8x8 Analog Line Node 8x8 AES Line Node 8x8
8x8 Analog Line Node
8x8 AES Line Node
8x8 Microphone Node
Installation & User’s Guide
Manual Version 2.5 rev May, 2009
Node Software 2.5.2g and higher
IMPORTANT NOTE:
Axia nodes are intended for use with an Ethernet
Switch that supports multicast and QoS (Quality
of Service). On a non-switched Ethernet hub, or a
switch that is not enabled for multicast, this will result in network congestion that could disrupt other
network activity.
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 only 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 present appareil numerique n’emet pas de bruits radioelectriques depassant les limites applicables aux appareils numeriques
(de les Class A) prescrites dans le Reglement sur le
brouillage radioelectrique edicte par le ministere des
Communications du Canada.”
Important Safety Information
To reduce the risk of electrical shock, do not expose
this product to rain or moisture. Keep liquids away
from the ventilation openings in the top and rear of
the unit. Do not shower or bathe with the unit.
Caution
The installation and servicing instructions in the
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.
Electrical Warning
To prevent risk of electric shock: Disconnect power
cord before servicing.
This equipment is designed to be operated from a
power source that includes a third “grounding” connection in addition to the power leads. Do not defeat
this safety feature. In addition to creating a potentially hazardous situation, defeating this safety ground
will prevent the internal line noise filter from functioning.
Ventilation Warning
The Axia 8x8 node uses convection cooling. Do not
block the ventilation openings in the side or top of
the unit. Failure to allow proper ventilation could
damage the unit or create a fire hazard. Do not place
the unit on a carpet, bedding, or other materials that
could interfere with the rear and top panel ventilation openings.
Customer Service
We support you...
By Phone/Fax in the USA.
• Customer service is available from 9:30 AM to 6:00 PM USA Eastern Time, Monday through Friday at
+1 216.241.7225. Fax: +1 216.241.4103. The 24-hour Telos/Omnia/Axia support line is +1 216.622.0247.
By Phone/Fax in the Europe
• Service is available from Axia Europe in Germany at +49 81 61 42 467. Fax: +49 81 61 42 402.
By E-Mail.
• The address is [email protected]
Via World Wide Web.
• The Axia Web site has a variety of information which may be useful for product selection and support. The URL
is http://www.AxiaAudio.com.
Feedback
We welcome feedback on any aspect of Axia products 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.
Updates
The operation of the Axia node is determined largely by software. Periodic updates may become available - to
determine if this is the case check our web site. Contact us to determine if a newer release is more suitable to your
needs.
Our electronic newsletter has announcements of major software updates for existing products, as well as keeping
you up to date on the latest Axia, Telos, and Omnia product releases. You may subscribe to update notifications here:
http://www.axiaaudio.com/signup.htm
Trademarks
Axia Audio
2101 Superior Ave. Cleveland, OH 44114 USA
+1 (216) 241-7225
[email protected]
Axia Europe
Johannisstraβe 6, 85354 Freising, Germany
+49 81 61 42 467
[email protected]
Copyright © 2009 by TLS Corporation. Published by Axia Audio. 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. All rights reserved.
Version 2.5, May, 2009
Introduction • iii
Telos Systems, Axia Audio, Livewire, the Livewire Logo, the Axia logo, SmartSurface, Element, SmartQ, Omnia,
the Omnia logo, and the Telos logo, are trademarks of TLS Corporation. All other trademarks are the property of
their respective holders.
Notice
About This Manual
All versions, claims of compatibility, trademarks, etc.
of hardware and software products not made by Axia
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.
Warranty
This product is covered by a five year limited warranty, the full text of which is included in the rear
section of this manual.
Service
You must contact Axia before returning any equipment for factory service. 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 at Axia.
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) 241-7225. All other customers should contact their local representative
to arrange for service.
If you have not done so, please review that material first. In it we explain the ideas that motivated
Livewire and how you can use and benefit from it,
as well as nitty-gritty details about wiring, connectors, and the like. Since Livewire is built on standard
networks, we also help you to understand general
network engineering so that you have the full background for Livewire’s fundamentals. After reading
Introduction to Livewire you will know what’s up
when you are speaking with gear vendors and the
network guys that are often hanging around radio
stations these days.
As always, we welcome your suggestions for improvement. Contact Axia Audio with your comments:
Axia Audio, a Telos Company
2101 Superior Avenue
Cleveland Ohio 44114 USA
Phone: +1.216.241.7225
Web: www.AxiaAudio.com
E-Mail: [email protected]
Introduction • iv
We strongly recommend being near the unit when
you call, so our Support Engineers can verify information about your configuration and the conditions
under which the problem occurs. If the unit must
return to Axia, we will need your serial number, located on the rear panel.
This manual covers the details of the Axia 8x8 Microphone, Analog and 8x8 AES nodes. However it
is assumed in this document that you are familiar
with Livewire’s basic concepts, as outlined in the
companion Introduction to Livewire: System Design
Reference & Primer manual.
Version 2.5, May, 2009
Table of Contents
Chapter Three: Advanced Programming . . . . . . . . 9
Assigning an IP Address Remotely . . . . . . . . . . 9
Customer Service . . . . . . . . . . . . . . . . . iii
Accessing the Node’s Web Pages . . . . . . . . . . . 9
Warranty . . . . . . . . . . . . . . . . . . . . . . iv
The 8x8 Node Home Page . . . . . . . . . . . . . 10
Service . . . . . . . . . . . . . . . . . . . . . . . iv
Sources (Local Inputs) . . . . . . . . . . . . . . 10
About This Manual . . . . . . . . . . . . . . . . . iv
Source Name and Channel . . . . . . . . . . . 10
. . .
. . . . .
A Note From The Founder/CEO of Telos
A Note From The President of Axia
vii
viii
Shareable . . . . . . . . . . . . . . . . . . . . 10
Mode . . . . . . . . . . . . . . . . . . . . . . 10
Gain (dB) . . . . . . . . . . . . . . . . . . . . 11
AES Mode . . . . . . . . . . . . . . . . . . . . 12
Show Source Allocation Status . . . . . . . . . 12
Chapter One: Introducing the Axia 8x8 Family . . . . 1
Description . . . . . . . . . . . . . . . . . . . . . . 1
Phantom Power . . . . . . . . . . . . . . . . . 12
Front Panel Controls and Indicators . . . . . . . . . 1
Destinations (Local Outputs) . . . . . . . . . . . 12
Status LED indicators . . . . . . . . . . . . . . . 1
Destination Name . . . . . . . . . . . . . . . . 12
LINK . . . . . . . . . . . . . . . . . . . . . . 1
Destination Channel . . . . . . . . . . . . . . 12
LIVEWIRE . . . . . . . . . . . . . . . . . . . 1
Destination Type . . . . . . . . . . . . . . . . 12
SYNC & MASTER with the Analog and Mic Node1
Output Load . . . . . . . . . . . . . . . . . . 14
SYNC & MASTER with the AES Node . . . . . 2
Output Gain . . . . . . . . . . . . . . . . . . 14
Bargraph LED Meters . . . . . . . . . . . . . . . 2
Meters . . . . . . . . . . . . . . . . . . . . . . . 14
INPUT METERS . . . . . . . . . . . . . . . . 2
Sources . . . . . . . . . . . . . . . . . . . . . 14
OUTPUT METERS . . . . . . . . . . . . . . 2
Destinations . . . . . . . . . . . . . . . . . . 14
Select and ID Buttons . . . . . . . . . . . . . . . 2
System Parameters . . . . . . . . . . . . . . . . . 15
Rear Panel . . . . . . . . . . . . . . . . . . . . . . . 2
IP Settings . . . . . . . . . . . . . . . . . . . 15
AC (Mains) Power . . . . . . . . . . . . . . . . . 3
Host name . . . . . . . . . . . . . . . . . . . 15
Livewire (100 Base-T) Connector . . . . . . . . 3
Network address (IP Address) . . . . . . . . . 15
Input Connectors . . . . . . . . . . . . . . . . . 3
Netmask (Subnet mask) . . . . . . . . . . . . 15
Analog Line Input Characteristics . . . . . . . 4
Gateway (Router) . . . . . . . . . . . . . . . . 15
Analog Microphone Input Characteristics . . . 4
Syslog Server (IP address) . . . . . . . . . . . 15
Output Connectors . . . . . . . . . . . . . . . . . 4
Syslog severity level filter . . . . . . . . . . . . 16
Analog Output Characteristics . . . . . . . . . 4
User password . . . . . . . . . . . . . . . . . 16
AES Input Characteristics . . . . . . . . . . . 4
Firmware version . . . . . . . . . . . . . . . . 16
Saving Bank 1 Software . . . . . . . . . . . . 16
Chapter Two: The 8x8 Node’s Front Panel . . . . . . 5
QOS & Network . . . . . . . . . . . . . . . . . . 17
Configuration & Testing . . . . . . . . . . . . . . . 5
Livewire Clock Master . . . . . . . . . . . . . 17
Powering up . . . . . . . . . . . . . . . . . . . . 5
Livewire Clock Mode . . . . . . . . . . . . . . 18
Basic Programming via the Front Panel . . . . . . 5
Receive Buffer Size . . . . . . . . . . . . . . . 18
Programming the unit’s IP address . . . . . . 5
801.1p tagging, 802.1p VLAN ID, 802.1q Priority, &
Checking the Audio Node Name . . . . . . . . 5
DSCP Class of Service . . . . . . . . . . . . . 18
Checking the Audio Node Software Version . . 5
AES Synchronization and Clock . . . . . . . . 18
Programming the Node’s Streaming Mode . . . 5
AES Sync Source & AES Master Timebase . . . 18
Programming the Tx Base Channel (TxBCH) . 6
AES Output Sync . . . . . . . . . . . . . . . . 19
Programming the Rx Base Channel (RxBCH) . 6
Restoring Defaults . . . . . . . . . . . . . . . . . . 7
Bench Testing . . . . . . . . . . . . . . . . . . . . . 7
Version 2.5, May, 2009
Introduction • v
Downloading new software . . . . . . . . . . . 16
Appendix A: Unbalanced Connections . . . . . . . . 21
Unbalanced Destinations . . . . . . . . . . . . . 21
Unbalanced Sources . . . . . . . . . . . . . . . . 21
Using two nodes back to back . . . . . . . . . 23
Connecting a “remote” using Ethernet Radio . 23
Appendix B: Axia Nodes and Ethernet Radios . . . . 23
IP Radio Settings and Recommendations . . . 24
Appendix C: Troubleshooting . . . . . . . . . . . . . 25
Appendix D: Specifications and Warranty . . . . . . 27
Axia System Specifications . . . . . . . . . . . . . . 27
Microphone Preamplifiers . . . . . . . . . . . . 27
Analog Line Inputs . . . . . . . . . . . . . . . . 27
Analog Line Outputs . . . . . . . . . . . . . . . 27
Digital Audio Inputs and Outputs . . . . . . . . . 27
Frequency Response . . . . . . . . . . . . . . . . 27
Dynamic Range . . . . . . . . . . . . . . . . . . 27
Equivalent Input Noise . . . . . . . . . . . . . . 28
Total Harmonic Distortion + Noise . . . . . . . . 28
Crosstalk Isolation, Stereo Separation and CMRR 28
Power Supply AC Input . . . . . . . . . . . . . . 28
Operating Temperatures . . . . . . . . . . . . . 28
Dimensions and Weight . . . . . . . . . . . . . . 28
Introduction • vi
Axia Node Limited Warranty . . . . . . . . . . . . 29
Version 2.5, May, 2009
It’s been a tradition since Telos’ very first product, the
Telos 10 digital phone system, that I share a few words
with you at the beginning of each manual. So here goes.
