Axia - FAQ

General

Around how much total latency is involved in an Axia network?

It takes 2.75ms for sound to travel from the mic, through all of the Axia hardware, and to its specified destination.

I want to share one codec between two studios. How can I ensure that the backfeed to that device is coming from the correct studio?

Backfeed is automatically generated and sent to a device - in this case, your codec - when it's loaded as a source on your console. So to ensure that the proper mix minus is going to your codec, simply unload the source from any other consoles that may be using it before loading it onto another console's fader. If an "X" appears in the Channel Number position above the source name (on an Element), or if the OLED display reads "Listen Only" (on an iQ or Radius), your codec is in use elsewhere an must be freed up before you can control it with another console.

What's the difference between Livestreams and Standard Streams in an Axia system, and how do I know which to use?

Although Axia networks move audio data around at a very high rate of speed (Livestreams have a data rate of 4000 packets per second), not all audio requires this sort of speed. Livestreams are intended for sources that are "live", i.e., which are produced by a human and must be monitored or auditioned in real-time. Standard Streams (200 pps) are intended for pre-recorded sources where small amounts of latency are not noticeable.

Example: You speak into a microphone and are listening on headphones. The microphone's stream should be a Livestream because no latency is desired when one monitors oneself.

Conversely: a CD player is playing audio and you're listening to it on your headphones. A delay of 100ms or less; when pushing play is not noticeable. You can use a Standard Stream for this as it uses less network bandwidth.

So the rule of thumb is this: if something is generating audio and a human will monitor that same audio, that source should be set to Live Stereo, otherwise Standard.

Can you please advise the format/syntax for entering a Livewire stream URL into Windows Media Player and Winamp players?

You can do this using a DirectShow filter. Typical consumer applications do not support multicast directly, but with the use of the filter, players can play Livewire streams. An Axia Livewire DS Filter is available at ftp://beta.zephyr.com/~tlscorp/pub/Axia/Updates/LivewireDS/

Your player will need a stream description file which contains information about the multicast stream. The Axia filter supports the Session Description Protocol standard, where the description of audio channel transport parameters needs to persist in an .sdp file. RFC 2327 has more details on the SDP format.

Build an SDP file to be opened by WMP. The file will contain a Livewire audio channel in multicast address. (More about this translation can be found on the Axia Forum at http://forums.axiaaudio.com/viewtopic.php?t=240 .)

Here's an example of the contents of an .sdp file for a Livewire channel number of 1211, generated by a node with IP address 192.168.2.121:

v=0
o=Node 1 1 IN IP4 192.168.2.121
s=TestSine
t=0 0
a=type:multicast
c=IN IP4 239.192.4.187
m=audio 5004 RTP/AVP 96
a=rtpmap:96 L24/48000/2

.sdp files can be saved locally on a computer and opened for playback as needed.

This works on Windows Media Player or any other player that supports the SDP standard. Some players, like WinAmp, have no multicast support and cannot play Livewire streams and do not support .sdp files.

Consoles

Element

Can I share show profiles across two different Elements?

Yes. Export the Show Profile settings using Element's Customize option in the Element Control Center. Do this on the Element that you're copying the Show Profile from, and the one you want to copy it to. Backing up the console's configuration gives you an XML file; open the files from your source Element and target Element with an appropriate editor and carefully copy all of the Show Profile settings from one to the other. Be sure to save your work. Once you restore the newly saved XML file to the destination Element, that profile will become available.

Is there any actual mixing going on in an Element, or can I take it offline to swap out a module?

All audio mixing is actually done within the StudioEngine, so you can certainly take your Element offline and it won't affect the audio being generated in that studio. But Element modules are hot-swappable, so there's no need to take the whole mixer offline just to replace one module!

Where are the adjustments for the mic EQ on Element?

You can make adjustments for the mic EQ in the Element Control Center, the console's web-based configurator. From the main screen, click on Sources & Profiles, then Source Profiles. Choose the source profile of the mic that you wish to adjust. At the bottom of the page, you will find the EQ controls.

