Fusion Console - The Telos Alliance

Fusion Console - The Telos Alliance
Fusion Console
Installation & User’s Guide Includes StudioEngine and PowerStation
Manual Rev 2 (3.1) - April 2017
p/n 1490-00109-004
Notices and Cautions
CAUTION:
The installation and service instructions in this manual are for use by qualified personnel only. To avoid electric
shock, do not perform any servicing other than that contained in the operating instructions unless you are qualified to
do so. Refer all servicing to qualified personnel.
This instrument has an autoranging line voltage input. Ensure the power voltage is within the specified range of
100-240v. The ~ symbol, if used, indicates an alternating current supply.
This symbol, wherever it appears, alerts you to the presence of uninsulated, dangerous voltage
inside the enclosure – voltage which may be sufficient to constitute a risk of shock.
This symbol, wherever it appears, alerts you to important operating and maintenance instructions.
Read the manual.
CAUTION: DOUBLE POLE/NEUTRAL FUSING
The instrument power supply incorporates an internal fuse. Hazardous voltages may still be present on some of
the primary parts even when the fuse has blown. If fuse replacement is required, replace fuse only with same type
and value for continued protection against fire.
WARNING:
The product’s power cord is the primary disconnect device. The socket outlet should be located near the device
and easily accessible. The unit should not be located such that access to the power cord is impaired. If the unit is
incorporated into an equipment rack, an easily accessible safety disconnect device should be included in the rack
design.
To reduce the risk of electrical shock, do not expose this product to rain or moisture. This unit is for indoor use only.
Notices and Cautions • ii
This equipment requires the free flow of air for adequate cooling. Do not block the ventilation openings in the
top and sides of the unit. Failure to allow proper ventilation could damage the unit or create a fire hazard. Do not
place the units on a carpet, bedding, or other materials that could interfere with any panel ventilation openings.
If the equipment is used in a manner not specified by the manufacturer, the protection provided by the equipment
may be impaired.
© 2017 Axia Audio - Rev 2.0
WARNUNG:
Die Installations-und Serviceanleitung in diesem Handbuch ist für die Benutzung durch qualifiziertes
Fachpersonal. Um Stromschläge zu vermeiden führen Sie keine andere Wartung durch als in dieser
Betriebsanleitung aufgeführt, es sei denn Sie sind dafür qualifiziert. Überlassen Sie alle Reparaturarbeiten
qualifiziertem Fachpersonal.
Dieses Gerät hat eine automatische Bereichseinstellung der Netzspannung.
Stellen sie sicher, dass die verwendete Netzspannung im Bereich von 100-240V liegt.
Das Symbol ~, falls verwendet, bezeichnet eine Wechselstromversorgung.
Dieses Symbol, wo immer es auftaucht, macht Sie auf nicht isolierte, gefährliche elektrische
Spannung (ausreichend um einen Stromschlag hervorzurufen) innerhalb des Gehäuses
aufmerksam. Spannungen.
Dieses Symbol, wo immer es auftaucht, weist Sie auf wichtige Bedienungs-und
Wartungsanleitung hin. Lesen Sie die Bedienungsanleitung.
ACHTUNG: ZWEIPOLIGE ABSICHERUNG / NULLEITER ABSICHERUNG
Das Netzteil des Gerätes hat eine interne Sicherung eingebaut. Auch wenn die Sicherung durchgebrannt ist,
können auf einigen primären Bauteilen noch gefährliche Spannungen vorhanden sein. Wenn ein Austausch der
Sicherung erforderlich ist, ersetzen Sie die Sicherung nur mit gleicher Art und Wert für den kontinuierlichen Schutz
gegen Feuer.
WARNUNG:
Um die Gefahr von Stromschlägen zu verringern, darf dieses Produkt nicht Regen oder Feuchtigkeit ausgesetzt
werden. Dieses Gerät ist nur für die Benützung im Innenbereich. Dieses Gerät erfordert freie Luftzirkulation für
eine ausreichende Kühlung. Blockieren sie nicht die Lüftungsschlitze auf der Geräteoberseite und den Seiten des
Gerätes. Unzureichende Belüftung kann das Gerät beschädigen oder Brandgefahr verursachen. Platzieren Sie
das Gerät nicht auf einem Teppich, Poster oder andere Materialien welche die Lüftungsöffnungen beeinträchtigen
könnten.
Wird das Gerät anders als in der, vom Hersteller angegebenen Weise verwendet, kann der, durch das Gerät
gegebene Schutz beeinträchtigt werden.
© 2017 Axia Audio - Rev 2.0
Notices and Cautions • iii
Das Gerätenetzkabel ist die Haupttrennvorrichtung. Die Steckdose sollte sich in der Nähe des Gerätes befinden
und leicht zugänglich sein. Das Gerät sollte nicht so angeordnet sein, dass der Zugang zum Netzkabel beeinträchtigt
ist. Wird das Gerät in ein Rack eingebaut, sollte eine leicht zugängliche Sicherheitstrennvorrichtung in den RackAufbau mit einbezogen werden.
USA CLASS A COMPUTING DEVICE INFORMATION TO USER. WARNING:
This equipment generates, uses, and can radiate radio-frequency energy. If it is not installed and used as directed
by this manual, it may cause interference to radio communication. This equipment complies with the limits for a
class a computing device, as specified by fcc rules, part 15, subpart j, which are designed to provide reasonable
protection against such interference when this type of equipment is operated in a commercial environment.
Operation of this equipment in a residential area is likely to cause interference. If it does, the user will be required to
eliminate the interference at the user’s expense. Note: objectionable interference to tv or radio reception can occur if
other devices are connected to this device without the use of shielded interconnect cables. Fcc rules require the use
of shielded cables.
CANADA WARNING:
“This digital apparatus does not exceed the class a limits for radio noise emissions set out in the radio interference
regulations of the Canadian department of communications.”
“Le présent appareil numérique n’émet pas de bruits radioélectriques dépassant les limites applicables aux
appareils numériques (de class a) prescrites dans le règlement sur le brouillage radioélectrique édicté par le
ministère des communications du Canada.”
CE CONFORMANCE INFORMATION:
This device complies with the requirements of the EEC council directives:
•
93/68/EEC (CE MARKING)
•
73/23/EEC (SAFETY – LOW VOLTAGE DIRECTIVE)
•
89/336/EEC (ELECTROMAGNETIC COMPATIBILITY)
Notices and Cautions • iv
Conformity is declared to those standards: EN50081-1, EN50082-1.
© 2017 Axia Audio - Rev 2.0
Fusion Manual
© 2014-2015 TLS Corp. Published by Axia Audio/TLS Corp. All rights reserved.
TRADEMARKS
Axia Audio and the Axia and Fusion logos are trademarks of TLS Corp. All other trademarks are the property of
their respective holders.
NOTICE
All versions, claims of compatibility, trademarks, etc. of hardware and software products not made by Axia
Audio which are mentioned in this manual or accompanying material are informational only. Axia makes no
endorsement of any particular product for any purpose, nor claims any responsibility for operation or accuracy. We
reserve the right to make improvements or changes in the products described in this manual which may affect the
product specifications, or to revise the manual without notice.
WARRANTY
This product is covered by a five year limited warranty, the full text of which is included in this manual.
UPDATES
The operation of Fusion is determined largely by software. We routinely release new versions to add features
and fix bugs. Check the Axia Audio web site for the latest. We encourage you to sign-up for the email notification
service offered on the site.
FEEDBACK
We welcome feedback on any aspect of Fusion, or this manual. In the past, many good ideas from users have
made their way into software revisions or new products. Please contact us with your comments.
You must contact Axia before returning any equipment for factory service. We will need your unit’s serial
number, located on the back of the unit. Axia will issue a return authorization number, which must be written on the
exterior of your shipping container. Please do not include cables or accessories unless specifically requested by the
Technical Support Engineer. Be sure to adequately insure your shipment for its replacement value. Packages without
proper authorization may be refused. US customers, please contact Axia Technical Support at +1-216-622-0247. All
other customers should contact local representative to make arrangements for service.
© 2017 Axia Audio - Rev 2.0
Notices and Cautions • v
SERVICE
We Support You...
BY PHONE / FAX:
•
You may reach our 24/7 Support team anytime around the clock by calling +1-216-622-0247.
•
For billing questions or other non-emergency technical questions, call +1-216-241-7225 between 9:30 am
to 6:00 PM, USA Eastern time, Monday through Friday.
•
Our Fax number is +1-216-241-4103.
BY E-MAIL:
•
Technical support is available at [email protected]
•
All other questions, please email [email protected]
VIA WORLD WIDE WEB:
The Axia Audio web site has a variety of information which may be useful for product selection and support.
The url is telosalliance.com.
REGISTER YOUR PRODUCT
Did you know that all Telos Alliance products come with a 5-Year Warranty? Take a moment to activate your
coverage online at http://telosalliance.com/product-registration/ .
THE TELOS ALLIANCE
1241 Superior Avenue E.
Cleveland, OH., 44114 USA
We Support You • vi
+1-216-241-7225 (phone)
+1-216-241-4103 (fax)
+1-216-622-0247 (24/7 technical support)
[email protected]
[email protected]
© 2017 Axia Audio - Rev 2.0
Table of Contents
Notices and Cautions . . . . . . . . . . . . . . . ii
Chapter Four: VMix and VMode . . . . . . . . . . . . 49
We Support You . . . . . . . . . . . . . . . . . . vi
Virtual Mixing with VMix . . . . . . . . . . . . . . 49
A Note From Frank Foti . . . . . . . . . . . . . . ix
What’s it all about? . . . . . . . . . . . . . . . . 49
VMix Main Controls . . . . . . . . . . . . . . . . 50
Quickstart #1 Fast Setup . . . . . . . . . . . . . . . . 1
VMix Submixer Controls . . . . . . . . . . . . . . 50
Fast Setup – with Axia StudioEngine . . . . . . . . . 1
Some VMix Examples . . . . . . . . . . . . . . . 52
Creating Your First Source And Assigning It To A Fader . . 5
Virtual VMix Control . . . . . . . . . . . . . . . 52
Setting Up And Testing Control Room Monitor Channels .
Combining VMix with Pathfinder Routing Control . 52
7
Manipulating Streams With VMode . . . . . . . . . 54
Quickstart #2 Fast Setup . . . . . . . . . . . . . . . . 9
The Input Side . . . . . . . . . . . . . . . . . . . 55
Fast Setup – with Axia PowerStation . . . . . . . . . 9
The Audio Modes . . . . . . . . . . . . . . . . . 55
Creating Your First Source And Assigning It To A Fader . . 12
The Output Side . . . . . . . . . . . . . . . . . . 56
Setting Up And Testing Control Room Monitor Channels .
14
Some VMode Examples . . . . . . . . . . . . . . . 57
Mono Stream From One Side of a Stereo Channel . . . 57
Chapter One: Fusion Anatomy . . . . . . . . . . . . . 17
Split Record Feed From Multiple Sources . . . . . 58
StudioEngine and PowerStation Mixing Engines . . 17
CAN4ETH . . . . . . . . . . . . . . . . . . . . . . 21
Chapter Five: Working With Phones . . . . . . . . . . 59
Fusion Power Supply . . . . . . . . . . . . . . . . . 22
Phone Setup Choices . . . . . . . . . . . . . . . . . 59
Console Modules . . . . . . . . . . . . . . . . . . . 22
Setting Up for EU Phone Operation . . . . . . . . . 60
Setting Up for US Phone Operation . . . . . . . . . 61
Chapter Two: Configuration Basics . . . . . . . . . . 23
Show Profile Settings For US Phone Operation . . 61
Working With Profiles . . . . . . . . . . . . . . . . 23
Source Profile Settings for US Phone Operation –
Basic Operation . . . . . . . . . . . . . . . . . . . . 23
VX & Nx Systems . . . . . . . . . . . . . . . . . . . . 63
Source Profiles . . . . . . . . . . . . . . . . . . . . 23
Source Profile Settings for US Phone Operation –
Create the Operator’s Mic Source Profile . . . . . 25
Hx6 & iQ6 Systems . . . . . . . . . . . . . . . . 65
Create a Guest Mic Source Profile . . . . . . . . 27
Setting up for GPIO control (“No Phone Control”) . . 67
Other Mic Profile Types . . . . . . . . . . . . . . 28
Additional Phone Type Source Profile Options . . . . 68
Create a CD Player Source (Line Source Type) . . 29
Create a Telephone Source (Phone Source Type) . 31
Appendix A: Advanced Configuration Reference . . . 69
Advanced Stuff: A Codec Source With
Custom Backfeed . . . . . . . . . . . . . . . . . . 33
Appendix B: Configuring GPIO . . . . . . . . . . . . 87
Show Profiles . . . . . . . . . . . . . . . . . . . . . 37
Chapter Three: Operation . . . . . . . . . . . . . . . 39
Appendix C: Specifications . . . . . . . . . . . . . . . 99
Appendix D: CE Declaration of Conformity . . . . . . 105
Main Display . . . . . . . . . . . . . . . . . . . . . 39
Basic Channel Controls . . . . . . . . . . . . . . . . 41
Appendix E: Warranty . . . . . . . . . . . . . . . . . 107
Expert Monitor Module Controls . . . . . . . . . . . 42
Standard Monitor Module Controls . . . . . . . . . . 44
Appendix F: New Features . . . . . . . . . . . . . . . 109
Call Controller + 2-Fader Phone Module . . . . . . . 46
IP Intercom Module – 20-Station . . . . . . . . . . 47
IP Intercom Module – 10-Station Filmcap . . . . . . 48
Switch Modules . . . . . . . . . . . . . . . . . . . . 48
© 2017 Axia Audio - Rev 2.0
Table of Contents • vii
How to Create A Show Profile . . . . . . . . . . . 37
War of the Waves
Dear Valued Customer,
It’s with great pride and a tip of the hat to an incredible team that I congratulate you on your new Telos Alliance
product. Everything we do here at the radio division of the Telos Alliance is with one end goal in mind: To help
broadcasters declare victory in extremely competitive environments. By purchasing this product from us, in essence,
you have declared war on your competition.
After all, the majority of Telos Alliance employees were broadcasters themselves once, and the products we’ve
developed over the years have been designed as solutions to specific issues faced on the front lines of our industry.
We’re right there in the trenches with you and have the weapons you need in your arsenal.
Telos Systems is a catalyst to out-of-this-world sound, with the most powerful and popular broadcast telephone
systems in the industry; IP/ISDN codecs and transceivers; plus processing/encoding for streaming audio. We built an
industry on the back of these amazing telephony systems, and they are still going strong.
While we at the Telos Alliance never forget our roots, we are also blazing trails in terms of new technologies like
stream-encoding and AoIP, so that all types of broadcasters can excel in this ever-evolving digital world.
Omnia Audio not only lets you stand out on the dial with your unique signature sound via legendary audio
processors, audio codecs, and microphone processing, it lets you give your listeners a better streaming experience
across devices with innovative stream encoding/processing software and hardware.
Axia Audio is a driving force behind the AES67 AoIP standard, and its networked AoIP radio consoles, audio
interfaces, networked intercom, and software products continue to move AoIP adoption forward and help
broadcasters streamline operations with cohesive, smart, and feature-rich AoIP ecosystems.
You work so hard on your programming day-in and day-out, it deserves technology that will optimize sound and
performance at every point in the airchain and online. Armed with Telos products, you have what you need to set
your competition squarely in your crosshairs.
With that, I’ll leave you to prep your armaments. I hope that you will enjoy your Telos Alliance products for many
years to come!
Sincerely,
Frank Foti
CEO, The Telos Alliance
© 2017 Axia Audio - Rev 2.0
A Note From Frank Foti • ix
Last, but certainly not least, 25-Seven has traditionally been known for its audio delays, but its Voltair watermark
monitor/processor has made a name for itself more recently as the disruptive product that helped broadcasters take
back their ratings and harness the true power of their listening audiences.
Quickstart #1
Fast Setup – with Axia StudioEngine
Setup of a networked audio mixing console, though we have made it as easy as possible, is challenging to
condense into a few pages. The following material is intended to help the busy engineer get up and running in a few
minutes.
This Quickstart section setup assumes a few things:
•
That the reader has some knowledge of network basics and network terminology,
•
That the reader is familiar with other Axia Livewire products,
•
That the reader is familiar with the anatomy of the Fusion system,
•
And that the reader has a correctly configured network switch
Once you’re up and running, please peruse the remainder of this manual to gain in-depth knowledge of more
advanced options.
AC Mains connections for redundant power supplies
Livewire NIC
To begin, connect a 1080p widescreen computer monitor to the DVI port on the Axia StudioEngine’s rear panel.
Also connect the RJ-45 port marked with the Livewire logo to a port on your AoIP network’s local switch using a
CAT-6 Ethernet cable.
Using the IEC cables supplied, connect the StudioEngine’s power supply input to AC mains.
© 2017 Axia Audio - Rev 2.0
Quickstart - StudioEngine • 1
DVI Monitor connector
Tilt overbridge forward (toward faders)
to gain access to connection panel.
Next, connect the Power cable supplied with your Fusion Power Supply to the Power connection on the board
located beneath the overbridge of the Fusion surface.
Also connect your Fusion to your AoIP network’s local switch, using a CAT-5e Ethernet cable.
AC Mains connection
Quickstart - StudioEngine • 2
Console power connection
Connect the Power cable from your Fusion console to the marked power supply port on the rear of your Fusion
Power Supply, and then connect the Fusion Power Supply to AC mains using the IEC cable supplied.
© 2017 Axia Audio - Rev 2.0
When the StudioEngine has booted, its display will indicate an error state. Don’t panic! An IP address needs to
be entered. To do this:
1. Tap the rotary encoder on the front panel, once. A menu view will appear.
2. Rotate the encoder to highlight “Engine IP settings”, and tap the encoder again to select it. A new menu
will appear.
3. Rotate the encoder to select “Net Address” and tap to select.
4. Rotate the encoder to move the onscreen cursor. A tap on any of the cursor position allows you to edit its
numeric value; rotate the encoder to increment or decrement the value and tap to accept the change. Repeat
this until the IP address you desire is entered.
5. Select the check mark to the right of the IP address to accept.
6. Repeat this process for the Netmask and Gateway fields, as needed.
7. Select “Console” and change the option to “Fusion”.
8. Select “OK.” After selecting OK, respond affirmative to the following notice to reboot the engine with its
new settings.
•
Username = user
•
Password = <leave blank>
Quickstart - StudioEngine • 3
The next step requires a Web browser on a PC connected to your AoIP network. Open your browser and enter the IP
you assigned to your StudioEngine in the URL bar; an authentication window will appear. Enter the following values:
© 2017 Axia Audio - Rev 2.0
Select “OK” and your Fusion console’s control center will be loaded.
1. From the links under the Mix Engine heading, select Network.
Quickstart - StudioEngine • 4
2. Click the Install button displayed at the bottom of the Network page. A new page will load.
3. Enter the value 1 into the Console count field and click the Apply Console Count button.
4. The software will display a list of the consoles found on your network. Select your Fusion from the drop-down list.
5. Click the Apply IP button to set the IP address of your Fusion, and to link it with the StudioEngine.
© 2017 Axia Audio - Rev 2.0
Your Fusion console is now connected! Notice that the displays above each fader strip read INACTIVE. One
more fast step will complete the setup process.