In radio broadcast studios we’re still picking up the
pieces that have fallen out from the digital audio revolution. We’re not using cart machines anymore because
PCs are so clearly a better way to store and play audio.
We’re replacing our analog mixing consoles with digital ones and routing audio digitally. But we’re still using decades-old analog or primitive digital methods to
connect our gear. Livewire has been developed by Telos
to provide a modern PC and computer network-oriented
way to connect and distribute professional audio around
a broadcast studio facility.
Your question may be, “Why Telos? Don’t you guys
make phone stuff?” Yes, we certainly do. But we’ve always been attracted to new and better ways
to make things happen in radio facilities. And
we’ve always looked for opportunities to
make networks of all kinds work for broadcasters. When DSP was first possible, we used
it to fix the ages-old phone hybrid problem. It
was the first use of DSP in radio broadcasting.
When ISDN and MP3 first happened, we saw
the possibility to make a truly useful codec. We were the
first to license and use MP3 and the first to incorporate
ISDN into a codec. We were active in the early days of
internet audio, and the first to use MP3 on the internet.
Inventing and adapting new technologies for broadcast
is what we’ve always been about. And we’ve always
been marrying audio with networks. It’s been our passion right from the start. In our genes, if you will. As a
pioneer in broadcast digital audio and DSP, we’ve grown
an R&D team with a lot of creative guys who are openeyed to new ideas. So it’s actually quite natural that we
would be playing marriage broker to computer networks
and studio audio.
What you get from this is nearly as hot as a couple
on their wedding night: On one RJ-45, two-way multiple
audio channels, sophisticated control and data capability, and built-in computer compatibility. You can use
Livewire as a simple sound card replacement – an audio
interface connecting to a PC with an RJ-45 cable. But
add an Ethernet switch and more interfaces to build a
system with as many inputs and outputs as you want.
Audio may be routed directly from interface to interface or to other PCs, so you now have an audio routing
system that does everything a traditional “mainframe”
audio router does – but at a lot lower cost and with a lot
more capability. Add real-time mixing/processing engines and control surfaces and you have a modern studio
facility with many advantages over the old ways of doing
things. OK, maybe this is not as thrilling as a wedding
night – perhaps kissing your first lover is a better analogy. (By the way, and way off-topic, did you know that
the person you were kissing was 72.8% water?)
While we’re on the subject of history… you’ve probably been soldering XLRs for a long time, so you feel a
bit, shall we say, “attached” to them. We understand. But
no problem – you’ll be needing them for microphones
for a long while, so your withdrawal symptoms won’t be
serious. But your facility already has plenty
of Ethernet and plenty of computers, so you
probably already know your way around an
RJ-45 as well. It’s really not that strange to
imagine live audio flowing over computer networks, and there’s little question that you are
going to be seeing a lot of it in the coming
years.
The 20th century was remarkable for its tremendous
innovation in machines of all kinds: power generators,
heating and air conditioning, cars, airplanes, factory automation, radio, TV, computers. At the dawn of the 21st,
it’s clear that the ongoing digitization and networking of
text, audio, and images will be a main technology story
for decades to come, and an exciting ride for those of us
fortunate to be in the thick of it.
Speaking of years, it has been a lot of them since I
wrote the Zephyr manual intro, and even more since the
Telos 10 – 20 years now. Amazing thing is, with all the
change around us, I’m still here and Telos is still growing
in new ways. As, no doubt, are you and your stations.
Version 2.5, May, 2009
Steve Church
Introduction • vii
A Note From The Founder/CEO of Telos
Introduction • viii
A Note From The President of Axia
20 years ago, I designed my first broadcast console
for PR&E. I look back on that time with great fondness;
we were building bullet-proof boards for the world’s
most prestigious broadcasters, making each new console
design bigger and fancier to accommodate a wider variety of source equipment and programming styles. The
console was the core of the studio; all other equipment
was on the periphery.
Then things changed: the PC found its way into broadcast audio delivery and production. At first, PC audio
applications were simple, used only by budget stations
to reduce operating expenses. But soon the applications
evolved and were embraced by larger stations. Slowly,
the PC was taking center stage in the radio studio.
Like many, I was captivated by the PC. Stations retired carts, phonographs, open-reel decks, cassettes —
even more modern digital equipment such
as DAT and CD players, replacing all with
PC apps. Client/server systems emerged and
entire facilities began using PCs to provide
most – or all – of their recorded audio. Yet
consoles continued to treat PCs as nothing
more than audio peripherals. I knew that we
console designers were going to have to rethink our designs to deal with computer-centric studios.
During this time, traditional broadcast console companies began producing digital versions. But early digital consoles were nearly identical in form and function
to their analog predecessors. It took a fresh look from a
European company outside broadcasting to merge two
products – audio routing switchers and broadcast consoles – into a central processing engine and attached
control surface. Eventually nearly every console and
routing switcher company followed suit, and a wide variety of digital “engines” and control surfaces flooded
the market.
But, advanced as these integrated systems were, they
still handled computer-based audio sources like their
analog ancestors. Sure, the router and console engine
were now integrated, but the most important studio element – the PC – was stuck in the past, interfaced with
100-year-old analog technology. The PC and console
couldn’t communicate in a meaningful way – strange,
considering that PCs everywhere were being networked,
fast becoming the world’s most popular and powerful
communication tool.
Then a group of Telos engineers developed a method
of using Ethernet to network real-time audio devices, allowing computers and consoles, controllers and peripherals to interact smoothly and intelligently. Powerful, flexible networks had finally come to our studios. As with
the transition from carts to computers, the benefits are
many and impressive. A few networked components can
replace routing switchers, consoles, processing peripherals, sound cards, distribution amps, selector switches
and myriad related devices.
This deceptively simple networked system costs a
fraction of other approaches, yet has capabilities surpassing anything else. The system is modular and can
be used to perform discrete functions in a traditional
environment. Concurrently, it easily scales to serve both
the humblest and the very largest of facilities.
Console, router, and computer work in harmony.
So, equipped with this new technology
and countless ideas, we launch Axia, the newest division of Telos. Axia is all about delivering innovative networked audio products to
future-minded broadcasters. On behalf of our
entire team, I welcome you as a charter client. Axia is
the culmination of nearly 40 man-years of some of the
most ambitious R&D ever applied to the radio industry.
And this is only the beginning. We have more products,
innovations, and partnerships in the pipeline.
You already know your Axia system is unlike anything else. So it shouldn’t be surprising that your new
system is loaded with new thinking, new approaches,
and new ideas in virtually every conceivable area. Some
concepts will challenge your traditional ideas of studio
audio systems, but we’re certain that once you have experienced the pleasures of the networked studio, you’ll
never want to go back. And now, for something completely different...