When I change sources on my Element console, the channel number above the fader sometimes changes to an 'X'. What does this mean?

This is the Element source sharing feature, warning you that the source you've selected is in use on another console. If you press Options for the channel with the "X," then scroll to that source in the source list, you can see which other studio has control of that source. This is to prevent you from taking control of a source that someone else is using, and perhaps taking them off the air!

If the source list tells you "none" (as in, the source is not in use elsewhere), but there is still an "X" displayed, please be sure that the "sharable" setting on the Source Profile page is set to "No".

Can we have split audio in the phone recorder? Like many facilities, we took the left channel and made it the Jocks voice from the prefader mic... and the right was caller only. We then summed them together for playback. The advantage here, is that you have raw caller audio, without the jock, for making promos and the like. How do we do this with our new Elements?

Element makes it VERY easy to do this!

Take the PGM4 Record bus and bring it into VMODE, panning it L or R (whichever you want). Then send the caller audio into another VMODE, panning to the opposite channel. Once you do that, you will sum the two VMODEs in a VMIX generating a new stereo channel that you can feed the recorder that will have a L/R split. Be sure, on the Source Profile for playback of your VMIX stream, that you SUM the two channels.

For more on VMIX and VMODE, see the Element manual at www.AxiaAudio.com/manuals/ .

How can we assign our Element's PREVIEW bus to the main monitors in the control room?

You can do this using VMIX. Assign the CR Mon channel and the Preview bus channel to channels 1 and 2 of a VMIX. Turn them on and send that VMIX output to your CR Mon speakers. The built-in logic, muting, and dim features of the preview and CR Mon channel will pass through the VMIX and work just as you would expect. If you need familiarization with VMIX, there is a chapter in the Element/PowerStation manual with plenty of information; get it from www.AxiaAudio.com/manuals/ .

What are some solutions for extending DVI monitor cables within the studio?

The Tech Tip in our March 2011 eNews featured some solutions for extending DVI monitor cables within the studio (if you didn't see it, you can read it at http://archive.constantcontact.com/fs084/1101974188054/archive/1104671623601.html ).

Afterward, we received some helpful information from one of our readers that we thought was worth passing on:

"Hi fellows this is Enrique from Prom TV Argentina, i would like share with you our experience with DVI cable. As we know DVI cable is for connect Power Station (main) with operator monitor, so if the maximum length between Power Station and Element Surface is 15 foot (DC CAN/Bus cable maximum length recommended) DVI cable must be similar. So we bought a high quality 15 foot HDMI-HDMI (full HD compliant) and two DVI female to HDMI male adapters. This is working properly since 1 year ago.

"I hope this help somebody to choose the right cable.

"Regards everyone - Enrique Spotti, Soluciones Tecnicas, Prom Tv - Mediacenter"

iQ/Radius

Could you please let me know what is the length of the CANBus cable for iQ and Radius?: The supplied CANBus cable is 20 feet (6.1 meters). This is the maximum recommended length.

Engines

Studio Engine

What is the normal temperature for a StudioEngine?

The Engine should be between 40-50 degrees Celsius, with maximum temperature spikes into the 50 degree range. Any piece of gear that is running consistently in the 60 degree Celsius range is not being properly ventilated.

How do I restore a StudioEngine to factory settings?

Press and hold the control knob for 6 seconds during boot up of the device.

What is the maximum length of the CANBus cable between PowerStation or an Axia Studio Engine and Element console? And what is the maximum additional length from the console to accessory modules using standard Category patch cables?

The maximum total length of all CANBus cabling associated with a specific Element should never exceed 100 feet (30 m). This includes the length of the power supply cable that connects the Element surface to the PowerStation or Studio Engine, since this is also a part of the CANBus. Best practices dictate keeping the CAN bus length to a minimum: a practical maximum is about 40 ft.

Proper termination is also required. See http://www.axiaaudio.com/manuals/files/Element_Termination_rev2_08142009.pdf for more information.