Press the
(star) and 2 keys on the numeric keypad of your console’s Monitor module. Hold them until the
displays change to read CAPTURE.
Release the
and 2 keys, then press the Enter key (Expert monitor module) or # key (Standard monitor module).
Congratulations! Your Fusion console is now configured and ready to load sources. The displays above each
fader strip will now contain a number indicating the fader position.
Creating Your First Source And Assigning It To A Fader
You’ll use your Web browser and the Fusion interface to create your first audio source in just a few fast steps.
1. In the side menu under the Console heading, select Sources, then click the Create new source profile button.
Quickstart - StudioEngine • 5
2. Enter a name for your new audio source in the Source name field.
© 2017 Axia Audio - Rev 2.0
3. Press the browse button to the right of the Primary source field; a pop-up window will appear with a list of
sources available on your Axia network. Select the desired source from the list, and click the OK button.
Quickstart - StudioEngine • 6
4. Your Source Profile has been created! Your screen should show an entry similar to that shown below.
© 2017 Axia Audio - Rev 2.0
Setting Up And Testing Control Room Monitor Channels
1. In the side menu, under the Mix Engine heading, select the Prog and mon out option.
2. Enter the planned Channel Numbers for the StudioEngine outputs (Livewire Sources).
Click the Apply button.
Quickstart - StudioEngine • 7
3. On your Fusion console, press the Options knob found at the top of any fader strip. Your video monitor’s
center section will change, showing the Channel Options display.
© 2017 Axia Audio - Rev 2.0
4. Rotate the Options encoder to highlight the Current Source window. (Highlighted options are bordered in
yellow), and then press the Encoder.
5. You’ll now see a list of your available Sources (in this case, just one: the one we just configured.)
6. Rotate the Options encoder to highlight the source you just created. Press the encoder to select the highlighted source.
Quickstart - StudioEngine • 8
7. Press the Program 1 key on the fader strip; it will illuminate to show you’ve assigned that fader to PGM-1.
Press the ON key at the bottom of the fader strip and move the fader up. Congratulations: you’ve got audio!
The meters on your display should be active, as shown below.
To hear the audio, make sure you have selected Program 1 as your CR Monitor source, using the controls on
your Monitor module, and that the volume is at an appropriate level (the level meters onscreen will show the relative
volume you’ve set). Also, make sure the xNode feeding your speaker has been configured to the CR Monitor source
you assigned a channel number to.
Now, sit back and enjoy some music as you read the remainder of this manual, or continue to explore the
Configuration pages of your StudioEngine; the details of these options can be found in the Configuration section of
this manual.
© 2017 Axia Audio - Rev 2.0
Quickstart #2
Fast Setup – with Axia PowerStation
Setup of a networked audio mixing console, though we have made it as easy as possible, is challenging to
condense into a few pages. The following material is intended to help the busy engineer get up and running in a few
minutes.
This Quickstart section setup assumes a few things:
•
That the reader has some knowledge of network basics and network terminology,
•
That the reader is familiar with other Axia Livewire products,
•
That the reader is familiar with the anatomy of the Fusion system,
•
And that the reader has a correctly configured network switch
Once you’re up and running, please peruse the remainder of this manual to gain in-depth knowledge of more
advanced options.
USB Port
DVI Monitor connector
To begin, connect a 1080p widescreen computer monitor to the DVI port on the Axia PowerStation Main rear panel.
Connect a USB keyboard to the USB port on the rear of your PowerStation Main.
© 2017 Axia Audio - Rev 2.0
Quickstart - PowerStation • 9
AC Mains connection
Livewire NIC
Using the IEC cable supplied, connect the PowerStation’s power supply input to AC mains. Allow PowerStation
to boot; a Setup screen will appear on your monitor.
To set up your PowerStation:
1. Press your keyboard’s up-arrow key until the Config IP addr line is highlighted in yellow text.
2. Press the Backspace key to clear values, and use the number keys to enter new values and assign an IP address to the PowerStation.
3. Use the arrow keys to move to the Config NetMask and Config Gateway fields, repeat the process to set
the needed values.
4. Use the down-arrow key to move to the Save & Reboot selection. Press the Enter key and the system will
reboot, with the IP address you’ve entered.
5. Remove the USB keyboard.
Quickstart - PowerStation • 10
Tilt overbridge forward (toward faders)
to gain access to connection panel.
Next, connect one end of the 6-pin Molex power cable supplied with your PowerStation to the Power connection
on the board located beneath the overbridge of the Fusion surface.
© 2017 Axia Audio - Rev 2.0
Connect the other end to the Molex connector marked SURFACE on the rear of your PowerStation.
Now, Connect a network patch cable from the board under the Fusion’s overbridge to any available port on the
PowerStation’s switch.
The next step requires a Web browser on a PC connected to any Ethernet port on your PowerStation’s switch
section. Open your browser and enter the IP you assigned to your PowerStation in the URL bar; an authentication
window will appear. Enter the following values:
•
Username = user
•
Password = <leave blank>
1. From the links under the Mix Engine heading, select Network.
2. Click the Install button displayed at the bottom of the Network page. A new page will load.
© 2017 Axia Audio - Rev 2.0
Quickstart - PowerStation • 11
Select “OK” and your Fusion console’s control center will be loaded.
3. Enter the value 1 into the Console count field and click the Apply Console Count button.
4. The software will display a list of the consoles found on your network. Select your Fusion from the
drop-down list.
5. Click the Apply IP button to set the IP address of your Fusion, and to link it with the PowerStation.
Your Fusion console is now connected! Notice that the displays above each fader strip read INACTIVE. One
more fast step will complete the setup process.
Press the
(star) and 2 keys on the numeric keypad of your console’s Monitor module. Hold them until the
displays change to read CAPTURE.
Release the
and 2 keys, then press the Enter key (Expert monitor module) or # key (Standard monitor module).
Congratulations! Your Fusion console is now configured and ready to load sources.
Creating Your First Source And Assigning It To A Fader
Quickstart - PowerStation • 12
You’ll use your Web browser and the Fusion interface to create your first audio source in just a few fast steps.
1. In the side menu under the Console heading, select Sources, then click the Create new source
profile button.
© 2017 Axia Audio - Rev 2.0
2. Enter a name for your new audio source in the Source name field.
3. Press the browse button to the right of the Primary source field; a pop-up window will appear with a list of
sources available on your Axia network. Select the desired source from the list, and click the OK button.
Quickstart - PowerStation • 13
4. Your Source Profile has been created! Your screen should show an entry similar to that shown below.
© 2017 Axia Audio - Rev 2.0
Setting Up And Testing Control Room Monitor Channels
1. In the side menu, under the Mix Engine heading, select the Prog and mon out option.
2. Enter the planned Channel Numbers for the PowerStation outputs (Livewire Sources from DSP).
Click the Apply button.
3. In the side menu, select the Destinations link under the I/O subsystem main heading.
You’ll be prompted for a Username and Password. Enter the following values:
Username = user
•
Password = <leave blank>
Quickstart - PowerStation • 14
•
© 2017 Axia Audio - Rev 2.0
4. The designated output on the back of the PowerStation which provides audio to the monitor speaker
amplifier needs to be configured for the CR Monitor DSP source. Type a descriptive name in the “Name”
field and select the Browse button next to the “Channel” field. From the popup, select the CR Monitor
source from the PowerStation.
Quickstart - PowerStation • 15
5. On your Fusion console, press the Options knob found at the top of any fader strip. Your video monitor’s
center section will change, showing the Channel Options display.
© 2017 Axia Audio - Rev 2.0
6. Rotate the Options encoder to highlight the Current Source window. (Highlighted options are bordered
in yellow), and then press the Encoder.
7. You’ll now see a list of your available Sources (in this case, just one: the one we just configured. If you
already have an Axia network deployed, you may see a larger list.)
8. Rotate the Options encoder to highlight the source you just created. Press the encoder to select the highlighted source.
Quickstart - PowerStation • 16
9. Press the Program 1 key on the fader strip; it will illuminate to show you’ve assigned that fader to PGM-1.
Press the ON key at the bottom of the fader strip and move the fader up. Congratulations: you’ve got audio!
The meters on your display should be active, as shown below.
To hear the audio, make sure you have selected Program 1 as your CR Monitor source, using the controls on
your Monitor module, and that the volume is at an appropriate level (the level meters onscreen will show the relative
volume you’ve set).
Now, sit back and enjoy some music as you read the remainder of this manual, or continue to explore the
Configuration pages of your PowerStation; the details of these options can be found in the Configuration section of
this manual.
© 2017 Axia Audio - Rev 2.0
Chapter 1:
Fusion Anatomy
The Fusion modular control surface is more than a just console — it is a complete studio system, with various
components that serve different functions.
This chapter gives an overview of the different components to help familiarize you;. The following chapters will
build on this familiarity as we dive deeper into Fusion’s features and capabilities.
StudioEngine and PowerStation Mixing Engines
The StudioEngine and PowerStation mixing engines serve as the “brains” of the studio. Each contains a CPU
that powers console operations, as well as the DSP mixing functionality.
Audio from the Axia network enters the mixing engine and is manipulated per the user’s actions on the Fusion
surface. The engine’s Console function communicates with the surface and maintains the logical states that are
required in a studio environment in addition to controlling the Mix Engine. Those logical states include:
•
Muting
•
ON/OFF commands
•
GPIO control
•
Mix conditions
•
Source ownership (which mixing surface has control of an active source)
•
Phone control
Fusion Anatomy • 17
…and many others. The PowerStation mixing engine has additional functionality, which includes networked I/O
and an integrated AoIP-tailored network switch.
© 2017 Axia Audio - Rev 2.0
The front-panel of the StudioEngine has a display and a dual-function rotary encoder. The display normally
provides basic system information, along with an “OK” status indicator. If the “OK”indicator is missing from the
screen, it’s because something needs attention — in its place will be an error message describing any items the unit’s
self-diagnostics consider out-of-the-ordinary.
The display also provides access to the StudioEngine’s menu options. From the OK status view, tapping the
encoder activates the menu system, which can be navigated with rotations of the knob, and selections made by taps
(pressing the knob in).
The rear of the Studio Engine has two power supplies, each with standard EAC connectors. The DVI-D video
connector is a digital video output intended to be connected to a monitor that supports 1080p resolution.
Fusion Anatomy • 18
Below the video connector are two RJ45 ports. The Livewire port is used for connection to the Axia network
(the second port is reserved for future use). The two USB ports are also for future use, or possible service
functionality.
© 2017 Axia Audio - Rev 2.0
The PowerStation MAIN front panel has no user interface, but does have a number of useful status indicators:
•
The Axia logo, glowing blue, indicates that power is present in the internal power supply (after redundantpower-supply isolation, if a PowerStation AUX with redundant power is connected)
•
The PSU indicator shows the status of the power supply:
»» A green indication means all is well.
»» An orange indication notes that the secondary (backup) power supply is not functioning. (If a PowerStation AUX unit is not installed, software detects this at setup; the orange indication will not be
produced in this case.)
•
The LIVEWIRE indication represents correct network performance. This should always be illuminated
when your PowerStation is connected to a Livewire network; if not, there may be a failure of the internal
switch or internal cable.
•
The MASTER indication shows that this PowerStation is source of network clock sync. If sync is provided
by an external master clock or another Livewire device, this will not be illuminated. Only one device can
be a source of clock sync (the Clock Master) within a network. If multiple Masters are illuminated in a facility, this suggests a network configuration error or loss of network connectivity to a facilities core switch.
•
The SYNC indication is typically illuminated, showing that the PowerStation I/O board is receiving network clock:
»» A solid green indication is normal behavior.
»» A blinking green indication tells you that PowerStation’s I/O is not synchronized with the network.
This can happen momentarily, at boot, as the device syncs with received network packets, or if there
is a change in source of clock from one Master to another.
»» A red indication suggests either that no network sync is present, or a possible failure of the I/O board.
•
The OK indicator lights to show that the DSP engine is functioning and is receiving clock from the network. If it is not lit, either there is no network sync available to the DSP engine, or the main CPU has failed.
© 2017 Axia Audio - Rev 2.0
Fusion Anatomy • 19
»» A red indication shows that the PowerStation Main’s internal power supply has failed, and that
power is now being provided by the backup supply in the PowerStation AUX (if installed).
The PowerStation AUX has the same front panel as the PowerStation MAIN, and front-panel indicators are similar.
•
PowerStation AUX has no OK indicator, since there is no main CPU in the AUX.
•
The LINK indicator lights to show that the I/O board has an active network link.
PowerStation MAIN connection panel
PowerStation AUX connection panel
Fusion Anatomy • 20
The rear of the PowerStation is divided into three categories: Input/Output, Network switch, and System ports.
The audio inputs on each PowerStation consist of:
•
Two (2) XLR microphone inputs
•
Four (4) RJ45 stereo analog inputs
•
Two (2) RJ45 AES digital audios inputs
•
Six (6) RJ45 stereo analog outputs
•
Two (2) RJ45 AES digital audio outputs
© 2017 Axia Audio - Rev 2.0
General Purpose Input Output (GPIO) ports are four (4) DA-15 male connectors. Each DA-15 has 5 solid state
relay output pins and 5 opto-isolated input pins
The integrated zero-configuration network switch has four (4) 100MB ports with PoE (Power over Ethernet),
ten (10) unpowered 100MB ports, and two (2) 1GB ports with parallel SFP ports. The system ports are a
D-subminiature (DA) backup power port, EAC AC power port, USB, DVI-D port for monitor connection, and a
6-pin Molex connector to supply the console with 48v power.
CAN4ETH
Underneath the overbridge of the Fusion surface is the board that links the console modules to the main CPU
of either your StudioEngine or PowerStation. The surface uses CANBus (Control Area Network) as a common
interface for modules, then links the surface’s mixing modules to its associated CPU via an Ethernet link.
Connection from Modules
Network connection
Main and Backup Power connections
The board has a single RJ45 network port for connection to the network. Near the network port are two (2) 6-pin
molex connectors for power input. The two ports provide the option to connect a redundant power supply, but only a
single power connection is required to power the surface.
The maximum amount of modules allowed on a single port depends on the type of modules, but the limitation
is power draw. A good rule of thumb is that three modules may be daisy-chained to each of the RJ45s on the
CAN4ETH board, but more may be possible — we invite you to your configuration with Axia Support if you have
concerns. Typically, your console will arrive as ordered in a factory-approved configuration.
Next to the RJ45 stack is a single 6-PIN Molex terminal that is in parallel with one of the CANBus ports. This is
reserved for future use; do not use this port unless specifically instructed to do so by Axia Support.
© 2017 Axia Audio - Rev 2.0
Fusion Anatomy • 21
On the other end of the board is a four (4) port RJ45 stack; each port is an independent CANBus port. These
connections are used to interface with the Fusion’s installed modules. If your console has more than four modules
installed (as many do), the modules themselves can be daisy-chained using the dual RJ45 connectors provided on
each module.
Fusion Power Supply
Fusion consoles require a 48-volts power supply to the CAN4ETH board to function.
If you are using an Axia PowerStation, this power is supplied by the Console port on the rear panel.
If you have chosen an Axia StudioEngine, a separate Axia Power Supply is required, which provides 200
Watts of power. Fusion supports redundant, auto-switching power supplies; to implement this configuration with a
StudioEngine installation, you will need two Fusion power supplies.
The rear of the Axia Power Supply hosts a D-subminiature connector for 48 volts console power, and an EAC
connector for AC mains power input.
Console Modules
The Fusion surface is a mixing desk of modular design. The composition of the surface consists of a console
frame (which is available in multiple sizes), and an assortment of modules to suit your operational requirements.
Fusion Anatomy • 22
On the underside of every Fusion module are two parallel RJ45 connectors, which connect to the console’s
CANBus interface using the jacks on the CAN4ETH board (as described above). These connectors provides both
power and data.
Also on the underside is a rotary switch, which provides each module with a unique ID. This switch is typically
set at the factory, but may be easily changed should additional modules be required for console expansion.
There are several different styles of modules: 4-Fader, 2-Fader + Call Controller, Expert Monitor/Navigation
module, Standard Monitor/Navigation module (which also hosts two faders), modules with Intercom functions, and
User modules with programmable keys. Some of these are shown above.
What’s Next
Follow on as we dive into deeper configuration territory in the following chapters.
© 2017 Axia Audio - Rev 2.0
Chapter 2:
Configuration Basics
In this chapter, we look past the Quickstart to some of the powerful tools available to you with your Fusion
console — such as choosing and modifying Source Profiles, Show Profiles, and other essentials. Let’s dive in!
Working With Profiles
A standard console configuration step is defining the sources that will be used on the console.
Along with declaring the audio source (giving it a friendly name), we also must define how loading the source to
a fader will modify the operation of the console, and how the source in turn is modified by user interaction. Once the
console has sources configured, we can go one additional step and define channel layout and monitor settings.
The Axia terminology for these source settings is “Profiles”. There are two types of profiles:
1. Source Profiles, which define the different audio sources and how they function within the studio system, and
2. Show Profiles, which define what sources are placed on which console faders, and how the console’s
Monitor section is configured.
This chapter reviews some specifics of Profiles and in turn is the basic knowledge needed to configure the Fusion
surface. A description on each specific option is available in the Appendixes.
Source Profiles
A good way to understand Source Profile configuration options is to jump in and build a few common sources
that almost any studio would need, such as:
• Operator Microphone (Operator source type) – the Board Op’s mic
• CD Player (Line source type) – any basic audio source
• Caller (Phone source type) – a source that would require a mix-minus return
• Codec with specialized return feed needs (Codec source type with custom backfeed)
Although the type and number of source profiles that need to be built at any facility differs from the next, these
five samples represent the basic types of sources found in most studios; with this foundation (followed by a review of
the full Source Profile options found in the Appendix), you should be able to build profiles to satisfy all of your needs.
With that, connect a computer to your Fusion’s StudioEngine or PowerStation, type its IP address into your Web
browser, and follow along.
© 2017 Axia Audio - Rev 2.0
Configuration Basics • 23
• Guest Microphone (CR Guest source type) – an additional microphone in the same studio
To get started, select the Sources link, as shown here:
Configuration Basics • 24
Once you’ve done this, you’ll see any sources that have already been configured, as well as to “Create new
source profile” by clicking on the button at the top of the page.
© 2017 Axia Audio - Rev 2.0
Create the Operator’s Mic Source Profile
The Operator’s mic is intended for the operator of the Fusion console, so let’s create this first.
Configuration Basics • 25
Click Create new source profile. You’ll see the following screen on your computer:
© 2017 Axia Audio - Rev 2.0
Select “Operator” from the Source type drop-down list. In the Source name field, type a useful name, like
“Host” or “Board Op”, that the DJ can easily identify. This is the name that will be shown on the display at the top
of fader strip the source is loaded to.
In the Primary Source field, enter the board mic’s Channel Number. (Each audio source in the network has its
own unique Channel ID number.) If you know the number, just type it in; if not, use the Browse button to the right
of the field to select the source from your network.
With just these three basic options, you have enough for a working Source Profile! You could easily leave the
remainder of the options at their default values, but a couple of additional adjustments will help eliminate Control
Room errors.
In the Source Availability box, you have the option to control what faders and Monitor channels the source may
be assigned to. Uncheck the CR Monitor and ST Monitor assignment boxes — it’s not very likely you’d assign the
Operator’s mic directly to the monitors!