Version 2.5, May, 2009
Michael “Catfish” Dosch
Chapter One:
Introducing the Axia 8x8
Family: Microphone, Analog
Line and AES Nodes
This section will allow you to get to know the 8x8
Node and describes the unit’s features, display, and connectors.
Description
NOTE: Only approved and properly programmed
Ethernet switches incorporating the proper Multicast and QoS standards should be used. See
www.AxiaAudio.com/switches/ for details.
Front Panel Controls and Indicators
The Livewire 8x8 Audio Node incorporates a number
of front panel indicators to allow the operator to verify
proper operation quickly and confidently.
Status LED indicators
All 8x8 Nodes can create 8 Livewire streams, each
of which becomes available to other devices on the
Livewire network. Each output (destination) can be assigned to deliver a Livewire stream acquired from the
network.
Basic point-to-point (e.g. “Livewire Snake”) applications require only two Livewire nodes and a CAT-5e or
CAT- 6 “Crossover Ethernet cable”. More sophisticated
multipoint networks can be built by connecting multiple
Livewire nodes to an appropriate Ethernet switch.
Four LEDs indicate the status of the Livewire and
Ethernet connections, as well as system synchronisation
as follows:
LINK
When illuminated continuously, this LED represents
the presence of a live Ethernet link to another Ethernet
100Base-T device. If no Ethernet link is present, this
LED flashes slowly.
LIVEWIRE
This LED indicates that the connected Ethernet segment has Livewire traffic present. If the LINK LED is
illuminated, and the LIVEWIRE LED fails to illuminate,
there are either no other Livewire devices connected, or
the Ethernet switch has not been programmed to pass
such traffic to the port to which this node is connected.
SYNC & MASTER with the Analog and Mic Node
Only one of these two LEDs should be illuminated,
with the one exception noted below. If neither LED il-
Figure 1-1: 8x8 Node - Front Panel
Version 2.5 May, 2009
1: Introducing the 8x8 Nodes • 1
The Axia 8x8 Audio Node has eight inputs and eight
outputs. These can generally be configured as stereo,
dual mono, 5.1 Surround, or a combination of the above.
Analog Line Nodes have balanced stereo analog inputs
and outputs whereas the AES Nodes have AES3 inputs
and outputs. Microphone nodes have 8 balanced, mono
microphone level inputs and eight balanced, stereo line
level outputs.
luminates, something is not correct. The SYNC LED
indicates the receipt of clock information from another
(Master) Livewire Node. The MASTER LED indicates
that this node is acting as the master clock source for the
Livewire network. More specifically:
SYNC – If Sync packets are being received by the
Livewire node, this LED will begin to flash. The
LED will continue to flash until the Livewire node
has locked its local clock to the network master.
Once the local node’s PLL is locked, the sync LED
will illuminate solidly.
1: Introducing the 8x8 Nodes • 2
MASTER – The Livewire system employs a sophisticated master/slave clocking system over the Ethernet network. By default, the system auto-selects a
clock master, however this can be changed if desired.
The system has the ability to automatically change
to a different clock master should the current master
become disconnected, or otherwise inoperable. This
happens transparently, without audio artifacts. This
LED indicates that this node is currently acting as
sync MASTER. There is only one MASTER permitted in an Axia network. If more than one device has
a MASTER LED illuminated, you may have wiring
problems or errors on your switch configuration.
SYNC & MASTER with the AES Node
When the AES Node is used in its default configuration, the SYNC and MASTER LEDs operate as described above. However, if the “AES sync input as
Livewire master timebase” option is set to YES from the
QoS web pages (see section 3, Advanced Programming)
these lights will operate differently, as described here:
SYNC – If a valid AES signal is received by the input
designated as AES sync source in the QoS web page,
this LED will begin to flash. The LED will continue
to flash until the Livewire node has locked its local
clock to the AES input signal. Once the local node’s
PLL is locked, the LED will illuminate solidly.
MASTER – This LED indicates that this node is
currently acting as MASTER clock source for the
Livewire network.
Bargraph LED Meters
The 8x8 Audio Nodes have a bargraph meter for each
input and each output.
INPUT METERS
The left eight meter-pairs represent the left and right
channels of each input. The meters are continuously
active, and indicate that audio is present at the associated input. The lowest LED segment will illuminate at
a signal level of -42 dBfs. The top-most LED segment
represents a level of 0 dBfs. This segment should not illuminate except in cases of extreme input overload.
OUTPUT METERS
The right-most 8 LED meter pairs represent the left
and right channels of each audio stream being received.
Each meter is associated with the audio on the corresponding output. The meter calibration is the same as
for the input meters.
The lowest Output LED segment has a special meaning. These LEDs will be illuminated whenever there is a
stream designated for that channel, even when audio is
not present. This acts as a “confidence meter” to indicate
that a valid output stream has in fact been assigned to
that particular output port.
Select and ID Buttons
These buttons are used to display and configure basic
functions of the node. Their functions will be discussed
in more detail in Chapter 2.
Rear Panel
The rear panel of both the Analog Node and the AES
Node are essentially identical and are pictured in Figure
1-2. The Microphone node uses 3-pin XLR-type connectors for the microphone input connectors and RJ-45
connectors for outputs similar to the Analog and AES
nodes. The rear panel of a Microphone node is shown in
Figure 1-4.
AC (Mains) Power
The AC receptacle connects mains power to the unit
Version 2.5, May, 2009
Figure 1-2: Analog and AES Node - Rear Panel
miniature modular jack (e.g. RJ45 style). The connector
pin functions are the same for both the AES and Analog
nodes and are as follows:
1
IMPORTANT! As with any piece of modern
electronic gear, it is advisable that precautions
be taken to prevent damage caused by power
surges. Standard line surge protectors can be
used to offer some degree of protection. It is
the user’s responsibility to ensure protection
adequate for their conditions is provided. This
equipment is designed to be operated from a
power source which includes a third “grounding” connection in addition to the power leads.
Do not defeat this safety feature. In addition to
creating a potentially hazardous situation, defeating this safety ground will prevent the internal line noise filter from functioning.
Livewire (100 Base-T) Connector
This connector is for connection to another Livewire
node, or an approved Ethernet switch. It has two integral
LEDs. The green “Link” LED indicates the presence of
a live signal (same as the front panel “Link” LED). The
“Activity” LED indicates that Ethernet packets are being
sent or received over the link.
IMPORTANT NOTE: Axia nodes are intended for
use with an Ethernet Switch that supports multicast and QOS (Quality of Service). If you attempt
to use them with non-switched Ethernet hubs,
or a switch that is not enabled for multicast, you
will experience network congestion that could
disrupt other network activity. Should you wish to connect a node to a nonLivewire network for access to the web configuration interface, etc, you must first confirm that
streaming is disabled as described in Chapter 2.
Input Connectors
All input and output connections to the Analog and
AES nodes are dual channel connectors, normally used
as L/R stereo pairs. Each pair of audio inputs on the
Analog Line and AES nodes share an 8-position / 8-pin
8
Figure 1-3: RJ-45 Pin Locations
IMPORTANT NOTE: Axia recommends using
balanced audio for analog audio connections.
If unbalanced sources are to be connected to
these inputs, we strongly recommend using a
balun (transformer) or balanced-to-unbalanced
buffer amplifier at the source device. Such
devices are readily available, for example the
­StudioHub “Match Jack”.
LINE and AES INPUT CONNECTORS
Pin
Function: Analog/AES
1
Left Channel Input + /AES +
2
Left Channel Input - /AES -
3
Right Channel Input +
4
Not Connected
5
Not Connected
6
Right Channel Input -
7
Not Connected
8
Not Connected
Each input of the Microphone node is a standard female XLR-3 connector. 48 Volt phantom power is available and is enabled through the node’s configuration
Version 2.5 May, 2009
1: Introducing the 8x8 Nodes • 3
with a standard IEC power cord. The power supply has a
“universal” AC input, accepting a range from 85 to 265
VAC, 47-63 Hz. A fuse is located inside on the power
supply circuit board
Figure 1-4: Microphone Node - Rear Panel
Output Connectors
web pages (more on this later, see section 3).
Each stereo audio output uses an 8-position / 8 pin
miniature modular jack (e.g. RJ45 style). The connector pin functions are the same for the Microphone, AES
and Analog nodes. Input and Output pin assignments are
identical (see above).
MICROPHONE INPUT (XLR-3)
Pin
Function
1
Signal Common
2
Signal +
3
Signal -
Analog Output Characteristics
• Level – +4 dBu (+24 dBu clip point)
• Impedance – < 50 Ohm
Analog Line Input Characteristics
Level: +4 dBu nom (+24 dBu clip point)
Impedance : >/= 10 K Ohm balanced.
AES Input Characteristics
• Balanced 110 _ (XLR)
• AES3/EBU Compliant
Analog Microphone Input Characteristics
Level: -83 to -28 dBu nominal, adjustable in 0.1 dB
steps
Headroom: 20 dB above nominal
Impedance: ≥ 4 K Ohm balanced
That’s the “10,000-foot view” of the Microphone,
Analog and AES Nodes. In Chapter 2, we'll learn how
we can set up a brand-new Node out-of-the-box, without
ever touching a PC! q
1: Introducing the 8x8 Nodes • 4
For additional technical information please see the
Specifications section.
