A

REPORT

ON

PUBLIC ADDRESS SYSTEM

BY

AKSHAY GAWANDE

Telecommunication Engineer
Dar Al-Handasah Consultants, Pune

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 CONTENTS
1) INTRODUCTION TO PUBLIC ADDRESS SYSTEM ……………………...…….3

2) TPYE OF PROTOCALS………………………………….………………………….4

I. AUDIO VIDEO BRIDGING (AVB) ………………………………………4

II. COBRANET……………………………………………………………….12
III. DANTE…………………………………………………………………….22

3. TPYE OF METHOD…………………………………………………………………...26

4. COMPARISON…………………………………………………………………………28

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 1. INTRODUCTION OF PUBLIC ADDRESS
SYSTEM
A public address system (PA system) is an electronic system comprising microphones,
amplifiers, loudspeakers and related equipment. It increases the apparent volume (loudness) of a
human voice, musical instrument, or other acoustic sound source or recorded sound or music. PA
systems are used in any public venue that requires that an announcer, performer etc. be sufficiently
audible at a distance or over a large area. Typical applications include sports stadiums, public
transportation vehicles and facilities, and live or recorded music venues and events. A PA system
may include multiple microphones or other sound sources, a mixing console to combine and
modify multiple sources, and multiple amplifiers and loudspeakers for louder volume or wider
distribution.

Simple PA systems are often used in small venues such as school auditoriums, churches,
and small bars. PA systems with many speakers are widely used to make announcements in public,
institutional and commercial buildings and locations—such as schools, stadiums, and passenger
vessels and aircraft. Intercom systems, installed in many buildings, have both speakers throughout
a building, and microphones in many rooms so occupants can respond to announcements. The
term, sound reinforcement system generally means a PA system specifically for live music or
performance.

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 3. TYPE TO PROTOCALS

I. AUDIO VIDEO BRIDGING (AVB)

Audio Video Bridging (AVB) is a common name for the set of technical standards developed by
the Institute of Electrical and Electronics Engineers (IEEE) Audio Video Bridging Task Group of
the IEEE 802.1 standards committee. This task group was renamed to Time-Sensitive Networking
Task Group in November 2012 to reflect the expanded scope of work. The charter of this
organization is to "provide the specifications that will allow time-synchronized low latency
streaming services through IEEE 802 networks". These consist of:

 IEEE 802.1BA: Audio Video Bridging (AVB) Systems;

 IEEE 802.1AS: Timing and Synchronization for Time-Sensitive Applications (gPTP);
 IEEE 802.1Qat: Stream Reservation Protocol (SRP); and
 IEEE 802.1Qav: Forwarding and Queuing for Time-Sensitive Streams (FQTSS).

IEEE 802.1Qat and 802.1Qav are amendments to the base IEEE 802.1Q document, which specifies
the operation of "Media Access Control (MAC) Bridges and Virtual Bridged Local Area
Networks", which are implemented by network devices typically called Ethernet switches.

To help ensure interoperability between devices that implement the AVB standards, the AVnu
Alliance develops device certification for the automotive, consumer, and professional audio and
video markets.

Audio Video Bridging, AVB, is a method for transport of audio and video streams over
Ethernet-based networks. It is based on ratified IEEE standards for Ethernet networks that define
signaling, transport and synchronization of the audio and video streams.

