA5F boot. A stiff sleeve will cause a sharp bend right at the sleeve and effectively do nothing. We use a soft silicone shrink tubing here. Silicone model airplane gas line is very flexible and will work well. There is a self fusing silicone tape that can also be used. Taper the thickness around the cable as it comes out of the boot so that the cable makes that gentle arc as described above. If you start the wrapping at a point away from the connector and then work toward the pins, the loose end will be trapped in the connector. This will also give you a natural taper if you decrease the wind angle as you get close to the pin end. It is not necessary or desirable to put the tape under the crimp tabs. You are only trying to increase the bend radius where the tiny cable exits the TA5F boot. See http://www.filmtools.com/rescuetape1.html for the self fusing tape.
2. The metal strain relief crimp tabs in the TA5F are very critical to the longevity of your connection. There are two sets of tabs. The tabs closest to the solder pins should be used to crimp over the shield of the cable but not the outer insulation of the mic cable. This shield grounding will also give the best protection against RF from the transmitter. (The shield still needs to be soldered to pin 1). The second set of tabs, farther back, should crimp over the insulation of the cable. This requires fairly exact placement of the cable but you can and should have slack in the inner wires of the cable before they are soldered to the pins of the connector anyway. By assembling the connector this way, you have strain transferred to the outer insulation by one set of tabs and strain placed on the inner wires by the other set. The tabs should be firmly mashed into the insulation and cables but not so much as to puncture the wires. There is some judgment required here. There should be some deformation of the insulation but not enough to cause failure. You want to crimp the strain relief tabs enough so that a good pull on the main cable does not move the cable past the strain relief.
3. As mentioned in note 2 above, there should be some definite slack in the internal wires when the crimping and soldering is all done. The shield is normally a fairly stout wire and only a little slack is necessary but the other wires should have a definite "kink" so there is no chance of cable movement transferring force to the soft solder connection or to the small signal wires. All force should be taken up by the strain relief.
4. When stripping the wires and the insulation of the cable, use a good round hole wire stripper or even a thermal stripper so that the wires and the shield wires are not nicked. If they are nicked they will break easily when even slightly flexed. Don't use too small a stripper setting for the same reason. When in doubt, flex the wire(s) a few times; if strands break off, the wire has been nicked. You might as well just start over.
5. Don't overheat the shield. This is usually the largest wire in the bundle and can easily conduct enough heat to melt through the insulation of the audio and bias wires. Since all the wires have to be short to get them into the small connectors, heat can be conducted quite quickly up the wire, melting adjacent insulation. Pre-tinning the end of all the wires before attempting assembly will keep the ends from fraying and enable you to solder much faster. This will help prevent insulation melt through. Along the same lines, don't heat the metal strain relief in any way after the tabs are crimped down. We had one dealer that soldered a chip resistor to the strain relief tabs after they were crimped to the cable. This melted the insulation and caused many mic failures days and weeks after the connectors were wired.

If any readers have suggestions, post in RAMPS or email me at larryf@lectrosonics.com.

The RM2 does not have an LCD readout and can not set the frequency of operation of the SM transmitter. The RM2 uses a potentiometer for the gain control setting of the SM where the RM has the LCD readout. This means that you can only set the gain to an approximate value where as on the RM it can be set exactly. The speaker on the RM2 is not quite as loud as on the RM so the maximum range is a few inches less. The lithium battery on the RM2 is inserted into a clip on the circuit board and requires removing the back cover to replace, though it should last for years.

From the Shoeps web site: "The SCHOEPS CMR microphone amplifier allows any SCHOEPS ”Colette” series capsule (except the BLM 03 C active boundary layer capsule) to be used with pocket transmitters." 

In the email below is the TA5F wiring that works with our older transmitters such as the UM200 and the newer servo style inputs such as the UM400a or the SM transmitters.

Hi Charlie,
The CMR adapter is on its way back to you. The rewiring was pretty simple. The problem was that the CMR adapter doesn't like any DC voltage applied to the audio line (our pin 3). The SM servo unit applies 2 Volts to pin 3 when pin 4 is not wired to ground. With the voltage switched off, the mic sounded great but had too much gain in my opinion. We added a 1.5 k resistor in series with the audio line (our pin 3 again) and now the mic has about the same gain as a COS-11 when used with a UM400a or an SM. It still works with an older transmitter such as a UM400 or UM200 with a few dB more sensitivity than with the SM. If you should want more gain with an SM, reduce the size of the resistor; 500 Ohms = +6 dB.

So here's the final wiring for full compatibility:

• Schoeps shield to pin 1
• Pin 4 jumped to pin 1 (shuts off servo bias)
• Schoeps blue wire to pin 2 (+5 Volts)
• Schoeps white wire to a 1.5k resitor in series with pin 3 (audio in)

Thanks for providing the mic adapter.
Best Regards,
Larry Fisher
Lectrosonics

Here is the latest information as of 9 Jan 2008

The output level is adjusted in the DSP and not by an output attenuator. Since the original design was seen as a "venue" (auditorium, etc.) unit originally, lower output levels were not seen a necessary or a desirable feature and would add cost to the user. The problem for sound mixers is that if the level were to be adjusted lower in the DSP, then the output D to A noise would become a problem.

The other thing is that we didn't realize was that some of the pro gear had a hole in their ability to handle input levels. It amazes me that line levels are so high and mic levels so low that they don't overlap.

As a band aid, we have a cable that has an attenuator at the mixer end to reduce ground loop noise and can be set for -20, -30 and -40 dB of attenuation. 

See MCAXLRATTEN

Deviation is the measure of how far a frequency modulated RF carrier can change frequency in response to a signal such as audio. The amount of deviation is limited to a maximum value by regulatory agencies or it can be limited to a maximum bandwidth that the signal can occupy centered on the carrier frequency. For instance, the FCC specifies a 75 kHz peak deviation and a maximum occupied bandwidth of 200 kHz. 

