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.

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 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.

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

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.

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

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.

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.

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.

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.

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.

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.

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.

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.

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.

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.