PowerStation

My PowerStation MAIN has a yellow indication on the PSU, but I have no AUX connected. What does this mean?: Check the setting on the PowerStation Engine Options page, and be sure that "use backup power" is set to NO. If you have it set to YES and do not have an AUX connected, then the MAIN is looking for the AUX power but can not find it, so it illuminates the PSU error status to let you know something is wrong. Checking "NO" will clear the error.

IP Intercom

Is there a way to mute the on-board speaker on IP Intercom rack units?

Yes. Just use the GPIO port of the intercom unit; page 33 of the IP Intercom manual shows pin 12 as the mute pin for the speaker. Just trip the pin whenever you want to silence the the speaer audio.

If you don't have the IP Intercom manual, just download it from /www.AxiaAudio.com/manuals/.

I have worked with the IP-Intercom IC-1 modules and noted that they have eight virtual GPIO ports. Do the ten and twenty-station OLED IP intercom units have eight virtual GPIO ports also?

Yes, they do! All IP Intercom units have GPIO. More data at www.AxiaAudio.com/intercom/

Nodes

Audio Node

If I need to reset an Axia Audio Node, can I do it via software, or do I have to unplug it?

Yes. Telnet to the IP address of the node. The user and pass will be the same one you use when logging into the webpage. From there, simply type in "reboot" to power cycle the unit.

Are Audio Node RJ-45 connectors shielded? I want to shield the cable from my audio device to the node, using a screened RJ-45 connector and FTP cable to provide a shielded link.

In most circumstances, it does not make any difference what type cable you use (shielded or non-shielded). Network cabling makes great analog audio cable since the twists are very precise. If you are looking to run this cable over longer distances, be sure your audio is properly balanced and operating at a standard line level such as +4 dB, you can run a few hundred meters without any issues.

Is there a way to get a true monaural output (L+R), not simply a dual mono split of the L and R, out of a mix engine and/or node?

Yes, you can do this using the VMODE function in the Studio Engine or PowerStation.

Select the source you want to "mono-ize" as an input into VMODE (input 1). Select "Upmix L+R" for Audio mode. This will generate a stream which has the sum of L+R on both channels of the Livewire stream.

If you want to create a stream that only has audio on one channel while not any audio on the other, take the stream just created above and set it as an input into VMODE (input 3)and select combine L,L and do not use the input below (input 4). This will create a stream which is the LEFT channel from input 3 and the LEFT channel of input 4. Since input 3 Left is L+R of the original stream and input 4 is not used, the resulting stream is L+R on the Left channel and NUL on the Right Channel of VMODE-3.

GPIO Nodes

We have Mika mic stand arms with the built-in on-air light. I need to be able to assign the GPIO to each mic in the control room and studio, BUT I also need to turn-on the outside studio on-air light when any mic is ON. How can I accomplish this?

In the Element manual, in the chapter that covers GPIO, the table for Control Room (CR) monitor logic shows that OUT-1 is CR Mute. Anytime the Monitor speakers mute, there will be continuity with this pin and the common out pin.

To harness this logic for use with an on-air light, navigate to the Element configuration page. Select a show profile and select the monitor section link. Under the "Control Room Monitor Options" heading, find the "GPIO channel for CR monitor:" property and enter the same Channel Number assigned to the CR Monitor channel. The number entered in the field is the number to be applied to the GPIO port that will trigger the on-air light.

If you have multiple show profiles, you will want to make sure each show profile has this setting applied. Remember, any changes made to a Show profile will not take effect until you load the reload the Profile.

Software

iProFiler

Can a PC running iProFiler connect directly to an Audio node without the need for a network switch?: Yes, using a crossover cable from the Node to the NIC of your PC. Be certain that all Node sources are configured as Standard stereo streams, and you are ready to log.

Pathfinder

Will PathfinderPC Server run on the 64-bit version of Windows 2008 Server?: Yes it will. Just be sure that you "run as administrator" by navigating to the executable file after installing under the Admin profile, and then right-clicking on the files and clicking the box that says "run as administrator".