FM is a form of spread spectrum modulation since the occupied bandwidth is greater than the bandwidth of the audio signal. For instance, at full modulation, a 1 kHz test tone broadcast by an AM station would occupy a little over 2 kHz of bandwidth but as wideband FM modulation it occupies more than 150 kHz of bandwidth. This additional occupied bandwidth has "process gain" just like any spread spectrum signal and suppresses interfering signals and noise. The greater the deviation, the greater the noise suppression effect. In general, 75 kHz deviation systems have over 3 dB better noise performance than 50 kHz systems, all other things being equal. With a compander in the system, the 3 dB RF link improvement due to the wider deviation sounds like a 6 dB improvement to the ear. There is a downside to the wider deviation and that is at very low levels of RF, the wider deviation loses its advantage over the narrower deviation systems and actually has a disadvantage. However, this occurs only when audio signal to noise ratios are at 12 dB or lower, which is effectively useless for wireless microphone purposes anyway.

See this link: Diversity types

Here is a URL that will take you to the Part 74 rules and regs regarding wireless microphone frequencies in UHF.

Please note that movie producers are defined in this as: "Motion picture producer. Motion picture producer refers to a person or organization engaged in the production or filming of motion pictures."

Part 74 then lists all the frequencies that auxiliary low powered stations can use , some of which are 470 to 806 and 944 to 952 MHz. Then movie producers are listed as one of the groups that can use low power auxiliary stations (wireless mics are one type station). All is well, right?

Then here is the gotcha:
"(d) Cable television operations, motion picture and television program producers may be authorized to operate low power auxiliary stations only in the bands allocated for TV broadcasting." 

TV broadcasting, however, is the UHF range of 470 to 806 MHz, some of which is gone or disappearing. Specifically, 944-952 MHz is not TV broadcast. Therefore, 944-952 is licensable only to broadcast entities but not to the groups in (d) above, including movie producers.

Frankly, the rules are confusing and seem to say that 944-952 is usable by movie production but then that section quoted above takes it away. 

Hope this makes the fog more palpable.

For a typical two wire lavaliere mic that specifies 5 Volts, the manufacturer is actually assuming that the transmitter has a 5 Volt supply in series with a bias resistor of 1k to 5k, depending on the brand of transmitter. The actual voltage at the mic will be 5 Volts minus the drop across the bias resistor. For example, a mic that is listed to draw 500 uAmp would produce a 2.5 Volt drop across a 5k bias resistor. The mic would only see 2.5 Volts (5V minus 2.5V drop). A different mic that pulled only 100 uA would see 4.5 Volts. So for most all transmitters, the voltage to the mic is all over the map. Generally the mics still work, because they actually can handle a wide range of voltages.

All the current Lectro transmitter models, such as the LMa, have a servo input that regulates the bias voltage to exactly 4 Volts under any condition of bias current. The voltage is set to 4 Volts by using the pin 2 to pin 4 wiring. This allows us to handle a wide range of microphones with any current draw with no concern about excessive voltage drop across the bias resistor and is unique to the Lectro transmitters. We chose 4 Volts because this was a typical design voltage and all the professional lavaliere mics we looked at worked very well at that voltage. The one exception is the tiny Countryman B6 and E6 models which require 2 Volts at high current. For the Countryman mics Pin 4 is NOT connected to Pin 2 and this sets the servo input to a regulated 2 Volts which is ideal for those lavaliere mics.

The new stainless steel SMA female connector is pressed into the aluminum front panel and is an interference fit. Any possible panel to barrel gaps are filled by us with a Loctite gap filler. The antenna wire itself has an O-ring in the nut assembly that seals the wire antenna to the nut of the male connector so any water running down the antenna cannot enter the connector. Then the only possible water entry is if the SR is upside down or angled down and water runs into the inverted nut, around the threads, into gaps between the center insulator and the barrel and down into the unit. You can seal against this improbable occurrence with a small drop (dab) of Vaseline or other petroleum jelly inside the SMA connector applied right on the white insulator. The Vaseline will prevent leakage by this path totally, even if the unit is dropped into water and if the SR (and attached camera) are submerged, you've got bigger problems anyway. The Vaseline does not affect the RF at all. If the Vaseline gets dusty when the SMA is removed, just clean it with a cloth or Q-tip and some clean Vaseline.

To test receivers under field conditions is a real pain and is rarely done properly, even at Lectro (!) in the past. To do proper receiver tests, all receivers must be on the same frequency, picking up ONE, I repeat ONE (!!!), transmitter and must use the same receive antennas with 2 two way splitters giving each receiver under test exactly the same signal at the same time. This removes transmitter antenna differences, RF interference differences and receiver antenna differences. Comparisons are rarely done this way and therefore always inexact (wrong).

Here are some things that we found that loused up our two transmitter, two receiver comparison tests big time:

1. Since transmitter antennas are rarely bent the same and therefore are at different distances from the body, the RF from the two transmitters is mismatched.
2. Since the transmitters are at different places on the body, reflections from objects in the area are aways different.
3. Since the transmitters are at different frequencies, the two points above are different as well as different RF noise from the environment. Keep in mind, RF that doesn't show up on the receiver's scan function can make large differences in reception when you are trying to receive weak signals, i.e., check range.
4. If the modulation (gain) of the two transmitters is not set exactly the same, one receiver can seem to have an advantage. This would show up more when comparing different brands.
5. The antenna placement of the receiver antennas causes just as many problems as the transmitter points made above. After all, the receiver antennas can't be in the same place simultaneously unless you use the splitter method described above.

This says that comparing two brands of receivers, transmitters, etc., is a crap shoot at worst and difficult at best. Of course, after being beaten unconscious by the sales people multiple times, I've learned to keep my mouth shut when a customer tells me how much better our equipment works than brand X. If you still want to compare two different pieces of wireless gear, I recommend multiple walks, switching frequencies multiple times, changing positions of the transmitters and moving the receivers around so that they occupy each others spots on different walks. You must also set the two transmitters up for correct or full modulation and then set the two receivers to have the same audio output level. To get valid results without the splitter setup and single transmitter described above, requires lots of tests and much walking.