Press the OK button at the bottom of the page and your new Mic source is ready to use – you’ve just created the
Operator Mic Source Profile.
Configuration Basics • 26
NOTE: There should only be a single Operator source type loaded to the console’s faders at any one time.
This is because the Operator source type contains some preset logic functions specific to the type, such as
muting of the Control Room (CR) monitors when the mic is open, as well as being the default source for any
Talkback commands that are engaged. Therefore, any additional Microphone source should be one of the
other microphone source types.
© 2017 Axia Audio - Rev 2.0
Create a Guest Mic Source Profile
Configuration Basics • 27
The “CR guest” source type is intended for microphones located at guest positions within the same control
room as the Fusion surface. The built-in logic functions will mute the CR monitors when the source is turned on,
and provide GPIO logic for an optional Guest Control Panel that uses GPIO to remotely control Channel ON/OFF/
MUTE and Talkback functions. The steps for setup are similar to the ones outlined in the last section.
© 2017 Axia Audio - Rev 2.0
From the Source Type droptown, select “CR guest” and type in a friendly name of up to 10 characters.
In the Primary Source field, enter the mic’s Channel Number, either by typing it in or using the Browse button
to the right of the field to select the source from your network’s source list.
You can leave the rest of the options at their default settings. Press the OK button at the bottom of the page and
you’ve just created a Guest Mic Source Profile.
Other Helpful Options: Knob Function
Pressing the Channel Options encoder at the top of any fader strip opens the Channel Option screen on your
Fusion’s monitor; rotation of the knob then navigates you through the options available to you: selecting a different
source, applying EQ or Pan/Balance settings, and other adjustments.
However, the action taken when the board operator rotates the Options knob without pressing it can be tailored to
your studio’s preferred operating style via the setting in the Source Profile’s Knob Function dropdown box.
•
If PreAmp Gain Adjust is selected , the board op will be able to use the knob to quickly boost or cut the
level of the source at its Input stage to compensate for audio that’s too “hot” or too low.
•
If Fader Trim Level is selected, the control does not affect the Input gain of the source, but simply adjusts
the range of the fader itself. This is useful when one of a group of similar audio sources is higher or lower
than its siblings, and the operator wants to maintain a similar physical fader position.
Other Helpful Options: Default Backfeed
The other Source Profile option that’s important to microphone sources is found in the Default Backfeed
Options box.
In Axia parlance, “Backfeed” refers to any audio that’s sent back to an audio source (such as a microphone,
codec or phone caller) from the console. When pressing the Talkback key on a fader strip, the board-op’s mic is
routed, pre-fader, to the “backfeed” of that channel — an IFB function, if you will. This may be a Mix-Minus for
phone or codec sources, or a private headphone feed for microphone positions.
Configuration Basics • 28
The Dim gain setting defines the amount of cut, in dB, by which the Backfeed’s normal audio is adjusted, so that
the board-op can be better heard in the mic-user’s headphones.
Other Mic Profile Types
In addition to the Operator and CR Guest mic profiles, several other types are provided: CR producer, Studio
guest, and External microphone are also available.
CR producer is intended for a Show Producer’s mic loated in the same studio as the mixing console, so its
GPIO logic functions mute the CR monitors, and specialized GPIO functionality permits the producer to talk (using
Backfeeds, Axia’s name for internal foldback, or IFB) to any source with a Backfeed that is assigned to the Preview
(cue) channel.
Studio guest is used for mics located in an adjacent studio — for instance, a talk studio for stations hosting Talk
formats, or a music format with a morning show crew. The Studio guest source type logic mutes the Studio monitor
© 2017 Axia Audio - Rev 2.0
mix when the mic is turned on, and provides GPIO logic that permits optional control panels to control ON/OFF/
MUTE states, and make use of the Talkback channel to the Control Room board operator.
The External microphone source type is used for any microphone that will benefit from a headphone feed, and
located in a space that does not require monitor speaker muting.
Create a CD Player Source (Line Source Type)
The Line Source type is the basic source for inputs other than microphones. It will not mute the monitor
speakers when ON, and doesn’t require a Backfeed (mix-minus or IFB).
Setting up this Source Profile is just as easy as the previous two profiles.
From the Source Type dropdown, select “Line”. Enter a name for the device, such as “CD Player” or “CD 1”.
In the Primary Source field, enter the input’s Channel Number, either by typing it in or using the Browse button
to the right of the field to select the source from your network’s source list.
© 2017 Axia Audio - Rev 2.0
Configuration Basics • 29
Line Source is perfect for creating Source Profiles for devices such as CD players, DAT decks, Satellite
receivers, PCs using standard audio outputs, etc.
Now click the OK button at the bottom of the screen; your CD player is configured and ready to be assigned to a fader.
Note that we left the “ST Monitor” box checked in the Source Availability section. Doing this allows the board
op to assign the CD player to the monitors in the adjacent studio for direct auditioning.
A more practical application for this option is when creating a Source Profile for an Air Monitor, so that talent
can directly monitor the over-the-air broadcast signal. To do this, you’d create a Line source type for the air monitor
receiver, and uncheck all the Channel availability boxes except the CR Monitor and ST Monitor. This way, the overthe-air signal can feed the monitors — but not be assigned to a fader and sent back to the transmitter!
Another check box that’s useful for Line source types is the one marked GPIO ready enabled. This controls the
OFF lamp on the fader strip that the source is assigned to. Some operation practices require an indication of source
readiness; when this box is checked, the fader’s OFF lamp will only illuminate when the device provides a “ready”
logic state.
Configuration Basics • 30
Many professional CD players provide GPIO closures for such states, as well as most modern Automation
systems, but make certain that you understand this feature prior to enabling and, that your device really does support
a “Ready” indication. If it doesn’t, your operators may think the OFF lamp is broken because it never illuminates!
© 2017 Axia Audio - Rev 2.0
Create a Telephone Source (Phone Source Type)
Configuration Basics • 31
Putting phones on-air is one of the basic operations of the modern studio. Fusion’s Phone Source Type helps
ease the task of handling outboard phone hybrids.
© 2017 Axia Audio - Rev 2.0
First, select “Phone” from the Source Type dropdown, and some phone-specific options will appear.
In the Source Name field, type the name you’d like to display on the Fusion’s channel display.
Now, to define the Phone controls. Select from one of the following options that best suits your studio’s phone gear:
•
No Phone Control. This is used for Telos Hx products, or for non-network-controlled hybrids from other manufacturers. You’ll still be able to control the hybrid via Fusion’s GPIO capabilities.
To do so, scroll down to the Hybrid Answer Mode dropdown box near the bottom of the screen. Select either
“Channel ON answer hybrid” or “Channel ON or Preview ON answers hybrid”.
In the first case, when the Phone source is assigned to a console fader strip, turning that channel ON picks up
the phone. In the second case, your operator may either turn the channel ON or use the channel’s PREVIEW
(cue) key to answer the phone.
•
EU Phone allows you to map specific lines/hybrids from a Telos Multi-Line phone system (such as TWOx12,
Nx12, Nx6 or VX) to a single fader.
•
In the example shown above, we’re mapping a hybrid from a VX Broadcast VoIP system, using the sections
highlighted in red:
»» Type in the IP address of the VX Server into the Server IP box.
Configuration Basics • 32
»» Select the VX radio button.
»» Enter the Studio name configured in your VX system into the Studio Name box.
»» Enter the number of the VX hybrid from the specified VX Studio into the Fixed Hybrid box.
»» Note: If the Server IP has authentication requirements, the authentication syntax in the Server IP
field is username:[email protected] ADDRESS . For example, user:[email protected] attempts to
log into the specified IP address with username user and password test.
•
US Phone ties the Source Profile to a Telos Call Controller module installed in your Fusion console. These
modules work with all modern Telos multi-line phone systems. Enter the Hybrid number to tell the fader which
hybrid the Source Profile controls.
© 2017 Axia Audio - Rev 2.0
The final step is to define how you want to handle mix-minus. Fusion (and all Axia consoles) automatically
generates mix-minus (N-1, “clean feed”) for each phone caller taken to air. To configure this, you’ll scroll to the
Default Backfeed Options box and select the desired audio mix from the Feed to Source dropdown.
Nine different Manual Backfeed mixes plus an Auto smart mode are available.
The default option is Auto. Choosing this option eliminates manual mix-minus building by switching the source
of the mix-minus based on the ON state of the fader the Phone source is loaded to. When the fader is in the OFF
state, the caller hears the off-line PHONE mix. The moment the channel is turned ON, the audio feeding the caller
switches to the Program 1 bus, minus the caller’s own audio.
Here are the manual options:
•
PGM-1 through PGM-4 feed the caller the output of the selected Program bus, minus their own voice.
•
AUX A through AUX D feed the caller the output of the selected Aux Send bus, minus their own voice.
•
PHONE is a mix designed to harmonize with typical, traditional radio operations. A channel is assigned
to the PHONE mix by selecting the fader strips PGM 4 key; the PHONE mix is then created pre-fader and
pre-ON/OFF, so that engaging any PGM 4 button will send the audio of that channel to the phone mix at
Unity gain.
We suggest leaving the selection at its default, Auto, unless special circumstances dictate otherwise. No matter
what you choose here, the board operator can quickly override it and select a different function, using the Options
knob at the top of the fader strip.
When your setup is complete, click the OK button at the bottom of the screen.
Advanced Stuff: A Codec Source With Custom Backfeed
If you’ve followed along and created a Phone Source Profile as described in the previous section, creating a
Profile for a codec will appear similar (minus the phone hybrid controls, of course). If the codec is bi-directional,
your codec Source Profile will need a Backfeed configuration as well.
By now, you’ll have noticed that, when you begin to create a Source Profile from scratch, all profile types have
either “Disabled” or “Default” displayed in the Feed to Source option field.
But your operation might have needs which require a more specialized Backfeed than the Default behavior. And
so we provide the “Custom” option, which allows high-level control of mix-minus behavior based on channel state
logic. Here’s the section of the Source Profile which pertains to this Custom Feed to Source functionality:
© 2017 Axia Audio - Rev 2.0
Configuration Basics • 33
The Default Backfeed behavior of the Codec provides a mono sum to both the Left and Right channels on the
return audio or named Backfeed.
However, when the Talkback key on the channel strip the codec is assigned to is pressed by the operator, the
board op’s mic audio (pre-fader) is routed to only the left channel; the right channel remains unchanged.
As you can see from examining, the Custom option provides significantly enhanced backfeed routing options,
including independent control of the L and R sides and a completely different IFB signal insertion point.
External Feed Src allows you to pick any audio channel in your Livewire network to send to your source’s backfeed.
Configuration Basics • 34
IFB Gain allows you to tailor the volume of the IFB audio with a cut or boost of up to 25 dB. Feed Volume
allows you to reduce the volume by up to 100 dB for either or both sides of the stereo return channel, respectively.
Feed Mode enables you to pick from a variety of Backfeed styles.
•
M-1 Mono Sum/Left/Right sends a mix-minus of the stereo Program output, minus the source, as either a
summed Mono signal, or as the Left or Right channel of the stereo Program output, minus the source.
•
Full Mix Left/Right/Mono Sum disables the Mix-Minus and supplies the complete mix to the Backfeed.
You may pick the right or left channel, or choose a summed Mono signal.
© 2017 Axia Audio - Rev 2.0
•
Notice that this option is active for both sides of the stereo Backfeed channel, so that you can completely
customize the style of the audio flowing back to your remote user.
Next come a series of Feed Source options that can be applied to any or all of 6 different console fader channel states
to completely customize backfeed based upon the logical state of the fader strip itself. These logical states include:
•
In Standard Mode:
»» While Channel is OFF
»» While Channel is OFF, But in Preview
»» While Channel is ON
•
In Record Mode:
»» Record Mode: While Channel is OFF
»» Record Mode: While Channel is OFF, But in Preview
»» Record Mode: While Channel is ON
Configuration Basics • 35
Note: Record Mode is a special “macro” mode that helps talent record audio for later use with a single
press of the Fusion console’s “Record” key, located on the Monitor Module. We’ll cover use of this function
in later chapters.
© 2017 Axia Audio - Rev 2.0
• Using the Feed Source dropdown boxes, you can pick the Backfeed source that
will be fed to the Codec’s IFB channel in each of the six possible channel states
noted above. Your choices are:Disconnected. Use this when you want to disable the
Backfeed (send no audio) for a particular channel state.
• Program 1 – Program 4. Sends the audio from the selected Program bus.
• Record. Sends audio from the console’s Record bus. The Record bus is a special
variant of PGM4. The Record mix is postfader and pre-On/Off, to provide offline
recording with volume control.
• Phone. Fusion has an off-line Phone bus that is actually a special variant of
PGM4. The Phone bus is mono-sum, prefader and pre-on/off to allow speaker-phone
style operation thru the Operator’s mic. Selecting Phone feeds the Phone bus, minus
the source, so that the listener can hear other Phone callers who are waiting in the air
queue.
•
Aux Send A – Aux Send D. Sends the audio from the selected Auxiliary mixing bus.
•
Preview. Sends the audio from any sources assigned to the Preview (cue) bus.
•
Studio Monitor. This is the source typically sent to Guest Studio monitors & headphones by the Control
Room board op. It is assigned using the “Studio Monitor” controls on the Fusion Monitor Module.
•
External. Sends the audio from the channel you specified in the External Feed Src. Box at the top of the
Custom Backfeed Options box.
Configuration Basics • 36
So, say you want to construct a Custom conditional backfeed for your Codec that sends the CR Monitor when
the channel is OFF, the contents of the Record bus when the channel is OFF but assigned to Preview, and a PGM-1
mix-minus when the channel is ON, you’d set it up like so:
© 2017 Axia Audio - Rev 2.0
Naturally, this level of complexity is not necessary in every radio station, but is available for specialized
situations where highly-tailored Backfeeds are required.
Show Profiles
Now that you know how Source Profiles work, let’s talk about Show Profiles.
Show Profiles are, essentially, collections of Source Profiles. A Show Profile keeps track of what sources are
loaded to each of your console’s faders. Using Show Profiles, each user can have the board set just the way they
like it — sources placed where they’re most useful, monitors set to the appropriate feed, headphones conforming to
personal preference.
You can also use Show Profiles to define different types of broadcasts – one for the morning show, one for
talk segments, one for musical guest interviews, one for unattended operation – that instantly recall your saved
configuration when loaded.
How to Create A Show Profile
The easiest way to create a Show Profile is to set up your Fusion console for a show, then save a Show Profile for
it by taking a “snapshot.”
•
Get started by assigning a source to each fader strip using the Options knob at the top of each fader strip,
and selecting a source from the Current Source selection box.
•
Once that’s done, assign each source to the Program bus you want it to feed, using the PGM keys at the top
of each fader strip.
•
Finally, make your Monitor and Headphone audio choices using the keys on your console’s Monitor Module.
Simply click on the Capture show profile link, and you’ll be prompted to name your new Show Profile. Type in
a name, click “OK”, and you’ve got a new Show Profile that can be loaded by pressing the Profile key at the top of
your console’s Monitor module.
© 2017 Axia Audio - Rev 2.0
Configuration Basics • 37
Now that your console is set up the way you want, connect a computer to your Fusion’s StudioEngine or
PowerStation, type its IP address into your Web browser, and select the Shows link, as shown here:
The other link shown here, New show profile, allows you to construct an entire Show Profile completely from
scratch using your computer’s on-screen controls. This is the “expert” way of making a show profile, which allows
access to settings unavailable to the console operator. We’ll cover these options in detail later on.
You can save up to 99 different Show Profiles to make console reconfiguration fast and easy. Don’t worry –
loading a Show Profile will never take an active audio source off the air: any changes to a fader’s source assignment
are queued until after the OFF key for that fader is pressed.
What’s Next
Configuration Basics • 38
With configuration basics covered, the next chapter moves on to Console Operations – a tutorial on the controls
the board operator will use to run the Fusion console.
© 2017 Axia Audio - Rev 2.0
Chapter 3:
Operation
Fusion was designed with speed in mind. On-air studios are fast-paced operations, and our design goal is for
every control, display and interface to enhance your Operator’s speed and accuracy.
Because different people work differently, we’ve made it possible to access some functions in multiple ways.
This way, the board operator can work the way they want — not the way somebody else thinks they should work!
This chapter will start with an overview of Fusion’s displays and controls to help familiarize you. Then, we’ll
have a look at some of the more specialized options.
Main Display
1. Main Meters. There are 6 bargraph-style loudness meters on-screen at all times. By default, they meter the
PGM-1 – PGM-4 mixing buses, plus the Preview (Cue) channel and the Monitor feed. You can change
the sources of the 3rd and 4th meters if you like.
2. Analog-Style Clock. This time-of-day clock can be slaved to a house NTP server. When using the built-in
Count Down timer, the inner gray ring turns red to indicate seconds left until Zero.
© 2017 Axia Audio - Rev 2.0
Operation • 39
This is the main Fusion display screen. 99% of the time, this is what you’ll see on your Studio display.
To augment Fusion’s metering capabilities during very busy programs, the Analog Clock can be replaced
with 4 more meters, which can be pre-defined in Monitor Settings page of any Show Profile.
3. Count Down / Count Up Timers. The Count Down timer is an interval timer that counts down to Zero
from any starting point set by the operator; useful for back-timing network joins or other precision programming shifts. It is manually started.
The Count Up timer is an elapsed-time timer that counts up from Zero. It can be set to start manually, or
start/stopped/reset automatically based upon the ON and OFF keys on channels you’ve defined.
4. Annunciator Strip lets board ops know when Mic channels are open, when audio is being auditioned in
the Preview (cue) channel, when Talkback is active in either direction, and when Fusion’s one-touch Record Mode is active.
5. Main Monitor Assignments / Confidence Meters let ops know what sources are being heard in the Control Room and Studio Monitor and Headphone feeds, and in the Preview channel. Adjacent confidence and
set-level meters display show set levels of the physical controls and incoming signals.
6. EXT 1 / EXT 2 Confidence Meters. These meters show the sources of audio assigned to either of the two
External Monitor selection keys, and audio levels of those sources.
7. Profile Window. Displays the name of the active Show Profile.
Operation • 40
8. Day/Date/Time Displays. Shows the current date and time in digital format.
© 2017 Axia Audio - Rev 2.0
Basic Channel Controls
Here’s the 4-Fader Module, the basic building
block of any Fusion console. The controls should
be familiar to any radio pro:
1. Channel Display. This window displays the
audio source assigned to the fader, plus confidence meters for incoming audio. If the operator
presses the Talkback key at the bottom of the
fader strip for any source with a Backfeed (mixminus or Talkback channel), the display will
note this, and supply confidence meters for the
Backfeed audio too.
2. Soft/Options Knob. Press this knob to enter
Channel Options mode and select the audio
source, adjust EQ, Mic dynamics, Pan/Mode
and send audio to any one of the four Aux Send
buses.
Rotate the knob without pressing to adjust fader
gain trim, or source Preamp gain (depending
upon the preset choice).
3. Main Bus Assignment Keys. Quickly
assign the source to any or all of the four main
Program buses.
4. Channel Fader. You know what this does,
right? Right!
5. A/B Keys. “Soft” keys that can be programmed to send routing or GPIO commands
via Axia Pathfinder routing products.
