Frequently Asked Questions - LecNet2 (DM Series)

FAQs - LecNet2 (DM Series)

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.