What’s Next
Version 2.5, May, 2009
Chapter Two:
1.
Operation Via the 8x8
Node’s Front Panel
2.
In this chapter, we’ll cover everything you need to
get a new Node up and running using only the unit’s
front panel controls.
3.
Configuration & Testing
Although all Axia Audio Nodes have built-in web
servers for configuration and administration (whose use
is covered in Chapter 3), you can set up basic node functions using only the front-panel controls — handy for
those times when no PC is available. Note that many
users will prefer to give the node an IP address using
the front panel controls and then perform the rest of the
configuration from the node’s web pages.
4.
5.
To program the node’s IP address follow these steps:
Press the <SELECT> button once. The IP address is
displayed; “0.0.0.0” is the factory default, so unless
the unit has previously been programmed, the screen
will show “000.000.000.000”.
Press and hold the <ID> button for 4 seconds. A
blinking cursor will appear below the first digit. Use
<SELECT> to change the digit indicated by the cursor (each press of this number will increment the displayed digit by one).
Press the <ID> button to jump to the next digit. Use
<SELECT> to change the digit indicated by the cursor. Continue until all digits of the IP address have
been entered.
Once the changes are complete, press the <ID> button repeatedly until no cursor is shown then press
<SELECT> to exit.
If you do not wish to save your changes do not press
<ID> after reaching the last digit. After approximately 10 seconds the display will return to the meter
screen and the previous settings will be restored. If
the <ID> button is pressed when the cursor is in the
last position, the new setting is saved.
When the Audio Node is powered on, you should observe the following:
• The 4 front-panel LED indicators should illuminate briefly,
• The LED meter display will perform a screen test,
• The node’s name will display (default name:
“LiveIO”),
• And after a brief period of time, the normal front
panel bargraph meters will display.
Checking the Audio Node Name
With the meter screen displayed, press <SELECT>
twice. The name of this Audio Node will be displayed.
The default is LiveIO. You may change the name of a
node by using the Browser interface (see Chapter 3).
Checking the Audio Node Software Version
With the meter screen displayed, press <SELECT>
three times. The software version will be will be displayed.
Basic Programming via the Front Panel
Basic programming of the Audio Node can be accomplished using the front panel display and the SELECT and ID buttons. The node’s name and other information can also be checked from the front panel.
Programming the unit’s IP address
Each Audio Node must have a unique IP address.
The only exception is when two nodes are connected in
the point-to-point (snake) configuration.
Programming the Node’s Streaming Mode
The streaming mode can be selected from the front
panel. The default mode is to have streaming disabled
(e.g. OFF), thereby allowing safe connection to a computer or LAN for programming. Possible settings for
streaming mode are:
•
OFF – This disables both streaming of Livewire audio and Livewire clock packets. This setting is useful
when you wish to connect the unit to a non-Livewire-
Version 2.5 May, 2009
2: Operation Via the 8x8's Front Panel • 5
Powering up
•
•
•
2: Operation Via the 8x8's Front Panel • 6
•
LAN or directly to a computer to configure the IP
address and other settings.
LIVE – This forces all sources to be enabled in the
Live Stereo mode and also permits clock packet generation. This is the typical setting for most Livewire
audio network applications.
STD (STANDARD) – This forces all sources to
be enabled in the Standard Stereo (slow) mode and
Livewire clock packet generation. This is the usual
setting for a Axia node used in conjunction with a
computer or other device where a small amount of
additional latency is insignificant.
SNAKE – The mode is used when two Axia nodes are
directly connected with a null Ethernet cable. This
enables Livestreams and Livewire clock packets only.
SNAKE mode differs from FAST mode by disallowing local loopback capability in the node.
CUSTOM – This indicates that the Streaming Mode
parameters are mixed have been configured using the
unit’s web page interface.
To change the Streaming Mode from the front panel
follow these steps:
1. Press the <SELECT> button repeatedly until MODE
is displayed.
2. Press and hold the <ID> button until a cursor appears
under the first letter of the current setting’s name.
3. Press <SELECT> to cycle through the choices, and
<ID> to confirm your entry.
About “Transmit” and “Receive”: The following sections refer to “receive” and “transmit”
channels in an Axia Audio Node. This concept needs
a little explanation.
Audio Node inputs take the audio from connected devices and transmit that audio
to the network (sources). Conversely, Audio Nodes receive Livewire audio streams
from the network and present them on the
Node’s audio outputs (destinations). So when we refer to “Transmit” channels, we’re
talking about channel (Livewire sources) fed by
Node Inputs, and “Receive” channels refer to
channels fed to Node Outputs. All clear?
Programming the Transmit Base Channel (TxBCH)
The Transmit Base Channel is the number assigned
to the first of the eight streams to be transmitted by this
unit. If the Transmit Base Channel is set to 00101 then
the system’s eight streams would be on channel numbers
101 through 108. Non-contiguous numbering is possible,
however in that case they must be assigned using the
node’s browser user interface, see section 3 Advanced
Programming.
If you have read the Introduction to Livewire; System Design Reference & Primer manual, you will know
that each Livewire stream must have a unique channel
number, so don’t forget that now.
To program the node’s Transmit Base Channel follow these steps:
1. Starting from the metering screen, press the <SELECT> button 5 times. The TxBCH value will display; factory default is “00001”, so unless the unit
has previously been programmed, the display will
show that value.
2. Press and hold the <ID> button for 4 seconds. A
blinking cursor will appear below the first digit. Use
<SELECT> to change the digit indicated by the cursor (each press of this number will increment the displayed digit by one).
3. Press the <ID> button to jump to the next digit. Use
<SELECT> to change the digit indicated by the cursor. Continue until all digits of the TxBCH have been
entered.
4. Once the changes are complete, press the <ID> button repeatedly until no cursor is shown then press
<SELECT> to exit.
If you do not wish to save your changes, do not press
<ID> after reaching the last digit. After approximately
10 seconds the display will return to the meter screen and
the old settings will be restored.
Programming the Receive Base Channel (RxBCH)
The Receive Base Channel is the number assigned to
the first of the eight streams to be received by this node
and output from its audio connectors. If the Receive
Base Channel is set to 00201 then the system will search
the Livewire network for eight streams on channel numbers 00201 through 00208. If a designated stream is
present, the node will indicate this by illuminating the
Version 2.5, May, 2009
Of course, in many scenarios you will want to use
non-contiguous numbering for receive channels. You
can assign these using the node’s Web interface as described in Chapter 3, but you can use the method below
just to get a basic setup without using a PC.
To program the node’s Receive Base Channel follow
these steps:
1. Starting from the metering screen, press the <SELECT> button 6 times. The default RxBCH is
“00001”, so unless the unit has previously been programmed, the screen will show that entry.
2. Press and hold the <ID> button for 8 seconds. A
blinking cursor will appear below the first digit. Use
<SELECT> to change the digit indicated by the cursor (each press of this number will increment the displayed digit by one).
3. Press the <ID> button to jump to the next digit. Use
<SELECT> to change the digit indicated by the cursor. Continue until all digits of the TxBCH have been
entered.
4. Once the changes are complete, press the <ID> button repeatedly until no cursor is shown then press
<SELECT> to exit.
5. If you do not wish to save your changes do not press
<ID> after reaching the last digit. After approximately 10 seconds the display will return to the meter
screen and the old settings will be restored.
Restoring Defaults
To restore an Audio Node to its default settings follow the steps below:
• Power the node OFF.
• Depress and hold the <SELECT> and <ID> buttons.
• Power ON the unit while continuing to hold the
above buttons.
• After about 8 seconds will see the word “RESET
3 S” displayed. If you release the buttons within 3
seconds, no changes will occur. If you continue to
hold the buttons, after 3 seconds the default set-
tings will be set and “REBOOT” will be displayed.
At this time release the <SELECT> and <ID> buttons. The node is now reset to factory defaults.
Bench Testing
Two Audio Nodes may be connected together in
“Point to point” mode (e.g. Ethernet snake mode) to verify operation of the units. When connected in this way
the audio fed to “input 1” of “node A” will be output on
“output 1” of “node B” whereas the audio on “input 1” of
“node B” will be output on “output 1” of “node A”. Likewise, the other inputs for the two nodes will be mapped
correspondingly. To connect two units in “point to point”
fashion follows these steps:
1. Restore default settings on both of the Audio Nodes
to be connected. See “Restore Defaults”, above.
2. Enable streaming on each unit as follows: Press the
<SELECT> button repeatedly until MODE is displayed. Press and hold the <ID> button until a cursor appears under the current setting (e.g. Off). Press
<SELECT> repeatedly until SNAKE is displayed.
Press <ID> to confirm your entry.
3. Connect the two units using a “Crossover 10/100
Base-T” CAT-5e or CAT-6 cable, 100 meters maximum (328 feet).
4. The LINK and LIVEWIRE LEDs should illuminate
on both nodes. The MASTER LED should illuminate on one unit and the SYNC LED should illuminate on the other unit.
5. The eight Output meters on each of the two units
should show the lowest segment illuminated to indicate streams are being received.