Way back with the early CR185's (VHF compact receiver), when dinosaurs roamed the earth, we lost a few output stages to cheap mixers that had a single switch for all or nothing phantom power. We decided that we couldn't expect our customers to always be 100% perfect (we do expect 99.8%) and so we protected the output stages with bridge diodes. However, if the diodes were conducting and protecting the receiver, the signal was killed until the phantom power was removed. Some customers didn't realize what was happening, and returned perfectly good units to us for repair. That's when we realized that 99.8% of the customers didn't read the manual. Near perfection of another kind. My apologies to those two people who did read the manual. I also might mention here, that I never read a manual until I've got the device operating. It's an ego thing.

So, we added a resistive series circuit to protect against capacitive discharge, resistive bleeders to ground to reduce peak voltages and non-polar capacitors for protection against mis-wiring (pin 2 to pin 1, etc.). We have had no output stage failures since we put in all the gimmickry. We have lost an output stage or two when it appeared the outputs got wired up to 110Vac in some manner. This can usually be spotted by the large quantities of charred circuit board.

None of this is mentioned in an FAQ because we don't consider phantom power a problem. I will add this note, since I haven't done an FAQ in a long time and it is a good question to ask.

Best Regards,

Larry Fisher
Lectrosonics

Here's a reply to a customer who had an SMd that would only run for 6 hours on lithium batteries.

Hi Tom,
There have been four problems with the SMd battery setup. Here's the problems:

1. The screws that hold the contact leg to the board were not tightened properly at the factory. Resolved by giving the assemblers a torque screw-driver just for those screws.
2. The spring behind the contacts was not cut properly. The "spring" is a piece of silicone tubing which is very stable and long lived but was being cut by hand, sometimes at an angle. This can be detected by looking at the battery height above the case with the door open. Commonly one battery would stand proud of the case and the offending battery would be 0.050" below the case edge. This was resolved by cutting the springs with a machined fixture that cuts the springs squarely and at a constant length.
3. The battery polarity protector inside the case was a hair too thick and kept some batteries with short nipples from touching the contact. The offending battery just happened to be the Eveready lithium batteries that we ship with the product. This was resolved by molding a thinner polarity protector.
4. The diameter of the machined ring on the inner surface of the battery door, also just happened to fit inside a circular depression on Eveready batteries of all types. This reduced the battery pressure and could cause a loss of contact if the SM's were shaken or dropped. This would cause the unit to shut down.

All of these are considered to be our design fault and come under the extended "we blew it" warranty and will be fixed at no charge or automatically upgraded if a unit comes in for other things. Also, these fixes have been in the units for some time. However, not all were done at once since we didn't find them all at once and they didn't fail on most units.

As far as battery life, the lower capacity "Advanced" lithium batteries run an SMd for 12.7 hours. We will have the "Ultimate" lithium numbers in a day or so but they should be around the 15 hour number.

Best Regards,
Larry Fisher
Lectrosonics

For a location sound mixer making film or TV content, there have been no changes in the rules. You qualify as a Part 74 user.

You have always been required to have a Part 74 license. This has been the case for decades and has not changed. The new rules exempt NON Part 74 users as long as they keep
their power down under 50mW. These users include churches, education, government, bands etc.

Again, for you as a location sound mixer, the rules remains exactly as they have been for decades. Nothing has changed except we can't use frequencies above 700Mhz anymore. 

If you want to get your license, you can get the form FCC601 from:

http://www.fcc.gov/Forms/Form601/601.html

You will also need Schedule H and forms 159 and 160 (also found on the above link). 
(four forms in total)

Prior to the new rules, churches, educators, musicians, or theater (anyone not involved in motion picture or TV production) were not authorized users and could not get a license because they were not Part 74 users. Under the new rules, you are exempted as long as you use less than 50mW. This might change because the rules are preliminary and not yet finalized. We will keep you posted.

You cannot qualify for a higher powered transmitter under the new rules and cannot get a license.

There have been NO changes in who qualifies for wireless microphone licenses and how much power their transmitter can have. Only those involved in television broadcast - licensed broadcasters, networks, and people making content for television or motion picture production - can get a license for wireless microphones. They can use up to 250mW. 

People who are NOT directly involved in production for broadcast and motion pictures cannot qualify for a license and, under the new rules, can use wireless microphones
without a license as long as they are 50mW or lower. 

They cannot get a lioense - exactly the same as before the new rules. The only difference now is the exemption for lower powered transmitters. 

If you are involved in TV or movies and want to get your license, you can get the form FCC601 from:

http://www.fcc.gov/Forms/Form601/601.html

You will also need Schedule H and forms 159 and 160 (also found on the above link). 
(four forms in total)

Yes. No if's, and's, or but's about it. This is not new - everyone was supposed to vacate that band LAST June in 2009. The FCC is getting serious about it now. The loss of this spectrum has been in the works since 1995. If you have gear in that range from Lectrosonics and it is under 5 years old, it may qualify for a frequency change program. It will not be free but it will be less costly than replacing all that gear - even with other brands who offer "rebates" or "trade-ins". 

The SNA 600 will operate well below 500 MHz. It is marked 500MHz because we didn't make receivers below 500 MHz until the great digital changeover of all the TV stations in 2009. The SNA-600 measures lower than a 2:1 SWR (Standing Wave Ratio) from 440 MHz to 600 MHz when the antenna arms are fully extended. Less than 2:1 SWR is considered a standard antenna range measurement. The marked range for fully extended is only down to 500 MHz but the antenna will operate much lower than that.

An HM plug on transmitter was tested with three types of AA batteries. Alkaline batteries ran for 5 hours with a dynamic mic plugged on. Alkaline batteries lasted for 3.5 hours with a Sanken CS1 plugged on with the HM providing the 48 Volt phantom power. New Sanyo rechargeable NiMh batteries, measured at 2500 mAh capacity, ran the HM for 9.25 hours with the dynamic and 7 hours with the CS1. Lithium Eveready AA's were outstanding at 16 hours with the dynamic mic and 12.75 hours with the CS1. Your mileage may vary and in the case of the alkalines, will depend on temperature.

Try using the 21750 phase reverser. URL below.

Click here for #21750 page

This will swap pins 2 and 3 on the XLR and the Oktava should now work. You can also switch the wires on the XLR in the microphone. In either case you will need to switch the phase polarity in the Lectrosonics receiver if you are using multiple mics picking up the same audio source to prevent phasing artifacts.