7. Talkback Key. Sends pre-fader audio from the board operator’s Mic channel to an audio source that has
a return audio path, such as a remote codec, Phone caller or Studio mic with a dedicated headphone feed.
Used in conjunction with the Preview key, the operator can easily have an off-air conversation with remote
talent or guests.
8. ON/OFF Keys. Turns the channel on and off.
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Operation • 41
6. Preview Key. Sends audio to the Preview (Cue) bus for off-air auditioning.
Expert Monitor Module Controls
Here’s the Expert Monitor Module. This is a dualwidth module with direct-access keys to every monitor and channel option function on the console.
1. Multi-Function Knobs. When advanced
controls are active, the center clock in Fusion’s
display changes to indicate the active function,
and the action of each knob when pressed or
rotated.
2. Channel Options section. When you press
a Soft/Options knob at the top of any fader strip,
pressing any of these keys takes you directly to
that option.
»» Source lets you change the audio source
assigned to the fader.
»» EQ & Dynamics adjusts the EQ curve
and Mic compression / expansion settings (if
allowed).
»» Pan & Mode lets you adjust pan/balance,
choose Stereo or Mono signal mode, and correct
signal phase issues.
»» Aux Sends permits assignment of the audio
source to any or all of the four Aux Send buses.
»» Feed To Source gives the operator manual
control over mix-minus feed, if present.
3. User Keys. “Soft” keys that can be programmed to send routing or GPIO commands
via Show Profile configuration or Axia Path-
Operation • 42
finder routing products.
4. Preview Controls. Adjusts the volume of the Preview (cue) speaker; turns Preview To Headphones on and off.
5. Studio Monitor Controls. Adjusts volume of the Studio Monitor speakers, and allows the board op to
Talkback to studio guests through the Studio Monitor feed.
6. Link Key. When unlit, operators can choose separate sources for the CR Monitor and CR Operator Heaphones. When lit, the same audio is fed to CR Monitors and Headphones.
© 2017 Axia Audio - Rev 2.0
7. Control Room Monitor controls. Choose any of the four Program buses, four Aux Sends or two External
sources to feed the CR Monitor speakers. Sources assigned to External keys are set in Show Profile, but
pressing either key for 5 seconds allows the operator to change the sources assigned to the External key
on-the-fly.
8. Control Room Headphone controls. Same controls as provided for the CR Monitor speakers, but affects
CR Headphone feed.
9. Global Options section. Unlike Channel Options keys (which affect an active fader strip), Global Options
affect the entire console.
»» Profile displays the entire list of pre-saved Show Profiles that the operator can load. Press once to
display the list; press-and-hold to restore the active Show Profile to its “saved” state after changes
have been made.
»» Monitor Options let the board op adjust Talkback gain in monitors, built-in Headphone Processing
settings, and choose Stereo or Mono monitor modes.
»» Meter Options lets the operator choose meter ballistics, and activate the “More Meters” display of
up to 4 additional onscreen meters, whose sources preset by the Engineer.
»» Sends & Returns key lets operator adjust the master volume of Fusion’s four Aux Send buses, and
volume and Program bus assignments of the two Aux Return buses. Press once to work with Sends,
twice for Returns.
»» Additional Options: Reserved for future expansion.
10.Timer controls. Affects options of the Count Down and Count Up onscreen timers.
»» Start/Stop, Freeze and Reset control the actions of the Count Up timer.
11.Record Mode Key. Puts the console into “Record Mode” – a macro function defined in each Show Profile.
Activating Record Mode allows changing control room and studio Monitor and Headphone sources to predefined bus selections, start and stop an associated recording device, and change meter options, easing onthe-fly off-air recording of phone callers or remote talent. Pressing once invokes Record Mode; a second
press returns the console to its previous state.
12.Keypad and Nav Controls. At the bottom right of the Expert monitor module is a Control Knob surrounded by a set of navigation keys (Up, Down, Right and Left), an Enter key, and a Help key. These controls
provide an alternative to the Multi-Function knobs for navigating Fusion option screens. Operators can use
the directional keys to navigate through on-screen lists, and the Enter key to “take” a selection. The Control
Knob can also be used to scroll through on-screen lists. The keypad is used for dialing Telos phone systems
or Codecs attached to the Livewire network.
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Operation • 43
»» Options lets the operator adjust the duration of the Count Down timer, and begin its countdown. You
can also switch the Count Up timer between Manual, Auto Reset and Auto-Add modes.
Standard Monitor Module Controls
For studios where expert monitor controls
are not needed, the Standard Monitor Module is a space-saving design that incorporates two faders in addition to the numeric
entry/dial pad and basic Monitor/Headphone controls.
1.
Channel Display and MultiFunction Knobs. Contains displays for
faders, plus knobs that can be pressed or
rotated to change options when advanced
functions are active.
2.
Numeric Entry Keypad. The keypad is used for dialing Telos phone systems or Codecs attached to the Livewire
network.
3.
Profile Key. Displays the entire list
of pre-saved Show Profiles that the operator can load. Press once to display the list;
press-and-hold to restore the active Show
Profile to its “saved” state after changes
have been made.
Operation • 44
4.
Record Mode Key. Puts the console into “Record Mode” – a macro function defined in each Show Profile. Activating Record Mode allows changing control
room and studio Monitor and Headphone
sources to pre-defined bus selections, start
and stop an associated recording device,
and change meter options, easing on-thefly off-air recording of phone callers or
remote talent. Pressing once invokes Record Mode; a second press returns the console to its previous state.
5. Timer controls. Affects options of the Count Down and Count Up onscreen timers.
© 2017 Axia Audio - Rev 2.0
6. Monitor Options Key. Pressing this key once allows the operator to adjust Monitor and Headphone behavior using the Multi-Function Knobs; pressing the key twice gives access to Meter Options. Settings include:
»» Adjustment of the Monitor Talkback and Dim levels.
»» Dynamics processing for the Headphone feed.
»» Whether the Preview channel is fed to a single headphone channel, both channels, or not at all.
»» Monitor Mode: Stereo, Left, Right, or Summed.
»» Meter Ballistics and Peak Display options.
»» Aux Send and Aux Return assignments and level controls.
7. Control Room and Studio Monitor controls.
»» Control Room Monitor controls allow the operator to choose any of the four Program buses, four
Aux Sends or two External sources to feed the CR Monitor speakers. Sources assigned to External
keys are set in Show Profile, but pressing either key for 5 seconds allows the operator to change the
sources assigned to the External key on-the-fly.
»» Rotating the Studio Monitor knob allows the operator to control the volume of the speakers in the
associated studio. Pressing the knob allows the board op to select the source feeding those speakers.
»» Pressing Talk To Studio allows the board op to Talkback to studio guests through the Studio Monitor feed.
8. Preview and Headphones Controls.
»» The Preview knob adjusts the volume of the Preview (cue) speaker.
»» The Preview to HP key turns Preview To Headphones on and off.
Operation • 45
»» Rotating the Headphones knob varies the volume of the studio Headphone feed.
© 2017 Axia Audio - Rev 2.0
Call Controller + 2-Fader Phone Module
This module integrates control of Telos Hx6, Nx12,
Nx6, TWOX12, Series 2101 and VX telephone systems
right into the console, and features an integrated Telos
Status Symbols™ to give instant information about and
control over incoming phone calls.
All fader strip controls operate the same way as described previously for the 4-Fader Module, so let’s look
at the phone-specific controls, which mimic the functions
of standalone Call Controller and VSet phone controllers
used with your Telos phone system.
1. Xfer Call key lets the board operator transfer a call
from the Fusion Phone module to a local handset for
off-console control or off-air conversations.
2. Block All key “busies out” all inactive phone lines,
blocking incoming calls in preparation for on-air contesting.
3. Line Selection Keys. There are two columns of
keys flanking the Status Symbols display; each column
controls a separate phone system hybrid (for phone systems so equipped). Pressing a key “takes” the adjacent
line; that line can then be screened, placed in a queue,
dropped or transferred off-console. Refer to your Telos
phone system Operation Manual for the specifics of
how these controls work.
4. Status Symbol display. These symbols are visual
representations of line and caller status. Refer to your
Telos phone system Operation Manual for the meaning
of these displays.
Operation • 46
5. Drop Keys. When a line is activated using the Line
Selection keys, pressing this key disconnects the selected line. One Drop key is provided per hybrid.
6. Next Key. When a show producer has established an On-Air Queue using their Telos phone controls or
software, pressing the Next key automatically drops the current call and takes the next queued call to air.
© 2017 Axia Audio - Rev 2.0
IP Intercom Module – 20-Station
This module provides on-console integration with the Axia IP Intercom system,
and gives access to 20 preprogrammed intercom stations, plus dial-up access to stations systemwide that have not been programmed to one of the speed-dial stations.
The Operator’s mic is used as the local talk source; the Preview (cue) speaker is used
to listen to other stations.
1. Preset Station Displays show what preset Intercom stations are assigned; useful
for frequently-called stations. Each display provides info for two stations; associated
Talk and Listen keys are above and below each display.
2. Talk key. Press to talk to the station indicated by the display. The adjacent LED
tally illuminates when the Talk key is pressed, or when the station at the other end is
listening to you.
3. Listen key. Press to listen to the station indicated by the display. The adjacent
LED tally illuminates when you are listening to the far-end station. If this tally
blinks, you must press it to allow the calling station to talk to you.
4. Non-Preset Station Display and Talk/Listen Controls. This display and Talk/
Listen controls are used to “dial up” stations not assigned to one of the 20 presets
below. To select a station to call, use the Select knob below.
5. Non-Preset Station Selectors and Group Call key.
»» Rotate the Select knob to scroll through a list of available Intercom stations on
the overbridge display above.
»» Tap the Assign key to assign the selected station to the overbridge Talk and
Listen keys.
»» Tap the Group Call key to call multiple stations with a single button press.
Please refer to the IP Intercom User Guide for details on how to use this function.
»» Press the Mute Mic key to mute the Operator’s mic feed to the IP Intercom system. The LED tally
will light when active.
»» Press the Mute Speaker key to mute IP Intercom calls to your console. The LED tally will light
when active.
»» Rotate the Volume knob to adjust the level of incoming Intercom calls.
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Operation • 47
6. Mic and Speaker Mute keys, and Volume control.
IP Intercom Module – 10-Station Filmcap
The 10-Station IP Intercom module gives access to 10 pre-set Intercom stations. As with
the 20-Station module, the Operator’s mic is used as the local talk source; the Preview (cue)
speaker is used to listen to other stations.
1. Preset Station keys. These stations are set by the Engineer, and the buttons labeled with
the appropriate station name. Press to talk, release to stop talking.
2.Mic and Speaker Mute keys, and Volume control.
»» Press the Mute Mic key to mute the Operator’s mic feed to the IP Intercom system. The
LED tally will light when active.
»» Press the Mute Speaker key to mute IP Intercom calls to your console. The LED tally
will light when active.
»» Rotate the Volume knob to adjust the level of incoming Intercom calls.
Switch Modules (not shown)
Fusion modules that provide remote control of studio devices and audio routing
functions are available. They are programmed for specific functions by the studio engineer.
• Film-Cap Switch Modules are pre-programmed to send closures to studio devices; useful
for starting and stopping recorders or sending routing activation closures to Axia PathfinderPC routing tools. One function per key is available.
Operation • 48
• SmartSwitch Modules have color screens built into each keycap, and can dynamically
display text and change color to indicate changing functions. These can be programmed
with “pages” of functions to offer many more functions than just the 10 keys available.
They may also be programmed as context-sensitive controls for routing salvos using PathfinderPC.
What’s Next
In the next chapter, we’ll learn about the use of Fusion’s powerful Virtual Mixer (VMix) capabilities, which allow
you to create custom mixes of networked sources and use them like you would a single audio source. We’ll also look at
Virtual Mode (VMode) functions, which allows you to manipulate incoming sources in many useful ways.
© 2017 Axia Audio - Rev 2.0
Chapter 4:
VMix and VMode
There’s a lot of DSP processing horsepower in Axia mixing engines. Rather than let that go to waste, we’ve used
it to power two tools that many broadcasters have found to be indispensible: VMix and VMode.
VMix (short for Virtual Mixer) provides up to 16 channels of “virtual mixing”, which can be used to pre-mix
up to 5 audio sources each for presentation on a single physical fader (or software fader, if Axia SoftSurface softconsole software is used). VMix works completely independently of the Fusion surface. In addition to static control
of VMix through its web pages, Axia’s Pathfinder routing control tools can also be used to dynamically control
VMix and create mixing functions based on a variety of system-wide parameters.
VMode (short for Virtual Mode) can be used to convert channel count, channel order, and/or channel content
between audio streams at its input and output. For example, passing some channels while cutting others, summing
stereo to mono, upmixing or downmixing between stereo and multi-channel, combining channels from two inputs
into one output. Select audio as input, define the “Mode” (conversion type, or in other words, the routing within an
internal matrix), and define the output as a network source into the AoIP network. There are 16 VMode instances in
a Fusion engine.
Virtual Mixing with VMix
What’s it all about?
In addition to the regular mixing capabilities of your Fusion console, there is an 80-input “virtual” mixer
accessed using the console’s HTTP interface (or with PathfinderPC routing tools).
This mixer consists of 80 stereo input channels, a direct output for each channel, 16
submixer outputs, and one master out. The 80 channels are divided equally among the 16 subgroups, providing 5
stereo channels feeding each subgroup mixer.
To understand this concept, think of VMix as a standalone piece of hardware. If you visualize wiring an external
line mixer to your network, Livewire audio sources would be connected to the mixer inputs; your VMix outputs then
become mixdowns that you can use anywhere else on the network, just like any other audio source.
To access VMix setup, open a browser on a computer connected to your Axia network and browse to the IP
Address assigned to your PowerStation or StudioEngine or, if you have Axia iProbe, open the UI page by rightclicking on the device’s icon in the left pane. Then, choose the V-Mixer and V-Mode item from the navigation bar.
The following image is an excerpt of the VMix/VMode setup screen. It shows the VMix Main and Sub-Mixer 1
controls. These sub-mix controls are duplicated for all 16 VMix submixers.
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VMix and VMode • 49
The various VMix outputs described above are sources that can feed your Livewire network and they can
be manipulated in the same manner as any other audio source. A VMix source can be applied to a console fader,
assigned to an audio node destination, or monitored by Pathfinder.
Note that adjustments made to VMix take effect as soon as you apply them — so changes saved “on the fly” will
affect your output streams immediately.
VMix Main Controls
The VMix Main output provides a summed mix of all 16 submixes. Unless you need a single output that
combines the audio from of all the submixes, you may leave this disabled — it doesn’t need to be enabled for the
submixers to work. There are only a few settings for this control:
• Out Name. You can enter a friendly name for the VMix Main output, which will be displayed as a source
name on your Axia network.
• Out Stream Type. Choose from Live Stereo, Standard Stereo, or leave it Disabled if no Main stream is
desired.
• The Status window will normally display “OK” when the stream is enabled.
• The Audio box will be green when audio is present.
VMix Submixer Controls
VMix and VMode • 50
In most cases, the VMix Subs are the only channels you will need to enable, since each Sub has its own direct
output. Only enable Submixes you intend to actively use; active submixes without any activity clutter up your
network with empty streams.
First, you’ll want to enable the Submixer you’re working with. At the top of the section, you’ll see (in this case)
Submix 1 displayed on the screen. Next to that are controls for:
• Gain. Set this at whatever output level you want the submix output stream volume to be.
• Channel. All Livewire audio streams are assigned a Channel number; put a unique value in this box.
• Out Stream Type. Choose from Live Stereo, or Standard Stereo. Disabled turns off the submixer.
© 2017 Axia Audio - Rev 2.0
Each VMix Sub submixer input includes an on/off setting, a gain setting, and automatic fade-up/fade-down time
parameters. Using one is easy; let’s walk through the steps:
• In the Src Name box, enter the name of the source you’ll be assigning to the input.
• In the Channel box, enter the unique Livewire channel number of your audio source.
• In Stream Type will normally be set to “From Source”, meaning that the source itself (a mic, CD player,
etc.) will be providing the audio. However, you can select “To Source” to use the source’s automatically
generated Backfeed (mix-minus audio) as in input.
Example: Selecting “From Source” when a phone hybrid is assigned as a VMix Sub input would use the
caller audio; selecting “To Source” would instead use the Mix Minus sent to the hybrid.
• Select the Enable box next to the input to turn the Submixer input on.
• Leave the Fade Time boxes set at their default.
The Fade Time function won’t be used in normal operation, but can be used to create cross fades between
sources when Pathfinder is dynamically making changes to the VMix.
If, say, 1.0 is entered in the first box, the submixer channel’s audio will rise from -∞ to the Gain value set in
the next field in 1.0 seconds. If the field is set to 0, the audio will simply turn on at the gain value specified.
The second box works the same way, but controls the ON to OFF fade time.
• In the Gain box, enter the setting, in dB, that you want for the Submixer input. Each input has its own
individual Gain setting.
That’s all you need to do to configure a VMix Submixer input stream. You can configure up to 5 input streams
per VMix Submixer.
At the bottom of each of the submix sections is an Apply button. Any changes you make will be saved when this
button is pressed. Be sure to save the changes for each submixer as they are configured.
In most cases only the Submix output itself needs a unique channel number, but if you so desire, each VMix
input can also be sent back to the network as a unique source, post the ON/OFF and gain stage of the VMix. Some
users find this useful for constructing “cascaded” mix channels to suit unique situations. To do so, enter values for
these controls:
• Out Name: The name you give the post-on/off submix channel to send back to the Axia network.
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VMix and VMode • 51
VMIx Submixer Advanced Options
There are some additional options provided for advanced users.
• Channel: the channel number you assign to the post-on/off submix channel to send back to the Axia network.
• Out Stream Type. Choose from Live Stereo, Standard Stereo to enable the direct output stream for this
VMix “fader”. Disabled means no stream will be generated.
Be sure to click Apply after you’ve finished making any options changes.
Some VMix Examples
Now that you know how to enable and set up VMix submixers, what might you do with it? Here are some
examples.
A Mix of Sources to Monitor
Some facilities may need to monitor one or more sources in addition to program audio, like the “squawk
channel” some satellite feed providers use to relay announcements.
If you wanted to monitor this “squawk” audio on your Preview speaker without taking up a fader assignment,
you could create a mix of the “squawk” source and the Preview mix from the engine. The VMix Sub Channel would
be the audio source that you would route to the Preview speaker.
Virtual VMix Control
Using third-party products like VMix from BSI, or by taking direct
control of a console’s mixing engine with Axia SmartSurface software,
VMix channels can be assigned to a “virtual control surface”, giving
studio talent or a producer direct control of VMix without the need for
a physical console.
Combining VMix with Pathfinder Routing Control
VMix and VMode • 52
Axia Pathfinder routing tools can be used to “control” VMix in two
different modes.
First, as a background controller, Pathfinder can monitor Livewire
system parameters or receive commands from external devices like
satellites, button panels, or automation systems and react to them
by changing the state of VMIX ON/OFF, Gain, Time Up, and/ or
Time Down fields. This provides many different possibilities for
facility automation, Intercom functions, or whatever else you might
imagine. For example, the combo of Pathfinder and VMix could
duplicate the function provided by other products that are controlling
audio switching in many radio facilities. (Refer to documentation on
Pathfinder for further information.)