6. Audio may now be fed into each input and will be received on the corresponding output of the other unit.
What’s Next
You’ve learned the basic front-panel operation of
Axia Audio Nodes. In Chapter 3, we'll take a tour of the
Audio Node Web Interface, where advanced options can
be set to customize your Node. q
Version 2.5 May, 2009
2: Operation Via the 8x8's Front Panel • 7
lowest LED segment on the corresponding output meter.
In our youth we
never dreamed that, one day, streams
2: Operation Via the 8x8's Front Panel • 8
might not have water.
Version 2.5, May, 2009
Advanced Programming
This chapter will walk you through the use of the
Audio Node’s built in web pages to configure advanced
features quickly, easily. — and remotely!
Assigning an IP Address Remotely
If you have not already assigned an IP address via
the front panel as described in Chapter 2, the node’s IP
address can be assigned via computer using a utility program called BootP that’s available in the Support section
of the Axia Audio web site.
1.
2.
3.
4.
To do so follow these steps:
Download and save the BootPS program. Temporarily disable your Windows Firewall. Double-click the
bootps.exe program. A DOS window will open.
Press the <ID> button on the Audio Node’s front
panel. bootps.exe will recognize the button press,
display the existing IP address and prompt you for
new IP address entry.
Enter the desired new IP address and press
<ENTER> on your computer keyboard.
Make note of the IP address you have entered.
All of the node’s parameters may be configured using
the Audio Node’s Web configuration pages. To access
the Web server from a computer, the computer and node
must be connected to the same LAN (or, the computer
and Node can be connected using a “crossover 10/100
Base-T” Ethernet cable). To connect, open your web
browser and enter the IP address of the node to be configured. Your browser should now display the node’s
home page, with links to the various functions available
A few things to remember: We assume you
know the basics of network architecture, but we
must mention that the first three numbers of the
IP address of the computer you are using will normally match those of the Node you are attempting to configure; i.e., 192.168.15.xxx. If they
don’t, the gear won’t be able to communicate
and you’ll just get frustrated. Microsoft Internet Explorer 5 and later, and
Mozilla Firefox 1.0 and later have been tested
with Axia Audio Nodes. Other browsers may work,
however they have not been tested.
Your browser must have the Java runtime library installed and enabled, and must allow “pop up” windows and display our meters. To obtain the Java runtime, visit
www.java.com .
You can now continue to assign additional
Node IP addresses, or shut down the bootps.
exe program.
Note that Axia’s iProbe software also contains BootP and using iProbe is another way to
assign IP addresses to Axia 8x8 nodes. Please
refer to the iProbe manual for details.
Figure 3-1: 8x8 Node - Home Page
Version 2.5 May, 2009
3: Advanced Programming • 9
Chapter Three:
Accessing the Node’s Web Pages
The 8x8 Node Home Page
The home page for Microphone, Analog Line and
AES nodes are identical. The home page of each node
gives you access to each of the configuration pages.
Let’s go through them now.
When you click on any link, you’ll be prompted for
a login and password. The default user name for all Axia
nodes is “user”. Other valid default user names (for
nodes running current software) are “USER”, “axia”
and “AXIA”. Leave the password field blank and click
OK. Once you have successfully logged in, you may
access any of the node’s web pages.
Sources (Local Inputs)
This is where you configure the local inputs to this
node, and assign Livewire channels and parameters to
each source. Once configuration is complete (or at any
time in the configuration process) click on Apply to save
your changes to the node.
The Sources screen for the 8x8 Microphone, Analog
and 8x8 AES nodes are very similar. The Analog 8x8
Node Sources screen is shown. All options will be discussed.
3: Advanced Programming • 10
Source Name and Channel
As described in the Introduction to Livewire; Sys-
tem Design Reference & Primer manual, each Livewire
stream must be assigned a unique channel number. The
channel number must be a number between 1 and 32767.
Livewire names may contain any printable character or spaces and can be up to 24 characters long (when
entering names excess characters will be truncated to 24
characters). Note however, that the displays on some Audio Nodes can display only 10 or 16 characters. In this
case the left-most characters will be displayed, so keep
this in mind.
You will want to develop a logical naming plan
for your facility. For example you may wish to
include the studio or rack name as part of your
names to make life simpler when identifying
sources in the future. We give some examples
in the Introduction to Livewire manual.
Shareable
This is a feature provided for backward compatibility with SmartSurface consoles. This interlock prevents
multiple consoles from sending simultaneous backfeeds
or logic commands to a single source. A red lock indicates a console has locked the source and it is available
to other consoles in listen-only mode.
Set all Node “Sharable” fields to “No” if you are
using Element consoles running v2.0 or later software
since the Element handles source sharing.
Figure 3-2: 8x8 Node - Sources Page
Version 2.5, May, 2009
Mode
Livewire sources can be Stereo, Mono or
Surround and either Standard or Live. They
can also be Enabled, or Disabled (we recommend leaving unused I/O Disabled to keep
from generating empty audio streams).
•
Standard Stereo – Generates a stereo
source. Use this for CD players, computers,
and other common sources.
•
Live Stereo – Generates a low latency stereo source. Use this for microphones,
phones, air monitors and other monitored
“live” sources.
•
Standard Mono – Generates mono
sources. In this mode, dual-mono audio sig-
Gain (dB)
This allows you to change the input
gain in the digital domain. Care must
be taken to ensure peak levels do not
exceed 0 dBfs to prevent clipping. For
Analog and AES nodes, up to +/- 12
dB of Gain may be selected in steps of
0.1 dB. Enter a value and click apply
to make the change. For Microphone
nodes, the gain range is from +18 to
+83. Most microphones will require a
gain setting of about +50.
The analog Line inputs clip point
remains at +24 dBu so this level
must not be exceeded. The rare
device with a clip point in excess
of +24 dBu will require an external pad.
Figure 3-3: 8x8 Node - Surround Mode
•
•
Tip: Standard (Slow) streams conserve network
bandwidth and are a better choice for delivering
audio to computers for recording and playback.
Analog nodes – The Gain setting
may be use to adjust for differing peak output levels
between different “+4 nominal” equipment, or it can be
used to accommodate analog sources that are below +4
nominal levels.
•
The default setting of 0dB accommodates input levels at nominal levels of +4 dBu with a clip point
of 24 dBu (e.g. 20 dB headroom). When feeding the node’s inputs from a “+4 nominal” device
that clips at some lower level (for example +18
dBu, e.g. 14 dB headroom), you can increase
the gain (by 6 dB in our example). to match this
device’s clip point to the node’s 0 dBfs point.
•
This adjustment can also be used in cases where a
low-level signal source must be used. Again you
should use the rated clip point of the source device
to determine the closest setting. Simply add gain to
bring this rated clip point up to +24 dBu.
AES nodes – in the case of AES nodes, this adjustment can be used to adjust system headroom. This
should be done with care and deliberation.
While we don’t usually recommend setting levels
“by eye,” if you choose to do so, you can view the Source
Version 2.5 May, 2009
3: Advanced Programming • 11
•
nals may be connected to node inputs 1, 3, 5 and
7. Livewire sources 1, 3, 5 and 7 are generated
from the left channels of inputs 1, 3, 5 and 7
respectively. Livewire sources 2, 4, 6 and 8 are
generated from the right channels of inputs 1, 3,
5 and 7 respectively.
Live Mono – Generates low-latency mono
sources. In this mode, dual-mono audio signals
may be connected to node inputs 1, 3, 5 and 7.
Livewire sources 1, 3, 5 and 7 are generated
from the left channels of inputs 1, 3, 5 and 7
respectively. Livewire sources 2, 4, 6 and 8 are
generated from the right channels of inputs 1, 3,
5 and 7 respectively.
Surround – Enables “5.1+Stereo” surround
mode as shown in Figure 3-3. This choice is
only available on ports 1 and 5. Selecting surround mode creates a bundle of 4 ports: 1-4 or
5-8.
Disabled – Audio source is disabled, no source
is generated and no network bandwidth used.
levels and adjust the Gain setting from the Meters page,
see below.
AES Mode
This option is only available on the Sources page for
the AES node. There are two possible options for the
setting:
•
•
Asynchronous – this is the usual setting and enables
sample rate conversion. Any valid AES source can
be used in this mode without concerns about dropouts due to mismatched clocks.
Synchronous – this setting can be used if the device
transmitting the AES signal is synchronized to the
Livewire network. For example, if the device synchronized its outputs to its input, and the input were
fed from an Axia AES node or, if the device had a
Sync input fed from an Axia AES node. Enabling
the Synchronous mode turns off sample rate conversion thereby reducing latency. This is perfect for
use with digital microphones or for “purist” applications.
3: Advanced Programming • 12
Show Source Allocation Status
Click on this link to display the console and fader to
which the source(s) on this node are assigned. This is
helpful when tracking down a source that’s reported as
being “locked” (non-sharable).
Phantom Power
If you are using phantom-powered condenser microphones, you can enable 48 VDC phantom power for individual channels here. It is recommended that you enable
phantom power only if it is required by your microphone.
Plugging in phantom powered microphones “hot” is not
recommended as the resulting transients can damage external equipment or your hearing!
Destinations (Local Outputs)
The page permits entering information related to this
node’s local outputs. Node outputs are always destinations to which Livewire audio streams are delivered. You
can name these outputs and select the stream to be delivered to each output.