NiMh and NiCd voltages change very little from 90% charge to 5% charge. The change is smaller than the difference between a new and used NiMh. The change is smaller than a cold and hot NiMh. The change is even smaller than the voltage drop across a slightly dirty battery contact. Because of all these reasons, there is no way of measuring the battery voltage and determining the remaining charge. The LED's are not a good indicator of the battery life of a NiMh. A fully charged battery can indicate red and a nearly discharged battery can indicate green, though that's rare. What the LED's can tell you is when the LEDs are off, the battery is dead. 

If the battery has the capacity to hold a full charge, then the timer is dependable. A good AA NiMh will run an SM at 100 mW for 4 hours. When the timer reaches 3 hours, it is probably time to start thinking of replacing the battery. If you must have battery life readouts and the timer is not satisfactory, then alkaline or lithium disposable batteries are your choice. You can test batteries in the transmitters by letting them run down and stop the receiver timer. This will give you a good idea of what your particular brand of batteries can do. If a battery is low capacity, discard it. It's not worth the danger of accidentally ending up in a high value situation.
Best Regards,
Larry Fisher
Lectrosonics

The biggest difference between the SRa and SRb receivers is the use of a six layer printed circuit board rather than the original four layer board. This single change reduced internal interference (birdies) to the point that gain could be reduced in multiple amplifiers in the RF and IF stages. In fact, one IF stage was simply removed since it was now unnecessary to the design. In spite of the reduced gain, the sensitivity of the receiver improved overall. As you would expect, intermodulation and overload performance improved greatly with the reduced gain. This improved board was first used by exchanging customers' boards on Block 606 in England as they were having the most problems.

With what we learned on the Block 606 changes, we added the same general changes and the six layer board to all the SRa units without saying much about it for several months. These improved units seemed to work very well and tested much better in production. While shipping the improved units, we went back to the SRa RF board and spent several months re-matching every amplifier and filter in the RF and IF sections. This was necessary because of all the other changes that had already been made. Many small improvements were made at each stage leading to fairly impressive additional improvements overall. One of which was an improvement in sensitivity of more than 6 dB. The last major change was to relocate the two IF filters. We found an optimum location by making up 5 different prototype boards, with the IF filters located in various trial positions with different shields and ground plane configurations. The final result had one IF filter on the top of the PC board and one underneath with a solid ground plane and our usual shielding. This simple change improved out of band selectivity by a huge 30 decibels.

With the improved sensitivity, better out of band rejection, improved intermodulation performance, and the elimination of internal spurious responses we decided to advance the model designator from "a" to "b", thus the SRb. As a service to our customers with SRa units, we offered an upgrade for the difference in retail price between the SRa and the newer SRb. The change in model designator also indicates that we think this is a major improvement in the SR receiver and it encourages our customers to get the upgrade.

Currently, there is not a LecNet 2 version of the TH3A. You can, however, interface the TH3A with the DM series but you must reserve one audio input and one audio output on the DM. Use the AUX IN and AUX OUT ports on the TH3A tied to an audio output and audio input respectively on the DM mixer.
The TH4 (LecNet 2) is in development and will interface with the DM series mixers via the Digital interface. It will not require an audio input or output from the DM mixers to connect.

Yes, the protocol for controlling the Venue wireless is easy to use and we are developing modules you can include in your programming. You can adjust levels, change frequencies, change operating modes (such as type of diversity), check transmitter battery levels and many other functions. You are not limited to just these two control systems. Because we have transport neutral protocol, there are many ways (including HTML pages) to control LecNet 2 devices.

Contact our control systems specialist, Frank Gonzales for assistance.

The AM series (also called LecNet) used a communications protocol developed before the dominance of third party control systems. Primarily intended for our software to control AM16/12, AM8, DSP4/4 etc, it was a hexidecimal programming code designed for use by computer programmers. The PT3 is a protocol translator that can easily convert AMX commands into a string of LecNet commands. Up to 92 AMX commands, (pulse, level, or channel) can be associated wih a LecNet command or string. The PT3 is not required, it is simply a programming aid designed to make AMX code writing easier. You CAN control with AMX directly but the code will be more complex. The PT3 cannot help with Crestron systems. The PT3 is NOT needed for the DM LecNet 2 series products.

Input gain is the same as input gain or trim on a standard mixer. It is the best control for optimizing your gainstaging and signal to noise ratio in the system. This is where you would set your mic trims during initial system setup. Once set, the input gain should be left alone.

RP gain (RP = remote panel or Rear panel) is an attenuator only gain control. It acts on the level for the input by attenuating the signal from your (now optimized) input gain. This is your best choice for remote control (hence the name RP). By controlling gain here, the end user cannot disrupt the gain structure of your system, yet have full control over the inputs assigned to that control.

For example, you can set the gain for your system so you are about 6dB below feedback using both input gain and output gain. With RP gain controls (input, output ot both) your end user can turn things down if they get too loud but never be able to turn them up into ear-bleeding feedback.

No. We took a different approach. Echo cancellation has no benefit for the local room in a video or audio conferencing installation. The only beneficiary is the room at the other end. Echo cancelling is needed to elminate the acoustical coupling of the loudspeakers in the room and the microphone. The only signal that needs to have echo cancellation is the OUTgoing channel to the far end. Then, the LecNet2 series takes a two pronged approach to eliminating that echo. When designing the DM series we included two extra data channels internally for use with the DMTH4 conferencing interface. These channels allow easy matrixing of the incoming and outgoing signals.

First, we integrate the incoming signal from the far side into our automixer algorithm. When the far side is speaking, that signal takes a priority (autoskewing) for the input and the mixer doesn't "open" the microphones in the room in response to that amplified signal. This action suppresses the generation of an "echo" back to the far side. The microphones however are never fully off, so some signal will stil get by.

That brings in the second tier of prevention, the internal echo canceller in the DMTH4. This echo canceller cleans up the remaining echo that might get past the automixer and gives a clena signal back to the far side.