© 2017 Axia Audio - Rev 2.0
Second, there’s Pathfinder’s VMix Control feature. This is a software fader control option that is provided with
the PRO versions of Pathfinder. VMix Control brings the operation of VMIX out of the background and provides a
graphical user interface with software faders, as shown here.
There are several other ways that Pathfinder can be used for background control of VMix. VMix functions can
be used both as qualifiers and actions in routing salvos. This means that a designer can select GPIO triggers, time
based events, user button pushes, serial port commands, and other options and combinations of options to decide
when to make changes to any Virtual Fader in a VMix. The user can make a gain change based on these events,
turn a channel off or on, and or adjust the fade times, giving complete control over the VMIXer based on any of the
stacking events qualifiers.
Finally, Pathfinder’s Software Authority protocol translator includes commands to control any VMix fader that’s
active, so any machine that can send user defined serial or TCP commands can also control and read VMix functions
through Pathfinder.
GPIO control of your VMIX with Pathfinder
Imagine that you have a night jock that should monitor all four radio stations in your cluster. To help make sure
this actually happens, you could send all four off-air signals as sources into a VMix submixer, and take the output
of that submix to a monitor. A Fusion accessory panel or external button wired to a GPIO port could then provide a
“press and hold” function to allow the jock to monitor the sources momentarily. (This example is only possible with
Pathfinder control of VMix.)
© 2017 Axia Audio - Rev 2.0
VMix and VMode • 53
Using these techniques VMIX can be used as a fully automated virtual mixer in the background of each console.
Manipulating Streams With VMode
Each of the 16 VMode instances supports audio streams with up to 8 channels at input and up to 8 channels at
output. Between the input and output, there is a matrix, where the conversion is performed, according to the selected
audio routing option (“Mode”). Livewire 8-channel streams carry 5.1 content in channels 1-6, and stereo content in
channels 7 and 8, and the matrix is partitioned the same way, to fit them natively:
• Channels 1 to 6 of the matrix form a multi-channel part
• Channels 7 and 8 of the matrix form a stereo part
Routing option (“Mode”) names refer to the parts of the matrix and conversions performed between them. They
do not refer to channel maps of the input/output streams.
For receiving and transmitting, two stream classes are distinguished:
• Class “mono/stereo”: streams with 1 or 2 channels
• Class “multi-channel”: streams with 3 to 8 channels
Input streams put their content into, and output streams take their content from the stereo or multi-channel parts
of the matrix, according to their mono/stereo or multi-channel classes. Multi-channel streams with 7 and 8 channels
span across the both parts of the matrix, starting from channel 1, in the natural order. Livewire 8-channel “Surround”
streams fit the matrix mapping without reordering.
VMix and VMode • 54
The VMode view provides status of the current configuration.
To change the configuration, select the far right “Configure…” button.
© 2017 Axia Audio - Rev 2.0
Input:
The Input selector is used to select any source native to the Engine or select an “EXTERNAL” source from the
network.
Source name is a text field used to document the input.
Address is used to define the Livewire channel number of an “EXTERNAL” source or the address of an AES67
stream.
Stream type is used to define the source. With AES67, the stream could be a linear 24 bit up to 8 channels or linear
16 bit. The Livewire type sources are identified. Make sure you select the correct type.
Mode:
Manipulation within the matrix is referred to as routing and will be configured by a mode selection. Two stream
classes are utilized within the different mode designations:
• Class “stereo”: streams with 1 or 2 channels (“Left” or “Right” may also be used in this class)
• Class “multi-channel”: streams with 3 to 8 channels
• Unity gain in all channels
• No up-/down-mixing
• Preserving the channel order regardless of the actual number of channels in the input and output streams
• No signal in output channels, where corresponding channels do not exist at input
Mode option “Pass stereo” passes only the stereo part of the matrix transparently, and blocks the multichannel part.
© 2017 Axia Audio - Rev 2.0
VMix and VMode • 55
Mode option “Pass multi-channel” passes all 8 matrix inputs to the corresponding outputs transparently.
“Transpose ...” options move two channels between the stereo part of the matrix and channels 1 and 2 of the
multi-channel part.
“Upmix ...” and “Downmix ...” options mix, copy, and/or apply gains to input channels, to obtain output channels.
Generally, upmixing is a process that creates a bigger channel count at output from a smaller channel count at input (for
example, multi-channel from stereo), and downmixing is the opposite (stereo from multi-channel).
“Split …” options create a mono sum from a stereo input, apply the specified attenuation, and send it to either
Left or Right output channel as indicated in the name of the option, while sending no signal to the other.
“Combine ...” options are special in that they take input from two input streams (designated A and B) and output
to one stream. The routing is indicated in the name of the option, Combine A, B, where the “side” identified with the
A field results as the Left side of the output and B field in the Right side . For example “Combine Right,Left” takes
the stereo Right side from the A stream input into the Left side of the output channel, and Stereo Left side from the B
stream input to the Right side of output.
Output:
The output of the matrix will result into a network source. You can define this source to be a Livewire source type
or one of the many AES67 options.
VMix and VMode • 56
Source name is used to identify the source in the network. Livewire sources are advertised and this name is used
in the information propagation. AES67 does not define a required method for how sources are advertised. Once this
is defined, this name may be utilized.
Address is used to define the Livewire channel number or an AES67 address.
Stream type defines the type of stream to be generated from the matrix output. Similar to the input type field, there
are many options available because of AES67 and the type used in Livewire are identified.
`
Packet time defines the latency and bandwidth consumption of the stream. Livewire has historically given this
field a user friendly name such as Live or Standard stereo. The time value is the amount of audio encapsulated within
the packet. The shorter time period permits the transmission of audio more frequently with the result of high bandwidth utilization and lower delay at receiver. The longer period results in better packet utilization and thus a lower
stream bandwidth with a slower reception at receiver. The ptime used by Livewire devices are identified.
© 2017 Axia Audio - Rev 2.0
Some VMode examples
Create a Mono Stream From One Side of a Stereo Channel
Sometimes, the program content that is fed on a satellite downlink is received on only the Left channel while
another content or information on the Right channel that you don’t want to air. Using VMode, you can split the Left
and Right sides and create a new source using just the channel you want. Here’s how:
• Select EXTERNAL
• Provide a useful name
• Define the source stream from the network you wish to input. The example assumes Livewire channel
19201.
• Define the type, the example assumes a Livewire stereo source.
• Define the stream address or Livewire channel number.
• Define the type, again we assume Livewire 2-channel.
• Define the ptime, in the example we define a Livewire Standard Stereo packet.
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VMix and VMode • 57
• “Upmix from Left” mode
Create a Split Record Feed from Multiple Sources
You want to create a two channel recording with independant content and gain control for each channel different
from Program 1 of the Fusion console. This single stream will be recorded on a stereo recorder. You can construct a
custom VMode stream to satisfy this requirement, like so:
The resulting Livewire 2-channel output would have the Left side of AUX A on the Left and Left side of AUX B
on the Right.
VMix and VMode • 58
Create a stereo Livewire stream from a 2-channel AES67 stream.
What’s Next
Telephones are an important part of many stations’ on-air programming. Since Axia Audio is a part of Telos,
Fusion features tight interoperation with Telos multi-line phone systems. In the next chapter, we’ll discuss how to set
up your console and phone system to work together seamlessly.
© 2017 Axia Audio - Rev 2.0
Chapter 5
Working With Phones
One of the advantages of a Livewire studio is the smooth integration of Telos telephone interface equipment with
the mixing console. Console telephone control modules let operators work phone without taking their eyes or hands
off the console; mix-minus is handled automatically, and each fader has its own mix-minus capability, so you will
never run out of busses.
Since advanced Telos telephone interfaces have Ethernet connections, they integrate easily with Axia networks,
exchanging control signaling between console and phone system without the need for the usual parallel connections.
This Chapter describes how to configure your console for use with Telos talkshow systems. But even if you don’t
own a Telos phone system, GPIO control of telephone equipment without a Livewire interface is also possible; refer
to the GPIO chapter for details.
For details on how to use the Fusion Call Controller module to control multi-line phone systems directly from
your console, please refer to the chapter entitled Operations, earlier in this manual.
Phone Setup Choices
There are three methods of setting up phone control with the Fusion console. These methods are referred to as:
• EU Phone (networked)
• US Phone (networked)
• No Phone Control (GPIO enabled control)
US Phone is the method most common in North America, where the operator has the ability to choose between
multiple incoming lines to feed the telephone hybrid. Commonly, two hybrids are presented on separate faders, and
the user dynamically switches between incoming lines; the Fusion Phone Controller module is used for this mode of
operation.
Most Telos multi-line call systems support both of these two methods.
No Phone Control is a way of controlling third-party hybrids which do not work with Telos control protocols;
instead, basic GPIO closures are used to “take” and “drop” lines.
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Working With Phones • 59
EU Phone is the method most used in European countries, wherein a single line is assigned to a single hybrid.
This is also known as “hybrid per fader” mode; that is, there is no switching between multiple lines on a single
hybrid. Each line is presented on its own dedicated console fader. If four phone lines are desired for use, each line’s
hybrid is presented on a separate fader. No on-console line controller module is used.
Setting Up for EU Phone Operation
First, create a Source Profile for your fixed hybrid (as described in the section in Chapter 2 of this manual
entitled “Working With Profiles”). Then:
1. Select the Phone Source type.
2. Select the appropriate Primary Source using the adjacent dropdown Source Selector box to pick the desired Livewire channel from your Telos phone system.
3. In the Phone Control portion of the Source Profile, there will be an EU Phone section as shown in the image below.
»» Enter the IP Address of the Telos phone system in the Server IP box.
»» Select the Telos phone system you’re interfacing with. Choose from Telos 2, Nx-Series (shown as
AP (Nx12))or VX and enter the appropriate settings for the device.
àà If you’re using a Telos TWO, enter the number of the line to be used in the Line box.
àà If using an Nx system, enter the Line and Hybrid numbers in their respective boxes. Select
the Use 2nd Show check box if split shows are enabled on your phone system.
Working With Phones • 60
àà If using a VX system, enter the Studio Name as configured in your VX Engine, and the
number of the Fixed Hybrid.
When finished, remember to save your Source Profile.
© 2017 Axia Audio - Rev 2.0
Setting Up for US Phone Operation
Using the US Phone method of operation requires the use of the Fusion Call Controller module. With this
method, you can use Fusion’s Show Profiles feature to instantly recall show setups that choose between different
phone systems, or even different phone system configurations.
For example, one Show Profile could call up the configuration needed for a Talk format, while another might
recall the configuration needed for music request lines, allowing a single studio to be the Control Room for any
station in a cluster.
Using this mode requires two steps: defining a Show Profile, then defining Source Profiles.
Show Profile Settings For US Phone Operation
First, create a new Show Profile using the instructions found in the “Show Profiles” section of Chapter 2 of this
manual.
Then, configure your new show profile. The Show Profile page of your Fusion’s Web interface displays the name
of the show, all the Channels (faders) of the surface, followed by several additional items, one of which is the Phone
link. If more than one Call Controller module is installed in your console, multiple instances of this configuration
option will be shown.
Working With Phones • 61
Select the Phone link to display the following screen:
© 2017 Axia Audio - Rev 2.0
The Phone Type is where you select the protocol used to communicate with the call server. Prior to the VX system,
the AP protocol was used. The AP option is to support legacy Telos products still in use, otherwise you would likely
select the VX option.
The Phone Server IP is where you identify the IP address of the Telos product that is managing the calls. In
cases where the non-default user name of password is to be used, you would add those in the text box. An example is
MyUser:[email protected] .
The Show Name field is used to identify a phone system configuration when using a Telos Series 2101 or VX
system. These names are defined in your phone system’s setup. Enter the configuration name you wish your Call
Controller to log into.
Note: with the VX system, this field can be left blank, permitting the changing of the show assigned with a
VSet phone, or the VX interface.
The Host / Studio Name field is used differently with different systems.
• With VX systems, use this field to define the “Studio” name. This information allows the Fusion Call Controller to interface with the appropriate configuration as defined in the VX. An incorrect name instructs the
Fusion to connect to a non-existing configuration, which is indicated by a circling icon on the top left of the
Call Controller.
• In legacy Telos phone systems, this field is used to log into the appropriate configuration:
»» “Hybrid1”, “Hybrid2”, “Hybrid1&2” (used in TWO-x-12)
»» “Hybrid 1&2”, “Hybrid 3&4”, “Hybrid 1-4” (used in Nx6 & Nx12)
The Show Password field is used to permit access between your Fusion console and any Telos Call system that
has password protection at show levels. If you have a password for the phone system Show you want to use, enter it
here.
Working With Phones • 62
The Reversed Hybrid option swaps the banks of the Call Controller. By default, the left column controls Hybrid
1, and the right column controls Hybrid 2; selecting this option reverses these.
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The Mode option defines the operation of the two Call Controller columns for line selection. The default is 12
lines, which provides control for Hybrid 1 and Hybrid 2. The 24 lines option is available for high call requirements
and, and when setting each button as a different line.
Once you’ve entered the information required for your phone system, save your changes and proceed to Source
Profile configuration.
Source Profile Settings for US Phone Operation – VX & Nx Systems
You’ll need to create a Source Profile for your fixed hybrid (as described in the section in Chapter 2 of this
manual entitled “Working With Profiles”). In legacy Telos Phone systems, you may have 1, 2, or even 4 hybrids;
with VX systems, the number may be many more.
To begin:
1. Select the Phone Source type.
2. Select the appropriate Primary Source using the adjacent dropdown Source Selector box to pick the desired Livewire channel from your Telos phone system.
3. In the Phone Control portion of the Source Profile, you’ll skip the EU Phone section and move to the US
Phone section. This will only appear if a Call Controller module is installed, as shown in the image below.
»» Enter the IP Address of the Telos phone system in the Server IP box.
»» Select the Telos phone system you’re interfacing with. Choose from Telos 2, Nx-Series (shown as
AP (Nx12))or VX and enter the appropriate settings for the device.
àà If you’re using a Telos TWO, enter the number of the line to be used in the Line box.
àà If using an Nx system, enter the Line and Hybrid numbers in their respective boxes. Select
the Use 2nd Show check box if split shows are enabled on your phone system.
»» Select the Call Controller 1 radio button (if you have multiple Call Controllers installed, select the
appropriate one).
»» In the Hybrid box, enter the hybrid number you want this Source Profile to be associated with.
»» The Fixed Line box is another method to do what the EU Phone method accomplishes; i.e., assigning a dedicated line to a dedicated fader. Most users with Call Controllers will not use this option.
»» The Mashing Allowed checkbox permits a single hybrid to handle multiple callers at once by allowing the board operator to select multiple line buttons. Check (or un-check) this box as you desire.
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Working With Phones • 63
àà If using a VX system, enter the Studio Name as configured in your VX Engine, and the
number of the Fixed Hybrid.
Working With Phones • 64
© 2017 Axia Audio - Rev 2.0
Source Profile Settings for US Phone Operation – Hx6 & iQ6 Systems
As with the other systems discussed here, you’ll begin by creating a Source Profile for each of your Hx6 or
iQ6 hybrids.
1. Select the Phone Source type.
3. In the Phone Control portion of the Source Profile, you’ll skip the EU Phone section and move to the US
Phone section.
»» Select the Call Controller 1 radio button, then and enter the hybrid number in the Hybrid box. The
iQ6 and Hx6 have two hybrids, so the option will be either 1 or 2.
»» Leave the Fixed Line entry at 0.
Repeat these steps to create a Source Profile for your phone system’s second hybrid.
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Working With Phones • 65
2. Select the appropriate Primary Source using the adjacent dropdown Source Selector box to pick the desired Livewire channel from your Telos phone system.
The Show Profile itself also needs configuration, to allow the console to log into the Telos Phone system as a
client. To do this, click on the desired Show Profile in your Fusion’s Web UI, and select the Phone link.
• Enter the server information in the Phone URL field, so:
vx:username:[email protected]
where xxx.xxx.xxx.xxx is the IP address of your phone system. (Hx6 and iQ6 systems use Telos VX control
protocol, so the vx prefix in this example is correct.)
A shortcut in Fusion assumes default values. If default values are used in your phone system, it would be
possible to enter
vx:xxx.xxx.xxx.xxx
in this field, without explicitly entering the username and password. Check your Telos product to verify
username and password. Default settings of the iQ6/Hx6 is a username of user with no password.
• In the Host/Studio Name box, enter the value of Hybrid 1&2.
Working With Phones • 66
Save your changes and load the Show Profile to your console. The Call Controller will show a dot in the first 6
line selections if the lines are present. If the first line selector is showing a spinning square, there is difficulty logging
into the Phone system. Check your settings and verify the Telos phone system is online.
Operating Options
When selecting a line directly using the Call Controller module, the default behavior
is for the Left column of buttons to assign the line to Hybrid 1, and the Right column to
assign the line to Hybrid 2. But when using phone systems with more than two hybrids, you
may assign lines to hybrids other than the default by using the appropriate Source Profile
options.
For instance, in the example illustrated in the screen capture above, the third Hybrid
in an Nx system is configured for use with a console that has a single call controller
installed. This Source Profile would be assigned to any of the standard fader strips on the
console (not to the fader associated with the Call Controller).
© 2017 Axia Audio - Rev 2.0
To select a line to use with this #3 hybrid, the board operator would press the A key on the fader strip. The words
Take Line will then appear on the channel display; the operator can then select any line from the Call Controller and
it will be assigned to this hybrid for use.
Setting up for GPIO control (“No Phone Control”)
The No Phone Control option is the default selection when you create a new Phone Source Profile. This type is
intended for single line hybrids like the Telos Hx1 and Hx2, but can also be used with older single-line Telos phone
systems as well as those made by other phone system vendors.
Please refer to the GPIO Telephone Hybrid Logic chart found in Appendix B of this manual for the appropriate
pinouts.
When configuring a Source Profile for this operational mode, the salient field is the Hybrid Answer Mode
dropdown box.
The default setting for Hybrid Answer mode is Normal, Auto Answer Diasbled. If your desire is to achieve
pulses on pins 4 and 5 of the GPIO logic port associated with your phone hybrid, then you must choose one of the
two other options. These other options will create pulses when the channel changes ON state or Preview state.
Wiring from a GPIO port that has been configured with this channel to the hybrid is the next step. Again, refer to
Appendix B for the appropriate pinout charts, but here’s a quick reference chart:
Axia GPIO port DA-15 P
Pin-1
Pin-7
Pin-2
Pin-4
Pin-3
Pin-5
Working With Phones • 67
Telos Hx1 DE-9 Pin
© 2017 Axia Audio - Rev 2.0
Additional Phone Type Source Profile Options
There are a few more phone-specific options that are available in Phone source profiles. Here’s an overview:
• Hybrid Answer Mode. Your choices are:
»» Normal, Auto Answer Disabled. This is the default; turning on the fader that the hybrid is assigned
to does not pick up the selected line.
»» Channel ON Answers Hybrid. Turning the fader on immediately picks up the selected line.
»» Channel ON or Preview ON Answers Hybrid. As above. Additionally, placing the fader channel in
Preview answers the call; the board operator can then talk to the caller through the Operator Mic.
• Feed To Source: This setting controls the mix-minus that’s automatically generated and fed back to the
phone caller.
»» Disabled means no mix-minus will be generated for the caller, so they will not hear you talking to
them!