The Destinations screen for the 8x8 Microphone,
Analog and 8x8 AES nodes are very similar. The Analog
8x8 Node Destinations screen is shown in Figure 3-5.
Destination Name
This is the name used to identify this destination (local output) within the Livewire network. While these
names are optional, we encourage you use them to describe what is wired to the node output.
Destination Channel
These are the Livewire channels to be routed to each
local output. If the channel to be output is not yet available on the network, you can manually enter the channel number here. In the usual case you can click on the
choose channel button to the right of this field, and a
Select Source screen will be displayed similar to the one
shown in Figure 3-6.
You can now click on the name or channel number of
the desired source to assign it to this Destination (local
output).
Figure 3-4: 8x8 Node - Source Allocation
Destination Type
There are five choices for this setting. Let’s take a
look at each one.
•
From Source: Stereo output of the same type,
Live or Standard, as the source.
• To Source: Backfeed to a bidirectional audio
Version 2.5, May, 2009
Figure 3-6: 8x8 Node - Destinations Pop-up.
•
source such as a phone or codec.
From Source: Dual Mono – Destination pairs
(1,2), (3,4), (5,6), (7,8) can be configured in dual
mono mode by changing type/mode of the oddnumbered port. The following options are avail-
We have used the term Backfeed in our discussion
above. Let us regress for a moment and review backfeeds.
You will recall from the Introduction to Livewire; System Design Reference & Primer manual that Livewire
permits special bidirectional streams for use with cases
where a source and destination are associated, such as a
codec or phone hybrid. The return feed to such devices
is usually a mix-minus (backfeed) generated by a mix
engine fed back to the device that is the primary audio
source (and usually the name of the stream in question).
In effect you have thus created a bidirectional Livewire
channel with a single channel number.
What does this all mean in practice? If the destination is a codec or hybrid you’ll set the Destination Type
Version 2.5 May, 2009
3: Advanced Programming • 13
able: “From source: Dual
Mono”, “To source: Dual
Mono”. The Left channel from Livewire stream
corresponding to the oddnumbered port feeds the
Left output. Left channel
from Livewire stream corresponding to the evennumbered port feeds Right
output. Both outputs will
deliver the same dualmono audio signal.
•
To Source: Dual
Mono - same channel
grouping scheme as above
but applies to backfeeds.
This type may be useful
to create dual-mono codec
Figure 3-5: 8x8 Analog Line Node - Destinations Page
backfeeds.
•
Surround:
Front
L, R – Stereo output of the same type, Live or
Standard, as the source. Used to create a group
of four outputs only for surround applications.
When Surround: Front L,R is selected for
Destination 1, destinations 2, 3 and 4 will automatically be assigned to Center, LFE; Back
L, R and Downmix Stereo L.R respectively. A
similar grouping applies to outputs 5 through 8.
to To Source and use the same Channel number as the
stream representing the Codec or Hybrids output (the
caller or far end codec audio).
Output Load
This option is only available on the 8x8 Analog
node’s Destinations screen. This setting has two options.
The usual selection is Hi-Z and is used when the node’s
outputs are fed to High impedance destination devices.
If the node is feeding 600 Ohm inputs (very rare these
days) the 600 ohm option should be selected. This boosts
the node’s output level by ~1 dB to maintain true +4 levels into 600 Ohm equipment to ensure unity gain. The
clip point remains at 24 dB.
Output Gain
An output gain control is provided to make adjustments that may be required if external equipment needs
a level other than +4dB. Signal level throughout an Axia
system should be +4dB since we know you will have
normalized these levels by adjusting source gain if necessary. The range of the output gain is +/- 12 dB. This
adjustment is commonly used for connections to consumer-grade or other
equipment that may
have unusual signal
levels.
each input or output. Note that the levels shown are in
the digital domain, and are therefore calibrated in dBfs.
The color-coding of the meters is somewhat arbitrary;
the meters turn red approximately 9 dB before the clip
point (e.g. 9 dB below digital full scale) but this does not
represent an overload condition.
Sources
While we recommend setting the gain setting for inputs based on the peak output clip point of the source
equipment (see Source Page, above), you can use the
meter screen to tweak input Gain settings “by eye” if
desired. Use the large arrows to adjust the level by 1 dB,
use the small arrows to adjust the level by .1 dB.
Destinations
These eight pairs of meters represent the Livewire
streams being output from this node. These are primarily for confidence monitoring. As with the front panel,
the lower-most segment indicates that the designated
Livewire stream is present, even if no audio is currently
playing. Use the large arrows to adjust the level by 1 dB,
use the small arrows to adjust the level by .1 dB.
3: Advanced Programming • 14
Meters
The
Meters
screen, shown in Figure 3-7 , is a metering
screen that shows the
audio level of all local
sources (local inputs)
and destinations (local outputs) for the
node. The screen is
divided into two sections, with inputs on
the left and outputs
on the right. Each
section has 8 pairs
of meters, with a left
and right meter for
Figure 3-7: 8x8 Node - Meters Page
Version 2.5, May, 2009
System Parameters
IP Settings
These are the usual IP-related settings (see Introduction to Livewire; System Design Reference & Primer
for an overview and some good references to additional
information). Your network administrator should be able
to provide the needed values. Each unit must have a
unique IP address.
Host name
The name is a 12-character, alphanumeric name for
this Node that may include
hyphens but NOT spaces;
those will be converted to
hyphens. This name is used
to identify the node on the
network. You may wish to include the location of the node
(studio or rack) in the name
for ease of reference.
NOTE: If you change the IP address you will lose
your browser connection when you click Apply,
and will need to reconnect using the new IP address.
Netmask (Subnet mask)
This is the IP subnet mask of the local unit. The typical setting that is suitable for most cases is 255.255.255.0 .
Gateway (Router)
This may be the IP address of the IP Router connecting the local IP network with some other IP network.
This is not used or required in most cases.
Syslog Server (IP address)
Various services generate syslog (RFC 3164) messages, which can be forwarded to a remote syslog daemon. The remote syslog daemon IP address can be entered on the System WEB page.
Network address (IP Address)
The IP address of the
node. Each Audio Node must
have a unique IP address.
The only exception is when
two nodes are connected in
the point-to-point (snake)
configuration. Normally this
would be set using the front
panel or using the BootP program, but it can be checked
or changed from this web
page, if needed.
Figure 3-8: 8x8 Node - System Page
Version 2.5 May, 2009
3: Advanced Programming • 15
The System Parameters page, shown in Figure 3-8,
allows configuring the node’s IP address and related settings. It also permits choosing between a primary and
secondary bank of software and to download new software into the secondary bank. The currently running
software version is displayed here as well. You must
click the Apply button for changes to take place.
Syslog severity level filter
You can customize syslog logging by choosing log
detail level:
• Emergency: system is unusable
• Alert: action must be taken immediately
• Critical: critical condition
• Error: error conditions
• Warning: warning conditions
• Notice: normal but significant condition
• Informational: informational messages
• Debug: debug-level message
• Only messages with a severity higher than that specified by the filter will be forwarded to the remote
logger.
User password
This is the password required to connect to the unit.
It must be at least 5 characters long and may be as long
as 8 characters. Only alphanumeric characters are permitted. To change the password you must enter the new
and old passwords and then click Apply. NOTE: If you
changed the IP or Firmware settings the unit will reboot.
If you have only entered a new password the unit will
not reboot.
IMPORTANT! Changing device passwords can
have serious implications on the operation of
you Pathfinder software. Consult the Pathfinder
manual before making changes to your pass-
3: Advanced Programming • 16
word scheme.
When logging into the node any of the following
“user names” may be used: user, USER, axia, Axia, AXIA.
The default password is blank for any of the above users.
IMPORTANT! If the unit was upgraded from an
earlier version, only the user name “user” will
be active unless the Restore Defaults process
described in Section 2 has been performed.
Firmware version
An Axia node has two internal memory “banks”.
Each bank contains room for a complete version of operating software. This approach allows a software update
to be completed and checked without danger of making
the unit inoperable if the download were to be incomplete or corrupted. It also provides and easy way to try a
new software version and still return to the old version.
The software version in each bank is displayed here.
To change banks simply click in the “radio button” for
the desired bank and then click on Apply.
IMPORTANT! The node will reboot after you click
Apply if you change the software version. This
will result in loss of audio locally, and at any unit
using the local sources of this node.
Saving Bank 1 Software
Software is always downloaded to bank 1 (the secondary bank). Downloading new software to your node
(see below) will overwrite any software currently in this
bank. If you wish to save the software currently in bank
1, you can save it by moving it to bank 0 as follows:
•
•
Click on Commit this version to Bank 0 box (see
Figure 3-8).
Click on Apply.
Downloading new software
A new version of software can be downloaded into
bank 1 as follows:
1. Go to the Axia web site www.axiaaudio.com/downloads/ and download the desired software update
for your node to your computer (this should be the
computer that you will use to access the node’s web
page). Your local computer operating system should
display a prompt to permit you to choose where you
wish to locate the downloaded file. You can choose
any convenient location, just be sure to note the drive
and location where the file is to be saved.
2. Open a web browser and connect to the node to be
updated. Enter the complete path and file name for
the software file (e.g. the file downloaded from the
Axia site), or click on the Browse button to locate the
file. Once the proper path and filename are displayed,
click on Apply to download the file.