This approach has two benefits. One is cost savings. By not having to place a DSP system at every input, we reduce material cost of the system. But more importantly, the latency (or delay) caused by per input cancellation is reduced. The DM1624 has a latency of only 2ms. Designs that have individual channel echo cancellers typically have latency of about 19ms. That delay can be detectable in a room where someone might be seated 20 feet froma speaker. 20ms of acoutical delay plus 19ms of latency yields a total delay of 39ms which is a noticeable delay in the the signal. The DM system will have minimal latency while effectively decoupling the speakers from the microphones for a successful conferencing system.

Yes, the DM has a extremely short latency of only 2ms, primarily caused by the digital to audio (D/A) or audio to digital (A/D) converters. If you stack 10 units, you will only see an slight increase to 3ms because the DANI bus requires no additional D/A or A/D conversions.

This delay is the equivalent of moving the speaker only 2 feet farther away.

Latency is important because is can add up in a large system. If you have a large delay (say 19ms) and then you add additional delays (both acoustical and electronic) those delays can become audible to the system operators and performers.

The software for the DM series mixers has a macro recorder. Since macros are used to make changes to the state of the system, we advise first programming the unit for its startup configuration. Set up all your microphone inputs, your crosspoints and output levels etc. Once you have the unit set the way you need it when you power up the installation, save it to a preset. Now you are ready to record macros.

Select "Macros" from the top tool bar in the DM Control panel software. Click on "Start Macro Recorder". Then make the changes you want in your system configuration. (Change input gains, mute crosspoints, engage the noise genrrator, etc.) The macro recorder will record any changes in state to the system while ignoring actions that are irrelevant such as changing tabs in the software.

HINT! You may need to make a change in gain at an input, a crosspoint, or a output. During a typical installation, this may be difficult to determine in advance until you have the system up and running. Use the "Pause" function to temporarily stop recording. Set the gain in question until you have found the right level. Now, before resuming recording the macro, take that gain down one dB. Click "Start Macro Recorder" and set the value back up one dB. That will safely record your new level without recording all the intermediate changes in gain you made while finding that correct level.

The macro recorder will record up to 64 commands. After you have finished making your system changes, click on "Stop Macro Recorder". The Macro Editor screen will come up so you can enter the name of the macro and review the commands you recorded. You can also edit the macro from within this screen. Hit "Done" to exit. The macro will be recorded within the DM1624 and you can also save the macro to a file on yor computer for reference or modification later.

NO! In fact, the PT3 cannot even communicate with the DM series.

The new LecNet2 Protocol is so easy to learn there is no need for a "Protocol Translator". The old LecNet language was designed before there was third party control and meant primarily for commincations with our software during setup. 

LecNet2 is designed for easy AMX or CRESTRON control. 

Example - the old code for controlling an AM1612 for the input gain for channel one to -10dB was:

"139,wait 40 ,1E,1,0,0,0,22"

The NEW code for any LecNet2 unit is 
ingn(1)=-10 (followed by a carraige return)

Pretty easy isn't it? To inquire about the gain on input one is simply
ingn(1)? (Followed by a carraige return)

Controlling LecNet2 units with AMX or CRestron is MUCH easier now so the PT3 is no longer needed.

Yes. When calculating the number of conductors you will need for a remote (or rear) panel control using pots (10K linear), switches and LED's, count up the number of devices and add two (one for ground and one for voltage). Example - if you will have 5 pots for level controls, 5 LED's and 5 switches, you will need 17 conductors. Five conductors for the wipers on the pots, 5 for each LED negative lead and 5 for each switch. You can make the voltage common to the CW contacts on the pots and the LED's positive lead (don't forget the 380Ohm resistor). The ground will be common to the CCW lead on the pots and the secod lead on the switches.

Common means one conductor leading to each component in parallel.

Phantom mode allows you to allow a microphone (or input) to participate in the automix function of an output WITHOUT the audio actually being a part of the mix. Each output of the DM series mixers acts as a completely independent automixer. Activity on one output has no effect on the automixing going to a different output. A microphone that is not sharing an output with a second mic cannot be affected by that secind mic. In most cases this is a good thing. But sometimes, you want interaction without the actual audio mixing. Phantom mode allows that.

Let's say you have a microphone on input 7 that is going to be recorded all by itself on output 8 in a multi-track recorder. If it is all alone on that output matrix, it will always be on. If someone on a different microphone speaks and microphone 7 "hears" that amplified sound, that other mic's audio will be picked up by mic 7 and recorded even though you don't want that signal on that track. BY putting all the other microphones in the room on the matrix to output 8 and setting them to PHANTOM mode, they will be active participants in the automix algorthm but their audio will not actually be sent to output 8. They are "phantom mics". So, when someone talks at mic 5, they won't be recorded and the automixer will prevent mic 7 from turning on for that signal. Our autoskewing part of the patented mix algorithm will prevent the same source from mixing from two inputs.

No. Hyperterminal does not use the extended ASCII character set which is required for proper communications with the DM series. When you loaded the software that came with your DM series unit, you received a program called LecNet2 Command Terminal. You can use this utility to test command strings, and check your system. In Windows, press Start, Programs, LECNET2, and finally LecNet2 Command Terminal. Select "Connect" at the top menu, choose the method of communications and the software will sign on to the unit - make sure your cables are attached first.

Yes. After attaching the button to the unit, go to the REAR PANEL CTRL tab in the software. Select the programmable input that the button is attached to.

(Did you forget which button? Press the actual button and watch the virtual LED's in the Programmable Inputs. The one that lights up when you pressed the button is the programmable input your button is actually attached to.)

In the FUNCTION window select the down arrow, highlight "RUN MACROS on CLOSE/OPEN". Select the macro you want to run when when the contact closes and then the one you want when the switch opens again.

When you enter the matrix page of the DM series mixers setup software, you will see that each cross point can be set in one of five different modes.

NOM stands for Number of Open Microphones. An open microphone is a mic that is at full or nearly full gain. For every doubling of the number of open microphones you lose 3dB of potential acoustical gain (PAG). The purpose of the automixer is to keep the total gain of the system constant. It does so by monitoring the number of open mics and adjusting the total gain downward as more mics become active. Any signal stream that has multiple microphones is considered a NOM Bus.

On the DM series, each output is setup as a separate NOM bus. When you set the cross point assignments (which inputs go to which outputs), you can choose the behavior regarding the NOM bus.