A critical component in a converged, universal network that is shared by legacy data traffic
and AV streams is Quality of Service (QoS) because it allows voice and video traffic to be
prioritized over data traffic ensuring timely and reliable delivery. AVB takes this concept to a new
level by adding stream signaling, automatic bandwidth reservation and traffic prioritization, as
well as time synchronization thus allowing high quality digital audio and video streams to be
reliably transported over Ethernet.
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 The standard QoS needs to be manually implemented on every device and interface in the path
between the sender and receiver. This includes identifying traffic, tagging it with QoS priority
values, and then configuring queuing and/or shaping policies that are applied per interface. This
approach labor-intensive and static, i.e. it does not allow for dynamic, on-demand updates... AVB,
on the other hand, has a fully automated end-to-end QoS. As the AV streams are setup all devices
along the path automatically and on-demand implement queuing, shaping, and prioritization for
each AV stream. When the stream stops, the policies are automatically removed and bandwidth
freed up. At the same time, AVB QoS mechanism limits the total amount of interface bandwidth
that can be reserved for AV streams, so that other applications get their fair share of the available
bandwidth. Implementing AVB puts demanding requirements on network switches. Arista
Ethernet switches provide extensible and programmable OS, low latency, wire speed throughput,
and port speeds of 100Mb and 1/10/40/100Gb, providing the ideal platform for demanding
applications leveraging a shared network infrastructure.

AVB Advantages

Audio Video Bridging, AVB, allows businesses to take advantage of a shared Ethernet
network infrastructure. This can imply share cost model and savings in addition to unification and
simplification of the overall network infrastructure. Ethernet is the de-facto standard used
everywhere today, so that businesses can take advantage of networks and cabling that is already
in place. Being so widely deployed, Ethernet is a well-known, mature, and well-supported
technology. The cost per port and bandwidth keeps going down for Ethernet switches, while port
speed, transport latency, and overall switch throughput keep improving.
Ethernet is inherently very scalable and flexible and also the standard used in the largest
data centers today. It is an ideal technology for all sizes AVB deployments. All kinds of
communication streams are supported by default: one-to-one, one-to-many, and many to- one.
With just one connection to a single Ethernet switch, an AVB end point can setup any of the above
communications with any of the other AVB end points connected to the same AVB domain. An
AVB domain can be a single switch or many Ethernet switches connected together allowing
thousands of AVB end points to communicate. Expanding an AVB domain can be as simple as
connecting additional Ethernet switches with zero impact to the existing AVB end points and
network connectivity.

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 To ensure interoperability between different vendors, AVB is based on a set of ratified
IEEE standards. In addition, AVnu alliance is an industry alliance that has been formed in order
to promote AVB and ensure compatibility. AVnu Alliance is also responsible for the certifications
of AVB devices, which are conducted at the University of New Hampshire Interoperability Lab.
Delay-sensitive digital AV traffic requires a low latency, low jitter network. Per the
standards, latency end-to-end for a Class-A AV stream must be within 2 ms. Arista Ethernet
switches belong to the category of ultra-low latency, best in breed, high-end data center products.
Time synchronization between all devices within an AVB domain allows for deterministic
delivery of AVB streams. One of the important concerns that is addressed by the Generalized
Precision Time Protocol (gPT) is “lip synching”, i.e. synchronization between audio and video
streams. A gPTP master clock becomes a reference clock for all devices in the AVB domain - this
allows the AVB end points to present the AV frames at the receiving end at the correct time.
Because of built-in high precision IEEE 1588 PTP clocks, Arista switches can act as gPTP masters
or serve as backup for external gPTP master in cases where the primary gPTP master becomes
unavailable.
AVB was designed to be “plug and play” allowing for rapid deployment of an AVB
network. End points and AVB- capable switches automatically and dynamically implement stream
signaling, quality of service, time synchronization, and stream delivery. A third party central AVB
controller can be used for automatic discovery and enumeration of end points, and for stream setup
and tear down.

How Audio Video Bridging Works

Core components of AVB are Time Synchronization, Signaling, Transport Protocol, AVB
Systems, Quality of Service, and Control Protocol. This white paper will focus on the operation of
Time Synchronization, Signaling, and Transport Protocol in order to illustrate the basic AVB
operation.