3. A successful download will be indicated by the new
version being displayed in the Bank 1 field. If the
download is unsuccessful the field for Bank 1 will
be blank.
4. To run the new software click on Bank 1 radio but-
Version 2.5, May, 2009
IMPORTANT! The node will reboot after you click
Apply when changing between software versions. This will result in loss of audio locally, and
at any unit using the local sources.
•
•
•
•
•
QOS & Network
This screen is slightly different for the Analog and
AES nodes. The Analog Line node QoS page is shown
in Figure 3-9.
The settings on this screen are advanced settings, and
generally the default settings should be used.
Livewire Clock Master
Livewire’s clocking system is automatic and largely
transparent to end users. By default, the Axia hardware
node with the lowest Ethernet IP address will be the
clock “master”. The system will automatically and transparently switch
to a new unit as clock master if needed. We do however, permit you to
force clock mastership to a particular
node or set certain nodes as “preferred” for clock mastership while
maintaining automatic operation.
For example you may prefer to have
nodes that are on UPS power be preferred clock masters. Note that in the
automatic modes clock mastership is
changed only when the current master becomes unavailable (adding a
new node will not change clock mastership regardless of the new node’s
setting). The only exception is the 7
(Always Master) setting.
You have the following choices
for this setting:
•
0 (always slave) “STL” – Unit
will never be master and is only
used with Ethernet radios.
•
0 (always slave) – This unit will never be used as
clock master.
3 (default) – The usual setting.
4 (Secondary Master) – Nodes with this setting will
be used as clock masters before those set to 3.
5 (Primary Master) – Nodes set to this setting will
be used as clock masters before those set to 4.
7 (Always Master) – This forces a particular node
to be clock master, even if another node is currently
clock master. If this mode becomes available then
the usual prioritization is used.
7 (Always Master) “STL Snake” – This forces a
particular node to be clock master. Use only when
two nodes are connected back to back without an
Ethernet Switch.
IMPORTANT! Only a single node on a Livewire
network should ever be set to 7 (Always Master).
For this reason we do not recommend using that
selection.
Figure 3-9: 8x8 Analog Node - QoS Page
Version 2.5 May, 2009
3: Advanced Programming • 17
ton and then click on Apply to reboot the node. It
will take approximately 20-30 seconds for the node
to reboot.
Livewire Clock Mode
Provided for compatibility with older revisions:
• IP low rate (default) – recommended setting
• Ethernet – compatible with 1.x firmware
• IP High rate – compatible with 2.1.x master
Receive Buffer Size
Determines the amount of buffering in the receiver.
Buffering is needed to compensate for jitter in network
packet delivery. Usually the biggest source of the jitter
is the source PC. Real-time performance varies widely
from one system to another; some computers can provide very low timing irregularities and allow the receive
buffer to be reduced to achieve lower audio delay. Default setting is 100 ms.
801.1p tagging, 802.1p VLAN ID, 802.1q Priority, &
DSCP Class of Service
802.1p tagging is necessary within the Livewire network to mark high-priority audio packets . This information is used by the Ethernet switches in the packet
scheduling and queuing mechanism. It provides low-jitter packet forwarding of Livewire clock and low-latency
audio streams.
3: Advanced Programming • 18
On the other hand, Standard streams don’t need tagging, because they are not low-latency. By default, standard streams are marked with Type of Service (DSCP
code points) information in the IP header which can be
used by L3 switches to provide better service to our audio streams than to best effort IP traffic.
is always 0 and cannot be changed. As a result Livewire
audio always uses the native VLAN assigned to the port
of the switch.
“DSCP Class of Service” is a standard describing the
tagging of IP frames with service information. Network
equipment can be set up to provide different forwarding delay and drop precedence depending on the service
information. Our defaults are compatible with most Ethernet equipment defaults for class of service Livewire
requires; you should not need to change them unless instructed by Axia Support.
AES Synchronization and Clock
These settings (shown in Figure 3-10) determine
two factors. The Livewire Clock Master Priority setting
determines the clock mastership options as described
above for the Analog Node. The AES node also permits
additional synchronization options to lock the node to an
AES source, as discussed below.
AES Sync Source & AES Master Timebase
If the AES sync input as Livewire master timebase option is set to YES, then this node will use the
selected AES sync source as the clocking source for this
node if and when it becomes the Livewire master clock
source. If this node becomes the Livewire clock master (see above), then the entire Livewire network will be
synchronized to the AES signal fed to the selected input.
In this case, the AES source must be a 48k clock source.
The Livewire network cannot be clocked on any other
sample rate.
There is an option to enable L2 802.1p tagging on
standard streams, and this may be used with switches
which do not use the DSCP information included in the
TOS field of the IP header. We do not enable this tagging
by default, because it wouldn’t work in cross-over Ethernet connection to PCs; most network cards do not accept
802.1p frames by default.
Of course Livewire clock mastership can change
(as described above) so the careful user will feed house
AES sync to multiple 8x8 AES nodes. In this case, those
nodes would all be set to priority 4 or 5 (see above) to ensure that these nodes will be the source for the Livewire
master clock whenever one of this group is available.
You should not need to change these default settings
unless you are building a system which is not based on
our recommendations.
If AES sync source is set to Livewire 48 kHz then the
system will simply use the unit’s internal clock source if
and when it becomes clock master.
In Axia nodes, the VLAN ID setting is read-only. It
Version 2.5, May, 2009
AES Output Sync
This sets the output sample rate and synchronisation
for this node’s AES outputs. It as two options:
• Livewire 48kHz.
This would cause the AES outputs be synced directly to the Livewire system clock and no sample rate
conversion will be performed. However the receiving unit would need to have sample rate conversion
or to be synchronous to the Livewire system clock
or dropouts due to buffer over/under run will occur.
• AES Sync in
In this case, the Livewire stream will be sample rate
converted to the clock stripped of the designated
AES input. This permits operation at rates other than
48 kHz, but only if an external source at that rate is
used.
Version 2.5 May, 2009
3: Advanced Programming • 19
Figure 3-10: 8x8 AES Node - QoS Page
World, now digital
Analog memories fade.
3: Advanced Programming • 20
The future beckons!
Version 2.5, May, 2009
Unbalanced Connections
We’ve told you, both earlier in this manual, and in
Introduction to Livewire; System Design Reference
& Primer, that Axia recommends balanced audio connections when connecting analog source and destination gear to the inputs and outputs, respectively, of Axia
nodes. Not only do we recommend this for the usual
reasons, but because inter-channel crosstalk between the
left and right channels of unbalanced signals sharing the
same Cat. 5 cable is a possibility. As we’ve mentioned
before, we recommend converting between balanced
and unbalanced at the unbalanced device and then using
the standard Cat. 5 connection from there to the Axia
node.
There are a number of active balanced-to-unbalanced
and unbalanced-to-balanced adaptors commercially
available at a reasonable cost (see www.studiohub.com
for a pair of units that will easily plug and play with our
gear). We’ll suggest one more time that this approach
is the way to go, and that using unbalanced cable runs
will compromise the performance of your state of the
art Axia audio network. However, if you are in a bind, or
otherwise determined to do so, here is how we recommend connecting Axia nodes to unbalanced equipment:
Unbalanced Destinations
To feed audio to an unbalanced destination from the
8x8 Analog node you must use a separate cable for the
left and right signals, and you will need a shielded RJ-45
plug so you can terminate the shield of the audio cables.
RJ-45 Pin 1 will feed the Left signal with the signal common (e.g. cable shield) connected to the RJ-45 shield.
Pin 3 will feed the Right signal with the signal common
(e.g. cable shield) connected to the RJ-45 shield.
An external pad may be required if the destination
equipment’s inputs cannot accept signals with peak levels of +24 dBu.
the shield. Doing so will not harm the node, however doing so will activate a feature that will increase the output
level by 6 dB, which is generally not desirable.
+
R
+
L
8
1
Feeding unbalanced device inputs from Axia 8x8
analog node outputs.
Unbalanced Sources
To feed an unbalanced signal from a source into the
inputs of the analog 8x8 node you must use a separate
cable for the left and right signals. We generally prefer
the method where the unbalanced signal is presented
across the differential balanced inputs of the node. The
handling of the shield will depend on the equipment and
grounding practices used.
If both pieces of equipment are grounded to a facility
grounding system then the shield may be left open at one
end (or both ends), as follows.
+
+
8
1
Axia node’s analog inputs fed from an unbalanced
source where both pieces of equipment are tied to a
facility ground.
Generally the unused output pin should not be tied to
Version 2.5 May, 2009
R
L
Appendices • 21
Appendix A:
Alternatively, if both pieces of equipment are not
both tied to a common facility ground, both sides of the
shield must be connected. In this case the “-“ side of the
nodes inputs are tied to the shield of the RJ-45 plug as
follows:
+
R
+
L
8
1
Appendices • 22
Axia node’s inputs fed from a floating source, with no
facility ground in common with the Axia node.
Version 2.5, May, 2009
Axia Nodes and Ethernet Radios
This tech note applies to Node Software v2.3.2a and
higher
There are several changes and additions to Axia Audio Nodes software beginning with v2.3.2a designed to
simplify the operation of STLs and audio snakes using
Axia nodes in conjunction with Ethernet Radios.