Auto Mode is full auto mixing mode where the microphone - AT THAT CROSS POINT- both contributes to and reacts to the NOM bus. Example - all normal microphones should be set to AUTO at each cross point. If other microphones that are active to this output, then all cross points set as AUTO (on that output) will react by lowering their gain in the system.

When you enter the matrix page of the DM series mixers setup software, you will see that each cross point can be set in one of five different modes.

NOM stands for Number of Open Microphones. An open microphone is a mic that is at full or nearly full gain. For every doubling of the number of open microphones you lose 3dB of potential acoustical gain (PAG). The purpose of the automixer is to keep the total gain of the system constant. It does so by monitoring the number of open mics and adjusting the total gain downward as more mics become active. Any signal stream that has multiple microphones is considered a NOM Bus.

On the DM series, each output is setup as a separate NOM bus. When you set the cross point assignments (which inputs go to which outputs), you can choose the behavior regarding the NOM bus.

DIRECT means the cross point is always on and neither reacts nor contributes to the NOM bus. It will not affect microphones or other inputs and it will not be affected by activity at other inputs. This is a good choice for multi-media inputs such as CD players or DVD's.

When you enter the matrix page of the DM series mixers setup software, you will see that each cross point can be set in one of five different modes.

NOM stands for Number of Open Microphones. An open microphone is a mic that is at full or nearly full gain. For every doubling of the number of open microphones you lose 3dB of potential acoustical gain (PAG). The purpose of the automixer is to keep the total gain of the system constant. It does so by monitoring the number of open mics and adjusting the total gain downward as more mics become active. Any signal stream that has multiple microphones is considered a NOM Bus.

On the DM series, each output is setup as a separate NOM bus. When you set the cross point assignments (which inputs go to which outputs), you can choose the behavior regarding the NOM bus.

In OVERRIDE mode, the cross point contributes heavily (an additional 12dB) to the NOM bus. This helps the OVERRIDE microphone overwhelm and suppress other microphones in AUTOMIX mode. EXAMPLE - Use OVERRIDE for an emergency paging microphone or for the Chairman of the Board.

This was a general question from the RAMPS group about various wireless transmitters generating low frequency noises when struck.

In general, there is mechanical coupling from the case into the inductors in the main oscillator in the transmitter. A thump on the case moves or bends the inductor, changes the inductance value by a tiny amount and changes the frequency of oscillation. Since a changing frequency is just FM, the FM receiver picks it up as a low frequency thump. There are various ways of reducing the mechanical sensitivity. Most involve very rigid coil assemblies such as inductors wound on ceramic forms. In our case, we use solid quarter wave ceramic resonators.

The cutest trick I've seen, was a (brand) unit that used a miniature Teflon insulated coaxial line as a resonator. They wound the coax stripped off the outer insulation in a tight cylindrical coil with about 6 turns. The entire shielded coax coil was then soldered on the outside into a solid mass. This made a nice rigid assembly with the center conductor acting as the inductive element since a short coax line with one end shorted looks like an inductor.

The other way to generate a thump is to use a capacitor in the audio circuity that is sensitive to mechanical stress. The wrong kind of ceramic capacitor with DC voltage on it can really generate a lot of voltage when stressed. NPO ceramic capacitor types are as good as most film caps or tantalums but X5R types are bad and Y5Z are horrible. NPO's have the least capacity for a given size and the other types have 5 to 50 times more capacity in a given size and that's why they exist. I tried 50 Volt Y5Z type capacitors in the design of the 48 Volt phantom supply for the UH200C. You could get about as much audio talking into the transmitter PC board as you could using a microphone. Fortunately some small 50 Volt tantalums came on the market that would fit in the same space and saved my bacon. I knew the problem existed, but the severity surprised me.

My advice is to whack the case of a transmitter with both your finger and with a pencil sized object. If you know how a transmitter is going to react to mechanical shock, you can prepare for it.

On the subject of mechanical stress and audio, try the same thing with your electret mic cables. Some are much worse than others. If you tap the cable close to the mic (6") you will get mechanical noise transmitted directly to the mic element. In the middle of the cable, it is due to flexing of the mic cable. Phantom powered are sensitive to this since there is DC voltage on the cable and flexing the cable changes the dimensions and the capacitance of the cable. The pro mic manufactures have taken this into consideration in the choice of cable.

In the Menu (top) banner of the Control Panel software, you will find a choice called "DEVICE SETUP". Click once and you will see in the drop down menu the choice "Phantom mix mode option". Click once - that will change the unit to phantom mode and allow cross points to be setup as Phantom (no audio) cross points. This action, in the DM1612, DM812 and DM1624 disables Background mode. The DM84 has all five modes available at all times. See AUTO, DIRECT, BACKGROUND and OVERRIDE.

When you enter the matrix page of the DM series mixers setup software, you will see that each cross point can be set in one of five different modes.

NOM stands for Number of Open Microphones. An open microphone is a mic that is at full or nearly full gain. For every doubling of the number of open microphones you lose 3dB of potential acoustical gain (PAG). The purpose of the automixer is to keep the total gain of the system constant. It does so by monitoring the number of open mics and adjusting the total gain downward as more mics become active. Any signal stream that has multiple microphones is considered a NOM Bus.

On the DM series, each output is setup as a separate NOM bus. When you set the cross point assignments (which inputs go to which outputs), you can choose the behavior regarding the NOM bus.

PHANTOM mode is just like AUTO as far as NOM action is concerned. It both contributes and reacts to NOM. The difference is that input at that output is not contributing any audio - no level at all. What is it used for? Let's say you want to record the judges microphone on track one of a multi-track recorder. If you send ONLY his microphone to that output (enable only the cross point in regular AUTO mode) then his microphone will be on all the time. The recording will pick up not only his voice but also the amplified voices from every other microphone in the system. But if you add all those microphones to that output by engaging their cross points in PHANTOM mode, then the auto mixing action will prevent the judge's mic from opening for every little noise in the system. Only when the judge speaks will the recording get a significant signal (the judge's voice). This is because the presence of the other mics in the room on the NOM bus will keep the judges mic down in level unless the judges himself actually speaks. Because they are PHANTOM mics on that output, you never hear their audio.