»» Default (the option most often chosen) enables mix-minus for the caller, and enables a couple of
additional options:
Working With Phones • 68
àà Dim Gain lets you trim the volume of the generated backfeed.
àà Feed Source defaults to Auto, which automatically generates a backfeed constructed of
PGM-1 minus caller audio when the fader is On, and feeds the offline Phone mix to the
caller when the fader is Off. This is the most commonly used mode, but you may also specify
any of the other Program or Aux Send buses.
»» Custom enables a power-user’s toolkit of mix-minus options to build custom backfeeds with selectable GPIO Monitor and Program Bus assignments based on channel On/Off/Preview state. (We’ll
cover details of these settings in the Advanced Configuration section of this manual.)
When you are done entering configurations, be sure to save your Show and Source profiles, and then load your
Show Profile and newly created Phone sources to your console.
© 2017 Axia Audio - Rev 2.0
Appendix A
Advanced Configuration Reference
This appendix lists the advanced options available when setting up Show Profiles, and the “magic key”
combinations available from the console surface.
Part 1 – Magic Key Sequences
The key sequences below will allow you access information about your Element using the main display screen.
These keys are located on your Monitor Module.
• IP Address Book — * + 4 + 7 (hold for 5 seconds): View and change console IP, Gateway and NTP Server
addresses, and set surface Web page password.
• Timer Options key (hold for 5 seconds): A momentary press invokes board-op timer start/stop functions,
but when held for 5 seconds, Engineer can reset time of day, enable/disable NTP sync, set time zone, and
choose 12-hour or 24-hour clock format.
• Capture Mode — * + 2 (hold for 5 seconds): Invokes “Capture” mode. Use this when adding or removing
modules to capture new console configuration.
• Test Mode — * + # + 2 (hold for 5 seconds): Invokes “Test” mode to test functions of installed modules.
• Module Information — # + * (hold for 5 seconds): Displays information about installed modules.
• Show Profile Reset: Press and hold the Show Profile key for 5 seconds to discard any changes made to the
last-loaded Show Profile. This resets the console to the baseline values of the last-loaded Show Profile.
With Expert Monitor Module only, pressing and holding ? for 5 seconds displays a list of the “magic key” menus
available.
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Advanced Configuration Reference • 69
• VOCO8 Control — * + 4 + 8 (hold for 5 seconds): Displays the VOCO8/Sound 4 control interface.
Part 2 – Show Profiles Options
Show Profiles are snapshots that allow you to set up a console the way your talent wants it, then save and recall
those settings with one button press. Up to 99 Show Profiles can be saved per console.
Advanced Configuration Reference • 70
The easiest way to create a Show Profile is to set up your console the way you like it using the controls on the
board itself – then use the Capture Show Profile link found on the Shows screen of your Fusion control center
web page. This will take a snapshot of the entire console – fader assignments, source EQ, mic dynmics, Monitor
assignments, even headphone EQ settings – and save them to a new Show Profile that you can use as-is, or edit later.
There are also some options, such as Record Mode setup, that are not able to be set from the Fusion surface itself,
and must be set up by editing the Show Profile itself. In the next few sections, we’ll give you a line-item reference
listing of the options found on each Show Profile screen.
Channel Screen Options
There’s a Channel Screen for each of your installed faders. Following is a listing of options presented on each screen.
NOTE: choosing The “Retain Source Setting” option for any item leaves that item unchanged when loading
the Show Profile.
General Controls Section
Source ID
• Dropdown box selects fader’s assigned source from saved Source Profiles.
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Feed To Source Mode
If the loaded source has a mix-minus, chooses the source for mix-minus creation.
• Auto lets the console choose a mix-minus source automatically.
• Phone, PROGRAM and AUX SEND options let you manually choose the bus to use as the basis of the
mix-minus.
Auto-Start Timer
• Enable: Console interval timer starts when fader ON key is pressed.
• Disable: Timer is unaffected by fader ON key.
Signal Mode
• Stereo: sets source signal mode to Stereo.
• Left / Right: takes the chosen side of the input signal and sends it to both sides of the stereo channel.
• Sum: Creates a sum of both input channels and sends the sum to both sides of the stereo channel.
Signal Mode Locked
• Unlocked: allows board op to change the Signal Mode from the surface.
• Locked: prevents board op from making changes.
• Select the Use: radio button to specify a fader trim cut or boost between -25 and +25 dB.
Fader Trim Lock
• Unlocked: allows board op to trim the fader using Channel controls on the console.
• Locked: prevents board op from making changes.
Panorama Position
• Select the Use: radio button to Pan the input signal left or right of center.
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Advanced Configuration Reference • 71
Fader Trim Gain
Phase
• Normal: No phase change to input signal.
• Invert Left: Reverses the phase of the left input channel only.
• Invert Right: Reverses the phase of the right input channel only.
• Invert Left and Right: Reverses the phase of both input channels.
EQ Active
• Bypass: Loads the input source with no EQ adjustment.
• Active: Loads the source with EQ active, and applies EQ settings specified in following sections.
EQ High Mode
• Shelf: Selects high-shelf style of EQ application.
• Peak: EQ is applied to a selected frequency, “notch filter” style.
EQ High/Mid/Low Frequency
• Select the Use: radio button to set the center frequency of the selected band.
EQ High/Mid/Low Gain
Advanced Configuration Reference • 72
• Select the Use: radio button to specify an EQ cut or boost between -25 and +15 dB to the selected frequency.
Assign To…
• On: Assigns fader strip to the specified Program or Aux Send bus upon Show Profile load.
• Off: Removes fader strip from the specified Program or Aux Send bus upon Show Profile load.
NOTE: Channels may be simultaneously assigned to any combination of Program 1 – 4 and Aux Send 1 – 4
mixing buses.
Aux Send A-B-C-D Pre/Post Fader
• Pre-Fader: Sends the input source to the selected Aux Send bus before fader gain adjustment.
• Post-Fader: Sends the input source to the selected Aux Send bus after fader gain adjustment.
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Aux Send A-B-C-D Pre/Post ON
• Pre-ON: Sends the input source to the selected Aux Send bus before the channel’s ON/OFF switch.
• Post-ON: Sends the input source to the selected Aux Send bus after the channel’s ON/OFF switch.
Aux Send A-B-C-D Gain
• Select the Use: radio button to specify any additional gain or cut to be applied to the assigned source before
sending to the selected Aux Send bus.
Noise Gate Status
• Bypass: Turns off the noise gate feeding the Vocal Compressor section.
• Active: Turns on the Compressor noise gate.
De-Esser Status
• Bypass: Turns off the Compressor De-Essing function.
• Active: Turns on De-Essing function.
Noise Gate Threshold
• Select the Use: radio button to specify the level, up to -50 dB, at which the Noise Gate activates to attenuate input signals that fall below that level.
• Select the Use: radio button to specify the amount, up to -30 dB, of attenuation to apply to the input signal
when the specified Threshold is met.
Compressor Threshold
• Select the Use: radio button to specify the ceiling, up to -30 dB, at which the Compressor activates to
attenuate input signals above that level.
Compressor Ratio
• Select the Use: radio button to specify the aggressiveness of the compressor, up to 16:1.
De-Esser Threshold
• Select the Use: radio button to specify the ceiling, up to -20 dB, at which the De-Esser activates to
attenuate sibilance.
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Advanced Configuration Reference • 73
Noise Gate Depth
De-Esser Ratio
• Select the Use: radio button to specify the aggressiveness of the De-Esser, up to 8:1.
Compressor Mode
• No Freeze: The compressor is “free” to operate no matter the input level.
• Freeze: Prevents the compressor from “sucking up” room noise during brief pauses in audio input.
Backward Feed Dim Gain
• Select the Use: radio button to specify the amount of attenuation, up to -30 dB, will be applied to program
audio being fed back to this source (if it has an associated Backfeed, such as an IFB or Talkback channel).
Channel On/Off Status
• Safe ON: If the fader is ON, and the Show Profile specifies a new source be loaded to this fader, the new
source will be queued until the fader is turned off.
If the new source is the same as the old source, the fader is immediately turned ON when the Show Profile
loads.
• Force OFF: Immediately turns the fader OFF when the Show Profile loads.
• Force ON: Immediately turns the fader ON when the Show Profile loads. If a new source is to be loaded to
the fader, it is changed immediately, regardless of whether audio from a previous source is passing through
the fader.
Advanced Configuration Reference • 74
Control Lock Map
• PGM1 – PGM4, Options, Preview, On/Off, Fader, Talkback, HP Source: Place a check mark in any of
these boxes to keep the board op from changing the Show Profile’s pre-selected options for this fader.
Group Start
Fusion contains a Group Start feature that enables the user to turn several faders ON by pressing the ON key of
the Master fader; useful in roundtable discussions or multi-talent bullpens.
This function can be controlled in this screen, or by setting the option in each individual Channel Options screen.
• Master: Designates this fader channel as a Group Start Master. Pressing its ON or OFF keys will turn
Slave faders on and off as well.
• Slave: Designates this fader channel as a Slave. It will mirror the ON/OFF state of the Group Start Master fader.
• Independent: Normal ON/OFF operation.
© 2017 Axia Audio - Rev 2.0
Fader Position
• Select the Use: radio button to specify the level at which the fader should be set when the Show Profile is
loaded. You may specify any level between -73 dB and +10 dB.
This setting operates in conjunction with the Channel On/Off Status described above. Using these two
settings, you can set a channel to turn ON, and its fader to assume a preset output value, when the Show
Profile is loaded. This is useful when creating a Show Profile for use with automation systems; combine
it with the Control Lock Map controls to set up a Show Profile for unattended, automated operation with
controls that cannot be inadvertently changed by careless operators.
Individual Headphones Section
Any Fusion input defined as a Microphone source can have a dedicated headphone feed, to facilitate individual
Talkback (IFB) to and from the board operator or other talent. The options below affect this dedicated headphone
feed, if enabled.
Current Source
• If a Microphone source is loaded to this channel, choose the audio source to be fed to its Individual Headphone Feed (Backfeed), if desired. Choices include all Program and Aux Send buses, Monitor bus, or a
direct feed from any individual source.
Source For Preset 1 / Preset 2
• Select the audio source assigned to the Preset keys on Axia Talent Headphone accessory panels. Choices
include all Program and Aux Send buses, Monitor bus, or a direct feed from any individual source.
• Select the Use: radio button to specify the level at which Individual Headphone channel should be fed. You
may specify any level between -85 dB and 0 dB.
Talkback Start Volume Low Limit
• Select the Use: radio button to specify the level at which Talkback will be fed to the Individual Headphone
channel. You may specify any level between -85 dB and 0 dB.
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Advanced Configuration Reference • 75
Headphones Master Gain
Record Mode Section
Fusion’s Record Mode is a “macro” that allows complex pre-defined operations to take place with a single button
press. The options below define bus assignments for the channel that occur when Record Mode is entered and exited.
Program 1 – 4
• Assign, While In Record Mode: Assigns this channel to the selected Program bus when Record Mode is
active.
• Remove, While In Record Mode: Removes this channel from the selected Program bus until Record
Mode is exited.
• No Change, While In Record Mode: This channel’s bus assignments do not change when Record Mode is
active.
ON/OFF
• Disable, While In Record Mode: Prevents the operator from changing the channel ON/OFF state when
Record Mode is active.
• No Change, While In Record Mode: Channel ON and OFF keys function normally during Record Mode.
When done editing the Channel options, be sure to click the Save Changes button. You’ll be returned to the
Show Profile options screen.
Auxiliary Send & Return Screen Options
This screen allows you to pre-select settings pertaining to the 4 Aux Send and 2 Aux Return mixing buses.
Advanced Configuration Reference • 76
Aux Send A-B-C-D Master Gain
• Select the Use: radio button to specify the master gain level at which the selected Aux Send bus will be set.
This is the gain applied to the Aux Send bus after any sources have been assigned to it. You may specify
any level between -60 dB and +10 dB.
Aux Send A-B-C-D On/Off Status
• Off: Turns off selected Aux Send bus upon Show Profile load.
• On: Turns on selected Aux Send bus upon Show Profile load.
Aux Return A-B Gain
• Select the Use: radio button to specify the gain level at which the selected Aux Return bus will be set. You
may specify any level between -60 dB and +10 dB.
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Aux Return A-B On/Off Status
• Off: Turns off selected Aux Return bus upon Show Profile load.
• On: Turns on selected Aux Return bus upon Show Profile load.
Aux Return A-B Signal Mode
• Stereo: sets bus signal mode to Stereo.
• Left / Right: takes the chosen side of the bus’ stereo signal and sends it to both sides of the stereo bus
output channel.
• Sum: Creates a sum of both stereo channels and sends the sum to both sides of the stereo bus output channel.
Aux Return A-B Panorama Position
• Select the Use: radio button to Pan the output signal of the bus to left or right of center.
Aux Return A-B Assign To PGM1-2-3-4
• On: Assigns the output of the specified Aux Return bus to the selected Program bus(es) upon Show Profile
load.
• Off: Removes the output of the specified Aux Return bus to the selected Program bus(es) upon Show Profile load.
• Dropdown box selects Aux Return’s assigned source from saved Source Profiles. Note that Program and
other Aux buses are not available in this dropdown; choices are limited to external devices only.
When done editing the Aux Send & Return options, be sure to click the Save Changes button. You’ll be returned
to the Show Profile options screen.
© 2017 Axia Audio - Rev 2.0
Advanced Configuration Reference • 77
Aux Return A-B Source ID
Monitor Section Screen Options
Options in this section permit you to define Control Room and Studio Monitors & Headphone selections, and
determine how on-screen Meters and Timers will behave when this Show Profile is loaded.
General Monitor Settings Section
Timer Mode (Onscreen “Count-Up” Timer)
• Manual: Gives board operator manual control of the event timer using the Timer keys found on the console’s Monitor Module.
NOTE: This setting interoperates with the Auto-Start Timer option. The Auto-Start Timer option can be set
in two ways:
• Check the Auto-Start Timer box in a Source Profile to enable the function whenever a source is loaded to
any fader.
• Use the Auto-Start Timer selection found in the Show Profiles Channel Options screen to override the
Source Profile setting for a particular Show.
• Auto-Reset: Timer resets to zero and begins counting when a fader is turned ON.
• Auto-Add: Timer begins counting when a fader is turned ON and stops counting when that fader is turned
OFF. In this mode, the timer will not reset to zero when it is restarted.
Show Tenths On
• No Timers: Tenths-of-a-second are hidden on both Count Up and Countdown timers.
Advanced Configuration Reference • 78
• Down Timer: Tenths are only shown on the Countdown timer.
• Up Timer: Tenths are only shown on the Count Up timer.
• Both Timers: Tenths are shown on both timers.
Preview Interlock Mode
• Disabled: Board op can “button mash” multiple Preview buttons to preview multiple sources.
• Enabled: Button-mashing is disabled; selecting a Preview button removes any other source from the Preview bus.
Switched Meter Source Select
• Program-4: Selects Program 4 mixing bus to be metered on the #4 on-screen meter display.
• CR Monitor/Preview: Switches the #4 meter to the Control Room Monitor channel. When a source is
placed in Preview, meter switches to display loudness of the source in Preview.
© 2017 Axia Audio - Rev 2.0
Program 3 Meter Input
• Dropdown box allows you to select from Program 3, Record bus, Phone bus or the sources assigned to
the Monitor Module’s External 1 and External 2 choices to be displayed on the #3 meter.
Preview Speaker Master Gain
• Select the Use: radio button to specify the level at which the Preview bus will be set when the Show Profile
is loaded. You may specify any level between -85 dB and 0 dB.
Preview Speaker Muted State Gain
• Select the Use: radio button to specify the level at which the Preview bus will be heard when it is muted.
You may specify any level between -85 dB and 0 dB.
NOTE: Normally, turning a Control Room microphone channel ON mutes the Preview speakers entirely. This
setting allows you to let the Preview bus be heard in the Control Room at a reduced level, even while CR
Mics are active.
Source ID For External Preview
• Dropdown box allows any Source not assigned to a fader to be heard in the Preview channel. Note that this
must be enabled by a pin on the CR Monitor GPIO.
Example: you have an intercom system you wish to feed into the console’s Preview channel. To do this, use
the Dropdown box to select the intercom’s audio source, then take the GPO from the intercom and use it to
gate open the External Preview input, which would be fed by the intercom audio.
• Select the Use: radio button to trim the level of the Talkback channel. You may specify any level of cut or
boost between -30 dB and +10 dB.
Feed To Source Sum Gain
• Select the Use: radio button to specify a gain compensation for mono-sum of Talkback audio sources. You
may specify between -6 dB and -3 dB of attenuation.
User Buttons GPIO Channel
• If you have a Fusion Expert Monitor Module and wish to control an external device using its 4 User keys,
enter the GPIO Channel Number of the device here.
GPIO Channel For Up Timer Control
• Allows an external device to trigger the Up Timer. Enter the GPIO Channel Number for the desired device.
© 2017 Axia Audio - Rev 2.0
Advanced Configuration Reference • 79
Talkback Gain
GPIO Channel For Down Timer Control
• Allows an external device to trigger the Down Timer. Enter the GPIO Channel Number for the desired
device.
Additional Meters Section
Extra Meter 1-2-3-4 Input
• This section allows to you specify meter sources for the “extra” meters that can be displayed in the center
section of Fusion’s on-screen display when using an Expert Monitor Module. The selections for each of the
meters are extensive and include all Program, Aux, External and Monitor busses, the Phone and Record
buses, all Fader Channel sources and backfeeds, all VMix inputs, Direct, Sub and Main outputs, as
well as all VMode inputs and outputs.
• The extra meters are displayed when the board operator presses the Meter Options key and presses the #6
Function knob to select the More Meters option.
Sources For External 1 & 2 Section
Source ID For External Input 1 – 2
• Use the Dropdown boxes to select sources to be loaded to the External 1 and External 2 Monitor Selection
keys when the Show Profile is loaded.
• Board operators can override these selections by pressing and holding the External keys for 5 seconds, then
choosing from the onscreen list of sources.
Advanced Configuration Reference • 80
Control Room Monitor Options Section
Source
• Use the radio buttons to choose the source that will be selected to feed the Control Room Monitor speakers
upon Show Profile load.
CR Monitor Master Gain
• Select the Use: radio button to specify the level of the CR Monitor speakers upon Show Profile load. You
may specify a value between -85 dB and 0 dB.
© 2017 Axia Audio - Rev 2.0
Signal Mode: CR Monitor
• Stereo: sets Monitor channel to Stereo.
• Left / Right: takes the chosen side of the Monitor channel and sends it to both speakers.
• Sum: Creates a sum of both sides of the Monitor channel and sends the sum to both speakers.
CR Monitor Dim Gain
• Select the Use: radio button to specify the amount of attenuation, up to -30 dB, to be applied to the CR
Monitor channel when Talkback or Preview are in use.
CR Monitor Muted State Gain
• Select the Use: radio button to specify the amount of attenuation, up to -85 dB, to be applied to the CR
Monitor channel when muted.
NOTE: Normally, turning a Control Room microphone channel ON mutes the CR Monitor speakers entirely.
This setting allows you to let the monitors to be heard in the Control Room at a reduced level, even while
CR Mics are active.