These additions include:
•
STL Slave and STL Snake modes on Clock
Master Priority options
•
IP Low Rate is now set as the default receive
Clock mode in the LW Clock Mode options
field. (Note that this setting only defines the RECEIVE type of stream. It does not change the
clock stream type when the node is acting as the
Master Clock. We recommend that you do not
change this setting.)
•
A Standard Stream Buffering option, which is
set to 100ms as default. (Note that this setting
should not be adjusted unless advised otherwise
by Axia Technical Support.)
•
A “Master/Sync” confidence tally is added to the
Router Selector Node display.
Note that the setup options described below require
that Node Software v2.3.2a or higher must be installed
to work correctly.
Using two nodes back to back without an Ethernet
switch
In this scenario, the clock sync mode will set the
Clock Rate to a Low Rate sync packet regardless of the
Livewire Clock Mode setting. This enables a more stable SYNC mode, eliminating the need for an Ethernet
switch between the nodes handling QOS of the clock
sync signal.
Navigate to the “QOS” web pages of the Audio
Nodes you’ll be using. Determine which one will be the
master and which the slave, and set the new “STL Snake”
and “STL Slave” clock priority modes to the appropriate
values.
Typically, you will set the Clock Master Priority option on the Node located in the studio to “7 (Always
Master) STL Snake”. The Node on the remote end of the
link should be set to “0 - (Always Slave) STL”.
All stream types must be set to Standard streams.
Leave Standard Stream buffering at 100ms (the default
setting).
Connecting a “remote” Audio Node to an existing
Axia network using Ethernet Radio
If you are using Studio Engines and/or existing nodes
connected to an Ethernet switch, then these instructions
assume that you have a current Livewire Network and
are adding a node at a remote location connected via an
IP radio. You must maintain the high rate Master clock
sync packets for these devices in order for all nodes to
sync properly. This is especially important for the wellbeing of the Axia Studio Mix Engines.
In this case, you will need to have at least ONE Audio Node on the main Livewire network designated as
the MASTER CLOCK and running version 2.3.2a software. It should be set to a higher priority than all other
nodes running earlier software versions. We recommend
choosing “7 - (Always Master)”. Do not select “7 – (Always Master) STL Snake” for this application.
The remote node at the receive end of the Ethernet
radio should likewise be running v2.3.2a or newer software. Its clock setting should be “0 - (Always Slave)
STL”.
Deep Tech: A node running version 2.2.0, when op-
Version 2.5 May, 2009
Appendices • 23
Appendix B:
erating as the current Clock Master, will generate two
clock streams: a High rate and a Low rate clock sync.
Nodes running version 2.1.x and earlier do not have this
dual clock feature and require the High rate sync to operate as well.
Streams sent to the “remote” node should all be
STANDARD streams. Leave Standard Stream buffering
at 100ms (the default setting) on the receiving node.
IP Radio Settings and Recommendations
Settings on your Ethernet radios will have to be
tweaked as needed. Unfortunately, due to the large number of Ethernet radios on the market, at the rate at which
these products change, we are unable to make specific
recommendations on which radio to choose, or their exact optimal settings.
Some Quality of Service options may assist or hinder the operation of the radio for multicast UDP data
packets. This may involve turning ON or OFF some or
all the “smarts” within the radio. User experience will
differ from model to model. We suggest that you contact
your radio’s manufacturer for additional support on the
operation of the radios in this mode. For the purposes of
passing Livewire streams reliably, we desire that the IP
radios behave as much as possible like a simple piece of
CAT6 cable, with minimal latency.
Appendices • 24
Questions on the operation of the Axia Audio Nodes
can be emailed to Axia support at [email protected]
com.
Version 2.5, May, 2009
Appendix C:
Troubleshooting
Here are some basic troubleshooting tips that might prove useful. Don’t forget that the Introduction to Livewire;
System Design Reference & Primer should be your companion and has many useful tips. Our on-line forum also
contains tips from users as well as our own Tech Tips section fount at http://forums.axiaaudio.com
Problem
Possible Solution
SYNC light on front panel is This indicates that the node is not able to lock onto a clock source. This is because
blinking.
there is no master clock or a network problem (not properly configured network
switch)
• check Ethernet switch configuration
• verify that there is a node assigned a “clock priority” value greater than 0 on
the network
No Meters are displayed on This page require Java be installed on the PC being used to display the node’s web
the “meters” http page.
pages. Java is a free download from www.java.com.
Need to reset node to factory 1. Power the node OFF.
defaults.
2. Depress and hold the <SELECT> and <ID> buttons.
3. Power ON the unit while continuing to hold the above buttons.
4. After about 8 seconds will see the word “RESET 3 S” displayed. If you release the buttons within 3 seconds no changes will occur. If you continue to
hold the buttons in 3 seconds the default settings will be set and “REBOOT”
will be displayed. At this time release the <SELECT> and <ID> buttons
Version 2.5 May, 2009
Appendices • 25
Audio from a node sounds • This indicates a problem with the source packet. This can be due to two debad. The meters on the web
vices producing data on the same multicast channel (source channel number)
page show audio. The meters
or also could be due to network problems. If the data is passing through a
on the front display are not
trunk cable that is heavily used (a lot of data, over 50%) data could be getting
present.
dropped. Note that the meters on the front display will show activity if the data
is valid. The meters on the web page may show activity, but this does not show
valid data is actually being received.
• Sometimes, on a small network, you may get output from a node but it has
drop-outs and is just not “quite right”. Check the clock (master/sync) to make
sure your node is getting or generating Livewire clock. See item on sync above.
Appendices • 26
Version 2.5, May, 2009
Appendix D:
Specifications and Warranty
Axia System 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
Nominal Output Level: +4 dBu
Maximum Output Level: +24 dBu
•
•
•
•
•
•
•
•
•
•
•
Reference Level: +4 dBu (-20 dB FSD)
Impedance: 110 Ohm, 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
• Any input to any output: +0.5 / -0.5 dB, 20 Hz to 20 kHz
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
Version 2.5 May, 2009
Specifications & Warranty • 27
Digital Audio Inputs and Outputs
• 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
Crosstalk Isolation and Stereo Separation and CMRR
•
•
•
•
•
Analog Line channel to channel isolation: 90 dB isolation minimum, 20 Hz to 20 kHz
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
Power Supply AC Input
• Auto-sensing supply, 90VAC to 240VAC, 50 Hz to 60 Hz, IEC receptacle, internal fuse
• Power consumption: 35 Watts
Operating Temperatures
• -10 degree C to +40 degree C, <90% humidity, no condensation
Dimensions and Weight
Microphone node: 1.75 inches x 17 inches x 10 inches, 6 pounds
Analog Line node: 1.75 inches x 17 inches x 10 inches, 6 pounds
AES/EBU node: 1.75 inches x 17 inches x 10 inches, 6 pounds
Router Selector node: 1.75 inches x 17 inches x 10 inches, 6 pounds
GPIO node: 1.75 inches x 17 inches x 13 inches, 8 pounds
Studio Mix Engine 3.5 inches x 17 inches x 15 inches, 10 pounds
Specifications & Warranty • 28
•
•
•
•
•
•
Version 2.5, May, 2009
Axia Node 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 TLS Corp., d/b/a Axia Audio (hereinafter “Axia Audio”).
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 by the end-user. Software-only items are warranted
to be free from defects in material and workmanship for a period of 90 days from the date of receipt by the end-user.
This warranty is void if the Product is subject to Acts of God, including (without limitation) lightning; improper
installation or misuse, including (without limitation) the failure to use telephone and power line surge protection devices; accident; neglect or damage.
EXCEPT FOR THE ABOVE-STATED WARRANTY, AXIA AUDIO MAKES NO WARRANTIES, EXPRESS
OR IMPLIED (INCLUDING IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE).
In no event will Axia Audio, its employees, agents or authorized dealers 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.
In order to invoke this Warranty, notice of a warranty claim must be received by Axia Audio within the above-stated
warranty period and warranty coverage must be authorized by Axia Audio. If Axia Audio authorizes the performance
of warranty service, the defective Product must be delivered, shipping prepaid, to: Axia Audio, 2101 Superior Avenue,
Cleveland, Ohio 44114.
Axia Audio at its option will either repair or replace the Product and such action shall be the full extent of Axia
Audio’s obligation under this Warranty. After the Product is repaired or replaced, Axia Audio will return it to the party
that sent the Product and Axia Audio will pay for the cost of shipping.
Axia Audio’s products are to be used with registered protective interface devices which satisfy regulatory requirements in their country of use.
rev 12/08/04 v 1.0 RKT
rev 12/28/04 v 1.0b RKT
rev 01-07-05 v1.0c RKT
rev 05/2009 v 2.5 BW/CN
Part # 1490-00038-001
Version 2.5 May, 2009
Specifications & Warranty • 29
Axia Audio’s authorized dealers are not authorized to assume for Axia Audio any additional obligations or liabilities in connection with the dealers’ sale of the Products.
Axia Audio, a Telos Company • 2101 Superior Ave. • Cleveland, Ohio, 44114, USA • +1.216.241.7225 • www.AxiaAudio.com
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