When you enter the matrix page of the DM series mixers setup software, you will see that each cross point can be set in one of five different modes.

NOM stands for Number of Open Microphones. An open microphone is a mic that is at full or nearly full gain. For every doubling of the number of open microphones you lose 3dB of potential acoustical gain (PAG). The purpose of the automixer is to keep the total gain of the system constant. It does so by monitoring the number of open mics and adjusting the total gain downward as more mics become active. Any signal stream that has multiple microphones is considered a NOM Bus.

On the DM series, each output is setup as a separate NOM bus. When you set the cross point assignments (which inputs go to which outputs), you can choose the behavior regarding the NOM bus.

PHANTOM mode is just like AUTO as far as NOM action is concerned. It both contributes and reacts to NOM. The difference is that input at that output is not contributing any audio - no level at all. What is it used for? Let's say you want to record the judges microphone on track one of a multi-track recorder. If you send ONLY his microphone to that output (enable only the cross point in regular AUTO mode) then his microphone will be on all the time. The recording will pick up not only his voice but also the amplified voices from every other microphone in the system. But if you add all those microphones to that output by engaging their cross points in PHANTOM mode, then the auto mixing action will prevent the judge's mic from opening for every little noise in the system. Only when the judge speaks will the recording get a significant signal (the judge's voice). This is because the presence of the other mics in the room on the NOM bus will keep the judges mic down in level unless the judges himself actually speaks. Because they are PHANTOM mics on that output, you never hear their audio.

First, let's assume that we will be setting this unit up with a DM series mixer. This means you will have a choice of 14 to 26 mixes to which you can add the incoming audio. Determine which outputs on the DM mixer will have telephone and codec audio. For illustration sakes, lets say you want the incoming audio to come out of outputs 1, 2 3, and 4.

In the Matrix page of the DMTH4, pick the MAIN MIX MATRIX tab. On the row marked TEL, right click the cross point in column 1 and left click on "Engage cross point gain (0db gain)". Do the same for columns 2, 3 and 4 (in the first row only!). This routes the telephone signal into outputs 1, 2, 3 and 4 of the mixers. Now your room can hear the incoming telephone signal.

Repeat on the second row for the CODEC row. This brings the incoming videoconferencing audio into the mix.

Now you need to route the microphones in the room to the outgoing phone lines. When you are setting up the DM mixer, you will have selected a mix bus for sending to the far side. That could be the Expansion Mix Matrix 1 or 2 or it could be one of the other mixes set up in the mixer. It is not important which you select except you MUST NEVER select an output which you have already used to bring in the INCOMING signal.

In our example that means outputs 1, 2, 3 or 4. We CANNOT select those outputs because that would cause an internal feedback loop.

To route the local room microphones to go OUT via the DMTH4, you must change to the OUT SOURCE page in the Control Panel software. In the TEL box you will see three radio buttons for Matrix, Pink Noise and 1 kHz tone. Select Matrix. Then click on the pull down menu arrow and select from the choices. Remember that, in this example, Final mix 1, 2, 3 or 4 are off limits. Those mixes already carry the telephone signal and would be a feedback loop. In this case select Exp Final Mix 1 (The default).

The codec is defaulted to Exp Final Mix 2. Let's use that.

Now, we must save and exit the software for the DMTH4 and set up the DM Mixer for sending the local microphones to the DMTH4. Open the Control Panel software for the mixer and connect. Select the Matrix page, and then click on the bottom tab labeled Expansion Mix Matrix. Select the cross points for the inputs for Exp Mix 1 for all mics you want on the telephone and Expansion mix bus 2 for the Codec. Save and exit.

Do you want the Codec audible on the phone and the telephone audible to the codec (a three way bridge between the room, the phone and the video system)? Go back to the DMTH4 software and go to the matrix page. Click on the Expansion Mix Matrix and send the Tel to Expansion bus 2 and the codec to Expansion bus 1. n Congratulations, you have just set up a three way bridging system.

Yes, when configuring an input logic pin with no connected pots on the unit for Analog RP Gain control, the designated pin's virual LED will light up. Once you have correctly connected the potentiometer and cycled it up/down once, the "LED" will coorectly track activity. Once it "sets", the "LED" in the software will come on when you have turned to control to about the halfway point.

Don't confuse the virtual LED in the control panel software with an LED connected to the unit as an indicator (such as the LED on the RCWVLS).

The RCWVLS, when correctly wired, will show activity on the Rear Panel control page and the actual LED on the volume control will be ON all the time.

Gain on the inputs should be set so that, when you are getting a normal signal from the source (Mic, CD, etc), the gauge will showing level around the 0 point on the scale. Using a test signal or acoustical source for the mics is very helpful since microphones can vary as much as 10dB even within the same model and brand.

Good starting points for input gain are:
Boundary mic = +40dB
Gooseneck mics = +35
Ceiling mics (Shame on you!)= +55
CD's and other similar consumer devices = +10
Professional line level signals (such as pro consoles) = 0

First, a quick test. Route all the microphone in the room to the far side only. Have no local amplification of the microphones within the room. If you are using a DM series automixer, this means turning off the crosspoints feeding local mics to local speakers. Then scratch (with your fingernail) the windscreens of the microphones. Does the scratching sound come out of the local speakers (even though you don't have them routed that way)? If the answer is yes, the phone line has a serious impedance mismatch cuased by a termination problem or short. The problem is NOT in the DMTH4 but in the phone line. Insist that the phone service provider have the line checked and re-pulled if necessary.

The DMTH4 is fairly robust in handling various phone lines and analog emulators on PBX systems. But a serious impedance mismatch in the hybrid bridge (on the phone system side) can cause an echo (or reflection) in the phone line back to your incoming side. This is the same as having an impedance mismatch in a video line that causes "ghosts" in your images - these ghosts are reflected signals.