GPIO Channel For CR Monitor
• Enter the GPIO channel number used to trigger On Air lamps or other GPIO functions. This option should
typically be programmed with the same logic channel number for all Show Profiles.
Control Room Headphones Options Section
• Use the radio buttons to choose the source that will be selected to feed the Control Room headphones upon
Show Profile load.
CR Headphones Master Gain
• Select the Use: radio button to specify the level of the CR Headphones upon Show Profile load. You may
specify a value between -85 dB and 0 dB.
Signal Mode: CR Headphones
• Stereo: sets Headphone channel to Stereo.
• Left / Right: takes the chosen side of the Headphone channel and sends it to both speakers.
• Sum: Creates a sum of both sides of the Headphone channel and sends the sum to both speakers.
© 2017 Axia Audio - Rev 2.0
Advanced Configuration Reference • 81
Source
CR Headphones Independent
• Follow Monitors: CR Headphone source mirrors CR Monitor source selection.
• Use Headphones Source Select: CR Headphones and CR Monitors are selected independently.
Preview-In-Headphones Mode
• Off: Sources assigned to Preview bus are not heard in CR Headphones.
• Stereo: Sources assigned to Preview bus are heard in both sides of the CR Headphones.
• Split: Sources assigned to Preview bus are summed to mono and heard only in right side of the CR Headphones. Regular audio is heard in left side.
CR Headphones EQ Active
• Bypass: No headphone EQ.
• Active: Headphones are EQd using the following settings.
CR Headphones EQ High Mode
• Shelf: Selects high-shelf style of EQ application.
• Peak: EQ is applied to a selected frequency, “notch filter” style.
Advanced Configuration Reference • 82
CR Headphones EQ High/Mid/Low Frequency
• Select the Use: radio button to set the center frequency of the selected band.
CR Headphones EQ High/Mid/Low Gain
• Select the Use: radio button to specify an EQ cut or boost between -25 and +15 dB to the selected frequency.
Studio Monitor Options Section
Source
• Use the radio buttons to choose the source that will be selected to feed the Studio Monitor speakers upon
Show Profile load.
Source ID For External Input
• Dropdown box chooses what source will be auditioned when External is chosen from the Studio monitor
assignment list.
© 2017 Axia Audio - Rev 2.0
Studio Monitor Master Gain
• Select the Use: radio button to specify the level of the Studio Monitor speakers upon Show Profile load.
You may specify a value between -85 dB and 0 dB.
Studio Monitor Dim Gain
• Select the Use: radio button to specify the amount of attenuation, up to -30 dB, to be applied to the Studio
Monitor channel when Talkback or Preview are in use.
Studio Monitor Muted State Gain
• Select the Use: radio button to specify the amount of attenuation, up to -85 dB, to be applied to the Studio
Monitor channel when muted.
NOTE: Normally, turning a Studio microphone channel ON mutes the Studio Monitor speakers entirely. This
setting allows you to let the monitors to be heard in the Studio at a reduced level, even while Studio Mics
are active.
GPIO Channel For Studio Monitor
• Enter the GPIO channel number used to trigger On Air lamps or other GPIO functions.
This option should typically be programmed with the same logic channel number for all
Show Profiles.
Master Module Control Lock Map Section
Place a check mark in any of these boxes to keep the board op from changing the Show Profile’s pre-selected options.
When done editing the Monitor Section options, be sure to click the Save Changes button. You’ll be returned to
the Show Profile options screen.
Record Mode Screen Options
Record Mode is like a “macro” that helps talent quickly prepare to record phone bits, interviews or other
program segments for later airing. Any source assigned to the Program-4 bus automatically feeds the Record and
Phone buses as well.
Sources assigned to Program-4/Record will follow the Record options in their Source Profiles; the options here
primarily affect the behavior of Monitors when Record Mode is invoked.
© 2017 Axia Audio - Rev 2.0
Advanced Configuration Reference • 83
• This functions like a “master lock” for console functions – a superset of the Control Lock Map found in
the Show Profiles Channel Options pages. You can essentially prevent any setting on the board from being
changed if you so desire.
Record Mode Configuration Section
Record Mode Activation
• Disabled: Disables Record Mode entirely for this Show Profile.
• Enabled: Allows activation of basic Record Mode for this Show Profile. CR Monitor and CR Headphone
assignments automatically switch to the Program 4 bus, and the bus assignment keys for channels assigned
to Program 4 flash.
• Flexible: Allows activation of Flexible Record Mode, with custom Monitor, Headphone and Meter options
set in the “Flexible Record Mode Options” section that follows.
GPIO Channel For Recorder Control
• Enter the GPIO channel number used to trigger your dedicated recording device. This option should
typically be programmed with the same logic channel number for all Show Profiles.
Flexible Record Mode Options Section
When “Flexible” is chosen as the Record Mode Activation option, the following options are active.
NOTE: The options below allow you to customize the Monitor, Headphone and Meter settings that are automatically selected when you ENTER Record Mode.
CR Monitor Source
Advanced Configuration Reference • 84
• Retain: Does not change the Monitor feed; keeps the Monitor selection the board operator was using prior
to entering Record Mode.
• Program 1 – 4, Record, Phone, Auxiliary A – D, External 1 – 2: Changes the Monitor feed to the
selected bus or Monitor channel when Record Mode is engaged.
• Recall: Re-loads the Monitor selection that was manually selected by the board operator the last time
Record Mode was active.
Studio Monitor Source
• Same options as described in “CR Monitor Source” above, but for the Studio Monitor feed.
CR Headphone Source
• Same options as described in “CR Monitor Source” above, but for the Control Room Headphone feed.
© 2017 Axia Audio - Rev 2.0
4th Meter Source
• Retain: Does not change the #4 Meter on the Fusion display; keeps the Meter selection the board operator
was using prior to entering Record Mode.
• Program 4, Record, Phone, Auxiliary A – D, External 1 – 3, Monitor: Changes the #4 Meter to the
selected bus, external source, or the audio of the CR Monitor when Record Mode is engaged.
• Recall: Re-loads the #4 Meter selection that was manually selected by the board operator the last time
Record Mode was active.
When done editing the Record Mode options, be sure to click the Save Changes button. You’ll be returned to the
Show Profile options screen.
Group Start Screen Options
Group Start
Fusion contains a Group Start feature that enables the user to turn several faders ON by pressing the ON key of
the Master fader; useful in roundtable discussions or multi-talent bullpens.
This function can be controlled in this screen, or by setting the option in each individual Channel Options screen.
• Master: Designates this fader channel as a Group Start Master. Pressing its ON or OFF keys will turn
Slave faders on and off as well.
• Slave: Designates this fader channel as a Slave. It will mirror the ON/OFF state of the Group Start Master
fader.
When done editing the Group Start options, be sure to click the Save Changes button. You’ll be returned to the
Show Profile options screen.
Phone Screen Options
The Phone screen is used when setting up a Telos talkshow system for use with your console. Please refer to
Chapter 5, “Working With Phone Hybrids”, for details. If no Call Controller module is installed, this screen will be
empty.
© 2017 Axia Audio - Rev 2.0
Advanced Configuration Reference • 85
• Independent: Normal ON/OFF operation.
Appendix B:
Configuring GPIO
The Axia IP-Audio system is capable of transporting routable machine logic along with each audio signal.
Unlike conventional logic connections which require each command circuit to be wired individually, Axia sends
machine controls over the same Ethernet your upon which your audio travels.
Fusion’s GPIO capabilities provide control of external audio devices, logic commands for routine studio/control
room operations such as tally lights, monitor muting, On-Air lights and more, and even “virtual” GPIO for routing
system commands, using Axia Pathfinder routing controls tools.
This Appendix provides a fast overview of these GPIO functions. Please refer to the Axia xNode User’s Manual
for more in-depth information on configuring GPIO.
GPIO Port Definitions
Axia PowerStation mix engines feature 4 built-in GPIO connections; Fusion consoles using StudioEngine mix
engines must be paired with xNode GPIO Nodes. Each
GPIO port can be associated with a device in your studio, and provides five opto- isolated inputs and five optoisolated outputs per device.
GPIO ports are pre-programmed to support several different types of devices; when you construct Source
Profiles, the GPIO type best suited for the Source Type you choose is associated with that Profile. When the source
is assigned to a console fader, this Source Profile selection tells the GPIO port what sort of command to send to the
attached device.
If the Source Profile defines the attached device as a microphone, the GPIO port sends logic for On, Off, Remote
Mute and Remote Talk commands on the appropriate pins. If the Source Profile is configured for a line input, the
GPIO port sends Start, Stop and Reset commands, plus closures for Ready lights, etc.
1.
2.
3.
4.
5.
6.
7.
8.
Microphone (Operator, Guest or Producer)
Line Input
Codec
Telephone Hybrid
Computer Playback Device
Control Room Monitor
Studio Monitor
Accessory Button Panel Device
The next few pages contain tables that explain what function the GPIO port pins provide in each different device mode.
© 2017 Axia Audio - Rev 2.0
Configuring GPIO • 87
Axia GPIO ports can deliver unique command sets for the following types of devices:
GPIO Operator’s Microphone Logic
Name
Pin
Type
Notes
ON Command
11
Active Low Input
Turns channel ON
OFF Command
12
Active Low Input
Turn channel OFF
TALK (to Monitor 2) Command
13
Active Low Input
Activates the TALK TO MON2
function and routes mic audio
to the Talkback bus.
MUTE Command
14
Active Low Input
Mutes channel outputs
TALK (to PREVIEWED
SOURCE) Command
15
Active Low Input
Activates the TALK button
on every source currently in
preview and routes mic audio
to the Talkback bus.
ON Lamp
1
Open Collector to Logic
Common Return
Illuminates when channel is
ON unless TALK or MUTE is
active
OFF Lamp
2
Open Collector to Logic
Common Return
Illuminates when channel is
OFF
TALK (to Monitor 2) Lamp
3
Open Collector to Logic
Common Return
Illuminates when TALK TO
MON2 is active
MUTE Lamp
4
Open Collector to Logic
Common Return
Illuminates when MUTE is
active
TALK (to PREVIEWED
SOURCE) Lamp
5
Open Collector to Logic
Common Return
Illuminates when TALK to
PREVIEWED SOURCE is active.
Source Common
7
Logic Common
Connect to ground of source
device or to Pin 8
Logic Common
8
Internal 5 Volt return
Can be connected to Pin 7 if
source is not providing common
Logic +5 Volt Supply
9
Logic Supply, Individually
Fused
Can be connected to Pin 10 if
source is not providing voltage; active only when source
has been assigned to channel.
Input Common
10
Common for all 5 inputs
Connect to power supply of
source device or to Pin 9
NOT CONNEC TED
6
INPUTS
OUTPUTS
Configuring GPIO • 88
POWER & COMMON
© 2017 Axia Audio - Rev 2.0
GPIO Control Room Guest Microphone Logic
Name
Pin
Type
Notes
ON Command
11
Active Low Input
Turns channel ON
OFF Command
12
Active Low Input
Turn channel OFF
TALK (to CR) Command
13
Active Low Input
Mutes channel outputs and
routes source audio to PVW
speakers
MUTE Command
14
Active Low Input
Mutes channel outputs
NOT CONNECTED
15
INPUTS
OUTPUTS
ON Lamp
1
Open Collector to Logic
Common Return
Illuminates when channel is
ON unless TALK or MUTE is
active
OFF Lamp
2
Open Collector to Logic
Common Return
Illuminates when channel is
OFF
TALK (to CR) Lamp
3
Open Collector to Logic
Common Return
Illuminates when TALK is
active
MUTE Lamp
4
Open Collector to Logic
Common Return
Illuminates when MUTE is
active
NOT CONNECTED
5
POWER & COMMON
7
Logic Common
Connect to ground of source
device or to Pin 8
Logic Common
8
Internal 5 Volt return
Can be connected to Pin 7 if
source is not providing common
Logic + 5 Volt supply
9
Logic Supply, Individually
Fused
Can be connected to Pin 10 if
source is not providing voltage; active only when source
has been assigned to channel.
Source Supply
10
Common for all 5 inputs
Connect to power supply of
source device or to Pin 9
NOT CONNEC TED
6
Configuring GPIO • 89
Source Common
© 2017 Axia Audio - Rev 2.0
GPIO Producer’s Microphone Logic
Name
Pin
Type
Notes
ON Command
11
Active Low Input
Turns channel ON
OFF Command
12
Active Low Input
Turn channel OFF
TALK (to MONITOR 2) Command
13
Active Low Input
Activates the TALK to MON2
function and routes mic audio
to the Talkback bus.
MUTE Command
14
Active Low Input
Mutes channel outputs
TALK (to PREVIEWED
SOURCE) Command
15
Active Low Input
Activates the TALK button
on every source currently in
Preview and routes mic audio
to the Talkback bus.
ON Lamp
1
Open Collector to Logic
Common Return
Illuminates when channel is
ON unless TALK or MUTE is
active
OFF Lamp
2
Open Collector to Logic
Common Return
Illuminates when channel is
OFF
TALK (to MONITOR 2) Lamp
3
Open Collector to Logic
Common Return
Illuminates when TALK to
MON2 is active.
MUTE Lamp
4
Open Collector to Logic
Common Return
Illuminates when MUTE is
active
TALK (to PREVIEWED
SOURCE) Lamp
5
Open Collector to Logic
Common Return
Illuminates when TALK to
PREVIEWED SOURCE is active.
Source Common
7
Logic Common
Connect to ground of source
device or to Pin 8
Logic Common
8
Internal 5 Volt return
Can be connected to Pin 7 if
source is not providing common
Logic + 5 Volt supply
9
Logic Supply, Individually
Fused
Can be connected to Pin 10 if
source is not providing voltage; active only when source
has been assigned to channel.
Source Supply
10
Common for all 5 inputs
Connect to power supply of
source device or to Pin 9
NOT CONNECTED
6
INPUTS
OUTPUTS
Configuring GPIO • 90
POWER & COMMON
© 2017 Axia Audio - Rev 2.0
GPIO Line Input Logic
Name
Pin
Type
Notes
ON Command
11
Active Low Input
Turns channel ON
OFF Command
12
Active Low Input
Turns channel OFF & sends
100 msec STOP pulse
PREVIEW Command
13
Active Low Input
Turns preview ON
RESET Command
14
Active Low Input
Turns channel OFF, while not
sending a STOP pulse
READY Command
15
Active Low Input
Illuminates OFF lamp to indicate source’s readiness
ON Lamp
1
Open Collector to Logic
Common Return
Illuminates when channel is
ON
OFF Lamp
2
Open Collector to Logic
Common Return
Illuminates when channel is
OFF and READY is active
PREVIEW Lamp
3
Open Collector to Logic
Common Return
Illuminates when PREVIEW
is ON
START Pulse
4
Open Collector to Logic
Common Return
A 100 msec pulse when the
channel status changes from
OFF to ON
STOP Pulse
5
Open Collector to Logic
Common Return
A 100 msec pulse when the
channel status changes from
ON to OFF
Source Common
7
Logic Common
Connect to ground of source
device or to Pin 8
Logic Common
8
Internal 5 Volt return
Can be connected to Pin 7 if
source is not providing common
Logic + 5 Volt supply
9
Logic Supply, Individually
Fused
Can be connected to Pin 10 if
source is not providing voltage; active only when source
has been assigned to channel.
Source Supply
10
Common for all 5 inputs
Connect to power supply of
source device or to Pin 9
NOT CONNECTED
6
INPUTS
OUTPUTS
Configuring GPIO • 91
POWER & COMMON
© 2017 Axia Audio - Rev 2.0
GPIO Codec Logic
Name
Pin
Type
Notes
ON Command
11
Active Low Input
Turns channel ON
OFF Command
12
Active Low Input
Turns channel OFF
TALK (to CR) Command
13
Active Low Input
Mutes channel outputs and
routes source audio to PVW
speakers
MUTE Command
14
Active Low Input
Mutes channel outputs
TALK (to SOURCE) Command
15
Active Low Input
Allows an external button
to activate channel TALK TO
SOURCE function.
ON Lamp
1
Open Collector to Logic
Com- mon Return
Illuminates when channel is
ON unless TALK or MUTE are
active
OFF Lamp
2
Open Collector to Logic
Common Return
Illuminates when channel is
OFF.
TALK (to CR) Lamp
3
Open Collector to Logic
Common Return
Illuminates when TALK is
active
MUTE Lamp
4
Open Collector to Logic
Common Return
Illuminates when MUTE is
active
TALK (to SOURCE) Lamp
5
Open Collector to Logic
Common Return
Illuminates when the channel
TALK TO SOURCE function is
active.
Source Common
7
Logic Common
Connect to ground of source
device or to Pin 8
Logic Common
8
Internal 5 Volt return
Can be connected to Pin 7 if
source is not providing common
Logic + 5 Volt supply
9
Logic Supply, Individually
Fused
Can be connected to Pin 10 if
source is not providing voltage; active only when source
has been assigned to channel.
Source Supply
10
Common for all 5 inputs
Connect to power supply of
source device or to Pin 9
NOT CONNECTED
6
INPUTS
OUTPUTS
Configuring GPIO • 92
POWER & COMMON
© 2017 Axia Audio - Rev 2.0
GPIO Telephone Hybrid Logic
Name
Pin
Type
Notes
ON Command
11
Active Low Input
Turns channel ON
OFF Command
12
Active Low Input
Turns channel OFF
PREVIEW Command
13
Active Low Input
Turns preview ON
RESET Command
14
Active Low Input
Turns channel off while not
sending a STOP pulse
READY Command
15
Active Low Input
Illuminates OFF lamp to indicate source’s readiness
ON Lamp
1
Open Collector to Logic
Common Return
Illuminates when channel is
ON
OFF Lamp
2
Open Collector to Logic
Common Return
Illuminates when channel is
OFF
PREVIEW Lamp
3
Open Collector to Logic
Common Return
Illuminates when PREVIEW
is ON
START Pulse
4
Open Collector to Logic
Common Return
A 100 ms PULSE is sent when
channel is first turned ON or
when PVW is first selected
STOP Pulse
5
Open Collector to Logic
Common Return
A 100 ms PULSE sent when
channel is turned OFF.
Source Common
7
Logic Common
Connect to ground of source
device or to Pin 8
Logic Common
8
Internal 5 Volt return
Can be connected to Pin 7 if
source is not providing common
Logic + 5 Volt supply
9
Logic Supply, Individually
Fused
Can be connected to Pin 10 if
source is not providing voltage; active only when source
has been assigned to channel.
Source Supply
10
Common for all 5 inputs
Connect to power supply of
source device or to Pin 9
NOT CONNECTED
6
INPUTS
OUTPUTS
Configuring GPIO • 93
POWER & COMMAND
© 2017 Axia Audio - Rev 2.0
GPIO Control Room Monitor Logic
Name
Pin
Type
Notes
MUTE CR Command
11
Active Low Input
Mutes CR monitors and Preview speakers
DIM CR Command
12
Active Low Input
Allows external dimming of
CR monitor speakers.
Enable EXT PREVIEW Command
13
Active Low Input
Feeds External Audio Input to
PREVIEW
TALK TO EXT Command
14
Active Low Input
Turns on Talk to External
Audio.
Not used.