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Time Synchronization

gPTP ensures that all devices in the AVB domain have synchronized clocks to ensure consistent
delivery of time- sensitive AV streams. Having synchronized clocks helps prevent “out of sync”
issues, such as “lip sync” problem where audio and video are not synchronized.
• If an external gPTP device is not available, the Arista switch can act as the gPTP master for the
domain
• Switch interfaces towards the gPTP Grand Master are called master interfaces, and interfaces
toward the gPTP devices that are not Grand Masters are called slave interfaces
• gPTP device priority can be manually set to predictably decide primary and backup gPTP Grand
Masters
• Initial and all subsequent gPTP Grand Master elections are completed automatically
• Switch’s interface must run gPTP in order to be declared capable of forwarding AVB traffic

Signaling

AVB operation is efficient because only stream signaling is send to all AVB interfaces
within the AVB domain. The AV stream itself doesn’t start until the negotiation between the Talker
and Listener is completed. The AV stream is then transported only between those two end points.
1. AVB signaling and stream setup are accomplished using Multiple Stream Reservation Protocol
(MSRP). To allow all other end points to automatically learn about available streams, the Talker
sends advertisements of all locally available streams, each with a unique stream ID. These
advertisements include data rate required, class, stream priority, and other parameters

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2. The switch forwards the Talker advertisements out all AVB capable interfaces that have enough
available bandwidth to accommodate these streams
3. The Talker advertisements are received by the Listeners

1. The Listener needs a way of communicating to the Talker what streams it’s interested in
receiving. It uses the same signaling protocol as the Talker - Multiple Stream Reservation Protocol
(MSRP). After learning about the available streams, the Listener sends Listener advertisements for
specific streams it wants to receive.
2. The switches will forward the Listener advertisements back to the Talker on the interface where
the Talker advertise was originally seen, and also reserve bandwidth along the path for the stream
specified
• The switch connected to the Listener will also map the stream’s unique multicast MAC address
to Listener’s interface
• The Listener also sends “Listener Ready” advertisement back to the Talker

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3. After learning about available streams, the Listener sends its own advertisements for the specific
streams it wants to receive
4. The switches will forward the Listener advertisements back to the Talker on the interface where
the Talker advertise was originally seen, and also reserve bandwidth along the path for the stream
specified
• The switch connected to the Listener will also map the stream’s unique multicast MAC address
to Listener’s interface
• The Listener also sends “Listener Ready” advertisement back to the Talker

Transport
Stream Transport Protocol converts AV frames in different format, such as Fire wire/IEEE 1394
to Ethernet frames that can then be transported over Ethernet networks.
1The Talker starts streaming the requested stream using the stream transport protocol.
• The Talker and Listener will also send periodic advertisements during stream transmission to
ensure the endpoints are still reachable.
2. The switches in the AVB domain will forward the Talker’s stream only toward the Listener’s
port.
• Multiple Listeners can subscribe to the same Talker stream
• A single Listener can also subscribe to multiple streams

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• Smart AVB switches along the stream’s path automatically reserve bandwidth and prioritize
AVB traffic ensuring reliable and predictable end-to-end transport of the streams

AVB and Arista

AVB is supported on Arista 7150 and 7500E switches. Arista switches offer low latency and wire
speed performance, making the switches ideal for delay-sensitive applications such as digital audio
and video. Arista Extensible Operating System (EOS) software is an extensible and programmable
switch OS. JSON-based eAPI, SDK, and Python Scripting are available to allow for advanced
integration and network visibility, automation, and for building tools for operations teams. By
leveraging Arista’s EOS extensibility and programmability, operations teams with minimum
network experience can monitor and troubleshoot networks and AVB. A single Arista switch can
provide between 24 - 1152 AVB interfaces at speeds of 100Mb and 1/10/40/100GB per interface.
Arista is a member of the AVnu Alliance http://www.avnu.org/about_us/our_members, which
promotes AVB standards, the adoption of the standards, and interoperability testing