A good incoming signal level is essential for the DM series automixers. Use the following guidelines -

Handheld microphones (held close to the mouth) = +35
Gooseneck microphones (like a podium mic) = +45
Boundary Mics on a table or desk = +50
Ceiling mics (shame on you!) = +60

CD, DVD or other conusmer multimedia players = +10
Professional Audio gear - (like sound consoles) = 0 to -8

A good incoming signal level is essential for the DM series automixers. Use the following guidelines -

Handheld microphones (held close to the mouth) = +35
Gooseneck microphones (like a podium mic) = +45
Boundary Mics on a table or desk = +50
Ceiling mics (shame on you!) = +60

CD, DVD or other conusmer multimedia players = +10
Professional Audio gear - (like sound consoles) = 0 to -8

There are several things to check - the most common errors are (in order)

1 - Are you using the correct cable? The orange/red cable packed with the DM series is for AMX and Crestron control systems. If you are using the black cable it willnot properly configure. For Crestron or AMX, the pin configuration is simple if you wish to make a custom cable. From the DB9 connector to the TRS connector the pin out is Pin2(DB9) to Tip(TRS), Pin 3 (DB9) to Ring (TRS), Pin 5 (DB9) to Sleeve (TRS ground).

2 - Do you have a separate serial port for each DM unit? Unlike the AM series, each unit requires a separate control port - one RS232 for each unit.

3 - Is your Baud rate correct? Try setting the Baud rate at 19,200 for both the control system port and on the DM units Faster speeds may result in dropped bits and increased errors. Unlike our setup software, AMX and Crestron systems are "fire and forget" and if the signal is not acknowledged properly, they won't try again.

4 - Is your syntax correct? Command lines should always be followed by a Carriage return.

The crosspoint delay concept comes up rarely. We have examined this concept in great detail. There are several flaws in the attempting crosspoint delays in a courtroom or conference room system.

1 - The concept of applying a separate delay for each cross point for zoned speaker systems makes an assumption of the NEED for delay in such as system. In the majority of most zoned systems however, there really is no need as the listener is usually sitting in the amplified zone of a single speaker and does not get significant contribution to their hearing levels from other speaker zones or even the original source. Indeed, if every speaker zone was audible to the single listener, the intelligibility factor would be greatly reduced. Every intelligibility calculation in the audio and acosutical world takes into account the "N" Factor or number of sources carrying the same signal that is audible to the listener. If zoned speaker systems had every speaker audible to each zone, the N factor would drive intelligibility into the ground - regardless of delay or not.

We have thousands of mix-minus zoned systems installed throughout the world and cross point delays have not been required for any of these. In the few facility designs I have known where this was specified, the spec for cross point delays was eventually tossed out as impractical.

Individual channel out delays would not effective in these type of installations. In almost every zoned installation these units have been used in, the delays are not implemented. The delays have been used only in the typical installation where synchronization of audio to make up for long distances (such as along rectangular performance hall) is required.

2 - Calculating each of 224 or 418 delays for every possible combination is a huge task with little return on the invested time.

3 - The assumption that by eliminating the delay at the inputs and outputs would be the same or a bit more DSP load is erroneous. The DM1612 for example has 28 delay functions built in now. If we engaged cross point delays, the system would require 224 crosspoints -a huge jump in the overhead. The DM1624 would jump from 40 delays to 416 delays.

Example - there are some flexible architecture DSP units on the market that allow crosspoint delays. Try setting one up with 224 crosspoint delays and see how much processing power you have left. Their intent is for a FEW crosspoint delays, not for every crosspoint. Ours is an optimized architecture which means the function, whether used or not, is available. It is vastly more efficient from a code space point of view (and manufacturing cost) and can deliver more capability per MB than a flexible scheme.

Crosspoint delay - while effective in certain limited applications - would be VERY counterproductive in the typical<

There is nothing wrong with the unit. The USB-- Serial indicator shows that the wrong driver has been incorrectly assigned by Windows to the unit. This is a problem in the USB protocol in Windows. When connecting a new DM to your computer for the first time, never allow Windows to automatically install the device driver - always install it manually. There is a fault in the Windows protocol which, when it sees the USB chip set, tries to install an incorrect driver.

The solution can be a bit complex. Follow these steps to correct the problem.

1. Connect to the serial port of the unit using the black serial cable. If you do not have this cable, you can build a new one as follows
From the TRS(stereo) 3.5mm connector to DB9
Tip to pin 2
Ring to Pin 3
Sleeve (Gnd) to Pin 5
jumper pin 4 to 6
jumper pin 7 to 8

2. Start the LecNet Command Terminal software. Connect via the serial port.

3. Enter
serial?

4. The unit will return a serial number -
example

501023

or it may return

000000

5. In either case enter

serial="700105"

this is a false serial number which will give the unit a new identity to your computer.

6. Enter

serial?

7. The unit should return

OK 700105

8. Power down the unit and disconnect the serial cable.

9. In Windows - open the Control Panel, click on SYSTEM, then Hardware, then Device Manager.
In the Device Manager list find the Universal Serial Bus Controllers. Then find the USB-- Serial driver - click on this and delete.

10. Exit device manager

11. Turn on the DM unit and then plug in the USB cable.
You should get New Device Found indicator. When the New Device Wizard starts - DO NOT allow Windows to
automatically install - Manually control the installation of the driver. REPEAT! DO NOT ALLOW ANY AUTOMATIC INSTALLATION OF THE DRIVER.

The driver is already on your computer in the
C:\Program Files\Lectrosonics\LecNet2\drivers directory.
Direct your computer to this directory ONLY and it will properly load the correct driver for the DM.

12. After you have installed the driver properly, you can go back to LecNet Command terminal and change the serial number back to the original serial number using the serial command. Please note that when you do so, you may get the NEW DEVICE FOUND indicator again and will have to install the driver (manually only, just like before!) because the serial number has changed.

There are two possibilities.

First, you may have more than one AM* device in your system. If they are not correctly set for the master/slave configuration (example, two units both set as masters), then all the LED's will come on. Check to make certain the master is the first unit in the chain (there will be no cable in the expansion out jack on the back panel). ALso make certain that all the slaves have the master slave switch set to the slave position.

The second possibility is that you have all the channels set to direct mode - on all the time. Double check the dip swithc on the back panel for each channel. If you want automixing for that channel, set the correct DIP in the auto position. There is a graphic on the back panel showing the dip switch settings.