15
Active Low Input
CR ON AIR Lamp
1
Open Collector to Logic
Common Return
Illuminates whenever CR
monitors are muted
DIM CR Lamp
2
Open Collector to Logic
Common Return
Illuminates whenever control
room monitors are DIMMED
PREVIEW Lamp
3
Open Collector to Logic
Common Return
Illuminates when PREVIEW is
active.
TALK TO EXT Lamp
4
Open Collector to Logic
Common Return
Illuminates when Talk to
External is active.
TALK (to CR) Active Lamp
5
Open Collector to Logic
Common Return
Active whenever a source has
activated its TALK (to CR)
function
Source Common
7
Logic Common
Connect to ground of source
device or to Pin 8
Logic Common
8
Internal 5 Volt return
Can be connected to Pin 7 if
source is not providing common
Logic + 5 Volt supply
9
Logic Supply, Individually
Fused
Can be connected to Pin 10 if
source is not providing voltage; active only when source
has been assigned to channel.
Source Supply
10
Common for all 5 inputs
Connect to power supply of
source device or to Pin 9
NOT CONNECTED
6
INPUTS
OUTPUTS
Configuring GPIO • 94
POWER & COMMON
© 2017 Axia Audio - Rev 2.0
GPIO Computer Playback Device Logic
Name
Pin
Type
Notes
ON Command
11
Active Low Input
Turns channel ON
OFF Command
12
Active Low Input
Turns channel OFF & sends
100 msec STOP pulse
PREVIEW Command
13
Active Low Input
Turns preview ON
Not Used
14
Active Low Input
READY Command
15
Active Low Input
Illuminates OFF lamp to indicate source’s readiness
NEXT Pulse
1
Open Collector to Logic
Common Return
A 100 mS PULSE sent when
ON button is depressed, except when initially turned ON.
OFF Lamp
2
Open Collector to Logic
Common Return
Illuminates when channel is
OFF and READY is active
PREVIEW Lamp
3
Open Collector to Logic
Common Return
Illuminates when PREVIEW
is ON
START Pulse
4
Open Collector to Logic
Common Return
A 100 ms PULSE sent when
channel is first turned ON.
STOP Pulse
5
Open Collector to Logic
Common Return
A 100 ms PULSE sent when
channel is turned OFF.
Source Common
7
Logic Common
Connect to ground of source
device or to Pin 8
Logic Common
8
Internal 5 Volt return
Can be connected to Pin 7 if
source is not providing common
Logic + 5 Volt supply
9
Logic Supply, Individually
Fused
Can be connected to Pin 10 if
source is not providing voltage; active only when source
has been assigned to channel.
Source Supply
10
Common for all 5 inputs
Connect to power supply of
source device or to Pin 9
NOT CONNECTED
6
INPUTS
OUTPUTS
Configuring GPIO • 95
POWER & COMMON
© 2017 Axia Audio - Rev 2.0
About GPIO Connections
I NPUT
VDC
5
6
12
24
48
External Series Resistor
0
0
680 @ 1/4 watt
1.8k @ 1/2 watt
3.9k @ 1 watt
The maximum voltage allowed for an external power supply for logic control is 48 volts DC. The use of a
current limiting resistor is required for some voltages.
If the equipment being controlled is electrically isolated, than the use of the
GPIO port’s power supply is acceptable as shown here.
Configuring GPIO • 96
Take note to use current limiting resistors per the above chart if the
voltage supplied is above 6vdc. The intention is to limit the current to
20mA for each GPI pin.
be
The GPO portion of the GPIO ports are solid state relays. Current should
limited to a combined 100mA through all the pins of a port. Maximum
allowed voltage is 24 volts. The following diagram shows the
recommended connections for outputs with the use of an external power
supply.
© 2017 Axia Audio - Rev 2.0
The Axia accessory modules use the 5vDC supply to
illuminate LED based buttons. So a one-to-one pin
connection is all that is needed between any accessory
modules and a GPIO port.
Configuring GPIO • 97
Note, all of the inputs and outputs on a specific GPIO
port are “grouped together”. The 5 “Outputs” are on 5
separate output pins, however, they share the same
“Common Return” connection on Pin #7. Similarly,
the 5 “Inputs” pins would be pulled to ground to activate them, and they share a common pin for a high-side rail,
on Pin #10. If more than one remotely-controlled device is to be connected to a single 15-pin I/O port, you must
make sure that the two units in question have the same ground potential or ground loops will occur. Therefore, it
is recommended that only one remote device be connected to each I/O port connector to assure complete electrical
isolation.
© 2017 Axia Audio - Rev 2.0
Appendix C:
Specifications
Microphone Preamplifiers
• Source Impedance: 150 Ohms
• Input Impedance: 4 k Ohms minimum, balanced
• Nominal Level Range: Adjustable, -75 dBu to -20 dBu
• Input Headroom: >20 dB above nominal input
• Output Level: +4 dBu, nominal
Analog Line Inputs
• Input Impedance: >40 k Ohms, balanced
• Nominal Level Range: Selectable, +4 dBu or -10dBv
• Input Headroom: 20 dB above nominal input
Analog Line Outputs
• Output Source Impedance: <50 Ohms balanced
• Output Load Impedance: 600 Ohms, minimum
Specifications • 99
• Nominal Output Level: +4 dBu
• Maximum Output Level: +24 dBu
© 2017 Axia Audio - Rev 2.0
Digital Audio Inputs and Outputs
• Reference Level: +4 dBu (-20 dB FSD)
• Impedance: 110 Ohms, balanced (XLR)
• Signal Format: AES-3 (AES/EBU)
• AES-3 Input Compliance: 24-bit with selectable sample rate conversion,
• 32 kHz to 96kHz input sample rate capable.
• AES-3 Output Compliance: 24-bit
• Digital Reference: Internal (network timebase) or external reference 48 kHz,
• +/- 2 ppm
• Internal Sampling Rate: 48 kHz
• Output Sample Rate: 44.1 kHz or 48 kHz
• A/D Conversions: 24-bit, Delta-Sigma, 256x oversampling
• D/A Conversions: 24-bit, Delta-Sigma, 256x oversampling
• Latency <3 ms, mic in to monitor out, including network and processor loop
Frequency Response
Specifications • 100
• Any input to any output: +0.5 / -0.5 dB, 20 Hz to 20 kHz
© 2017 Axia Audio - Rev 2.0
Dynamic Range
• Analog Input to Analog Output: 102 dB referenced to 0 dBFS,
• 105 dB “A” weighted to 0 dBFS
• Analog Input to Digital Output: 105 dB referenced to 0 dBFS
• Digital Input to Analog Output: 103 dB referenced to 0 dBFS, 106 dB “A” weighted
• Digital Input to Digital Output: 138 dB
Equivalent Input Noise
• Microphone Preamp: -128 dBu, 150 ohm source, reference -50 dBu input level
Total Harmonic Distortion + Noise
• Mic Pre Input to Analog Line Output: <0.005%, 1 kHz, -38 dBu input,
• +18 dBu output
• Analog Input to Analog Output: <0.008%, 1 kHz, +18 dBu input, +18 dBu output
• Digital Input to Digital Output: <0.0003%, 1 kHz, -20 dBFS
• Digital Input to Analog Output: <0.005%, 1 kHz, -6 dBFS input, +18 dBu output
• Analog Line channel to channel isolation: 90 dB isolation minimum, 20 Hz to 20 kH
• Microphone channel to channel isolation: 80 dB isolation minimum, 20 Hz to 20 kHz
• Analog Line Stereo separation: 85 dB isolation minimum, 20Hz to 20 kHz
• Analog Line Input CMRR: >60 dB, 20 Hz to 20 kHz
• Microphone Input CMRR: >55 dB, 20 Hz to 20 kHz
© 2017 Axia Audio - Rev 2.0
Specifications • 101
Crosstalk Isolation, Stereo Separation and CMRR
Audio Processing
Equalizer
• Frequency Bands: 20Hz to 320Hz, 125Hz to 2KHz, 1.25KHz to 20KHz.
• Cut/Boost range on each band: -25dB to +15dB.
• Q-factor: Automatic - bandwidth varies based on amount of cut or boost.
Compressor
• Threshold: -30dB to 0dB Ratio: 1:1 to 16:1
• Post-processor Trim Level: Adjustable from -20dB to +20dB
Expander/Noise Gate
• Threshold: -50dB to 0dB Ratio: -30dB to 0dB
De-esser
• Threshold: -20dB to 0dB Ratio: 1:1 to 8:1
Power Supply AC Input, StudioEngine
• Auto-sensing, field-replaceable modular supply, 90VAC to 240VAC, 50 Hz to 60 Hz, IEC receptacle, internal fuse
• Power consumption: 100 Watts
Specifications • 102
Power Supply AC Input, Element Power Supply/GPIO
• Auto-sensing supply, 90VAC to 240VAC, 50 Hz to 60 Hz, IEC receptacle, internal fuse
• Power consumption: 150 Watts
© 2017 Axia Audio - Rev 2.0
Power Supply AC Input, PowerStation Aux & Main
• Auto-sensing supply, 90VAC to 240VAC, 50 Hz to 60 Hz, IEC receptacle, internal fuse
• Power consumption: 500 Watts
Operating Temperatures
Specifications • 103
• -10 degrees C to +40 degrees C, <90% humidity, no condensation
© 2017 Axia Audio - Rev 2.0
Appendix D
CE Declaration of Conformity
CE Declaration of Conformity • 105
Declaration of Conformity
© 2017 Axia Audio - Rev 2.0
Appendix E:
Warranty
Telos Alliance Limited Warranty
This Warranty covers “the Products,” which are defined as the various audio equipment, parts, software and
accessories manufactured, sold and/or distributed by or on behalf of TLS Corp. and its affiliated companies,
collectively doing business as The Telos Alliance (hereinafter “Telos”).
With the exception of software-only items, the Products are warranted to be free from defects in material and
workmanship for a period of five (5) years from the date of receipt of such Product by the end-user (such date of
receipt the “Receipt Date”). Software-only items are warranted to be free from defects in material and workmanship
for a period of 90 days from the Receipt Date. Telos will repair or replace (in its discretion) defective Products
returned to Telos within the warranty period, subject to the provisions and limitations set forth herein.
This warranty will be void if the Product: (i) has been subjected, directly or indirectly, to Acts of God, including
(without limitation) lightning strikes or resultant power surges; (ii) has been improperly installed or misused,
including (without limitation) the failure to use telephone and power line surge protection devices; (iii) has been
damaged by accident or neglect. As with all sensitive electronic equipment, to help prevent damage and or loss
of data, we strongly recommend the use of an uninterruptible power supply (UPS) with all of our Products. Telos
products are to be used with registered protective interface devices which satisfy regulatory requirements in their
country of use.
This Warranty is void if the associated equipment was purchased or otherwise obtained through sales channels
not authorized by Telos.
EXCEPT FOR THE ABOVE-STATED EXPRESS WARRANTY, TELOS MAKES NO WARRANTIES,
EXPRESS OR IMPLIED (INCLUDING IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS
FOR A PARTICULAR PURPOSE).
In order to invoke this Warranty, the Product must be registered via Telos’ website (found at: http://telosalliance.
com/legal/warranty) at time of receipt by end-user and notice of a warranty claim must be received by Telos within
the above stated warranty period and warranty coverage must be authorized by Telos. Contact may be made via
email: [email protected] or via telephone: (+1) 216-241-7225. If Telos authorizes the performance of
warranty service, the defective Product must be delivered to: Telos, 1241 Superior Avenue, Cleveland, Ohio 44114
or other company repair center as may be specified by Telos at the time of claim.
© 2017 Axia Audio - Rev 2.0
Warranty • 107
In no event will Telos, its directors, officers, employees, agents, owners, consultants or advisors (its “Affiliates”),
or authorized dealers or their respective Affiliates, be liable for incidental or consequential damages, or for loss,
damage, or expense directly or indirectly arising from the use of any Product or the inability to use any Product
either separately or in combination with other equipment or materials, or from any other cause.
Shipping Costs and Warranty Service:
If the date the customer’s notice of warranty claim is received by Telos (such date the “Warranty Claim Notice
Date”) is within the first 90 days following the Receipt Date, Telos will pay the costs of shipping such warranted
Product to and from the end user’s location, and the cost of repair or replacement of such warranted Product.
If the Warranty Claim Notice Date occurs after the first 90 days following the Receipt Date and before the end of
the second (2nd) year, the customer will pay the freight to return the warranted Product to Telos. Telos will then, at
its sole discretion, repair or replace the warranted Product and return it to the end user at Telos’ expense.
If the Warranty Claim Notice Date occurs between the end of the second (2nd) year following the Receipt Date
and the completion of the fifth (5th) year, the customer will pay the costs of shipping such warranted Product to
and from the end user’s location. Telos will then, in its sole discretion, repair or replace the warranted Product at
Telos’ expense. Telos also reserves the right, if it is not economically justifiable to repair the warranted Product, to
offer a replacement product of comparable performance and condition direct to the customer at a discounted price,
accepting the failed warranted Product as a trade-in.
The end user will in all cases be responsible for all duties and taxes associated with the shipment, return and
servicing of the warranted Product.
No distributor, dealer, or reseller of Telos products is authorized under any circumstances to extend, expand or
otherwise modify in any way the warranty provided by Telos, and any attempt to do so is null and void and shall not
be effective as against Telos or its Affiliates.
Out of warranty units returned to the factory for repair may be subject to a $500 evaluation fee, which fee must
be prepaid prior to shipping the unit to Telos. If no repairs are required, the $500 fee will be retained by Telos as an
evaluation charge. If repairs are required, the $500 fee will be applied to the total cost of the repair.
Warranty • 108
To activate your product warranty, visit http://www.telosalliance.com/product-registration .
© 2017 Axia Audio - Rev 2.0
Appendix F
New Features
Version 3.1 adds new features to the Fusion console and those features are noted in this Appendix.
• Add support for loudness metering based on ITU.BS1770 and supporting EBU R128.
• Add support for “Live mode” control of Voco8.
• Add support for Source “Unload”. Removes a source from console that has ownership of a source,
permitting other console to take ownership.
• Add AES67 support in Studio Engine
• Support Accessory Panel Support and control of Custom Headphone backfeeds.
• Add Phase meter to display.
Loudness Metering
Appendix F • 109
With version 3.1, there is an added sub section to the customize option for enabling Loudness metering.
Loudness metering is based on ITU.BS1770. Some of the options provided are specific to EBU R128.
© 2017 Axia Audio - Rev 2.0
Enable Loudness Meter
Checking this option and pressing save will turn on loudness metering on the main display view.
The loudness meters present numerical values for the momentary (shortest time scale of 0.4s), short
(intermediate time scale, 3s), and integrated (segment-wise time scale based on timer value) scales. The maximum
value recorded for momentary and short scales within the segment time is also shown. To the right of the numerical
meters is the max recorded true peak values. Below the true peak meter is the measured source. The options for
measured source are the various mix busses of the surface (PGM1, PGM2, etc). To select another bus to view, rotate
the first encoder on the overbridge of the master module.
Target Loudness, Loudness Tolerance
Two text fields are provided for entering your desired target loudness value and the acceptable +/- deviation from
that target. These fields are used for the options below the text fields. Depending on region, these values can vary.
For example EBU R128 calls for -23 LUFS and the American Calm Act recommends -24 LKFS (LUFS = LKFS).
Relative Loudness
The relative loudness option changes the numerical presentation so that the number is in reference to the target.
A value of 0 LU indicates the target value is achieved. Lower than target value will be negative (-) LU while positive
(+) LU indicates higher than target.
Appendix F • 110
Color Indication
To have the meter change color to indicate cold, good, hot, select the Color Indication option and make sure the
Tolerance is set appropriate. Green is an indication that the loudness is within the tolerance of the target. Red is an
indication of above the tolerance value and blue is an indication of being below tolerance.
© 2017 Axia Audio - Rev 2.0
Loudness Range
Loudness range (LRA) is a supplemental measurement for loudness. It is a statistical distribution of measured
loudness that quantifies the variations. The measurement is presented in LU. A typical promo would have an LRA of
5 +/-1 LU. A typical theatrical drama would have an LRA of 15 +/-1 LU.
Control
Other than enabling the loudness meter for your facility’s needs, the program segment length is needed to be
defined by the operator or other control point. When loudness metering is enabled, the timer control buttons are used
for control of the integration timer and no longer for the count up timer. To control the up (and down) timer, press
the timer option button to control the timers with the overbridge knobs. The control of integration time will be the
START/STOP and RESET buttons. In addition, a GPIO channel number may be defined to provide control of the
integration time from another location other than the operator.
The pinout is:
Pin
Function
1
START
2
STOP
3
RESET
Voco8 control
The Omnia Voco8 is a powerful microphone processor with many option and controls. For easy recall of the
presets during a show in progress, the Fusion console provides an interface to control the Voco 8 in Live mode. For
setup, go to the Customize page and locate the sub section known as Sound4 Remote Control. The Voco8 uses the
Sound4 control protocol. To connect to the server, enter in the Server IP and the appropriate login and password.
Use the first knob to highlight which item to edit. With the second knob, you select a preset to load. By pressing
the 2nd knob inward, you will load the Sound4 session.
© 2017 Axia Audio - Rev 2.0
Appendix F • 111
From the console surface, press the Additional Options button on the Monitor/Navigation module or hold * + 4 +
8 keys for 5 seconds (Monitor+2 faders). This will open the control options view to the screen.
Source “unload”
When loading shared resources, typically a facility codec, and said resource is in ownership by another studio,
the console will load the source into Listen Only mode. This permits the console to have access of the audio from
source, but is not able to control the source or provide backfeed audio. The console will provide indication of
who is the owner of the source so that the source so action can take place if the newly requesting console can take
ownership. In some cases, the last operator of that other studio may have left, requiring someone to walk but
customers have told us they don’t want to walk down to the other studio. The Request Unload option now appears
on general Channel Option screen when a source is loaded in Listen Only mode. Pressing the Request Unload
function will open a new view requiring the operator to rotate the first knob and select the confirmation that the
source is to be unloaded. By pressing the knob in, after selecting to unload, requires a three step process to unload
the source from the other console. This is a security feature to avoid accidental operations. If the source is in the ON
state on the active owner, the source will not unload as an extra protection and the requesting console will remain in
Listen Only mode.
AES67 support
Version 3.1 introduces support for AES67 in the Studio Engine. This support is introduced in VMode and is
further discussed in the VMODE section of Chapter 4 of the manual. With VMode, it is possible to convert AES67
streams with Livewire streams.
Appendix F • 112
Accessory Panel Support
Version 3.1 also provides support of customized backfeed control with the Fusion Accessory panels. In addition,
support for a Fader panel will be introduce. With the Fader panel, a positions can control the level of their own
microphone away from the operator control or control a source other than the microphone, often a playout computer.
© 2017 Axia Audio - Rev 2.0
Phase meter
Appendix F • 113
A phase meter is introduced at the top of each mix bus meter. A presentation of full green (+1) indicates the Left
and Right channels are in phase together. A presentation of full red (-1) indicates the Left and Right channels are
out of phase with one another. A presentation of a single yellow center indicator states complete stereo separation
between the left and right channels.
© 2017 Axia Audio - Rev 2.0
The Telos Alliance • 1241 Superior Ave. • Cleveland, Ohio, 44114, USA • +1.216.241.7225 • TelosAlliance.com
© 2017 TLS Corp. Axia® The Telos Alliance.® All Rights Reserved. C16/3/16002
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