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Summary

The trend is to deploy a universal network for all applications including Voice over IP, Video over
IP, and Audio Video Bridging (AVB) in order to take advantage of a shared network infrastructure
and shared cost. The flexibility and scalability of Ethernet allow for small and large AVB
deployments and seamless expansion without down time. The combination of low latency, high
throughput Ethernet switches, and fully automated end-to-end Quality of Service ensures timely
and reliable delivery of delay-sensitive AVB streams. Generalized Precision Time Protocol (gPTP)
ensures that all devices within the AVB domain are time-synchronized. A 3rd party central AVB
controller can be used to discover and manage the AVB end points, while also allowing the
operations teams to setup and tear down AVB streams within the domain. AVB is standards-based
ensuring interoperability between different vendors.
Arista offers both fixed configuration and modular switches that support AVB. The Arista AVB
Ethernet switches provide low latency, wire speed performance, and also have built-in, high
precision PTP clocks. Arista’s Extensible Operating System (EOS) provides tools for automation,
visibility, and integration with external controllers and monitoring tools. Arista’s AVB switches
still have all of the capabilities and features of non-AVB Ethernet switches, allowing other
applications to coexist with AVB running on the switches.

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 II.COBRANET
CobraNet is a combination of software, hardware, and network protocols designed to
deliver uncompressed, multi-channel, low-latency digital audio over a standard Ethernet network.
Developed in the 1990s, CobraNet is widely regarded as the first commercially successful
implementation of audio over Ethernet. CobraNet was designed for and is primarily used in large
commercial audio installations such as convention centers, stadiums, airports, theme parks, and
concert halls. It has applications where a large number of audio channels must be transmitted over
long distances or to multiple locations.

CobraNet is an alternative to analog audio, which suffers from signal degradation over long
cable runs due to electromagnetic interference, high-frequency attenuation, and voltage drop.
Additionally, the use of digital multiplexing allows audio to be transmitted using less cabling than
analog audio.

CobraNet was developed in 1996 by Boulder, Colorado-based Peak Audio. Initial

demonstrations were of a 10 Mbit/s point-to-point system with limited channel capacity. The first
permanent installation of CobraNet in this early form was to provide background music throughout
Disney's Animal Kingdom theme park.The first commercial use of CobraNet as an interoperable
standard was during the half-time show at Super Bowl XXXI in 1997.

CobraNet was first introduced as an interoperable standard in collaboration with manufacturer

QSC Audio Products. QSC was the first to license the technology from Peak Audio and marketed
it under the RAVE brand. At this point CobraNet had graduated to fast Ethernet and used a unique
collision avoidance technique to carry up to 64 channels per Ethernet collision domain.

CobraNet was subsequently enhanced to support and eventually require a switched Ethernet
network. An SNMP agent was added for remote control and monitoring. Support for higher sample
rates, increased bit resolutions and lowered latency capabilities were later introduced in an
incremental and backwards-compatible manner. In May 2001, Cirrus Logic announced that it had
acquired the assets of Peak Audio. Leveraging Cirrus DSP technology, a low-cost SoC
implementation of CobraNet was developed and marketed.

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Advantages

Using CobraNet and fast Ethernet, 64 channels of uncompressed digital audio are carried
through a single category 5 cable. Using gigabit and/or fiber optic Ethernet variants, the cost of
cabling per audio channel is reduced compared to the fast Ethernet connections. CobraNet data
can also coexist with data traffic over existing Ethernet networks, a single network infrastructure
can serve audio distribution and other networking needs. Audio routing can be changed at any time
with network commands, and do not require rewiring. Use of Ethernet by CobraNet offers many
high availability features such as Spanning Tree Protocol, link aggregation, and network
management. For critical applications, CobraNet devices can be wired with redundant connections
to the network. In this configuration, if one CobraNet device, cable, or Ethernet switch fails, the
other takes over almost immediately. Audio is transmitted in digital form, and provides reduced
susceptibility to electromagnetic interference, crosstalk, coloration, and attenuation owing to cable
impedance.

Disadvantages

Delays over the CobraNet transmission medium itself are at least 1 1⁄3 milliseconds per network
traversal. For some applications, these delays can be unacceptable – especially when combined
with further delays resulting from propagation time, digital signal processing and the conversions
between analog and digital. Also, licensing the technology or purchasing the required CobraNet
interfaces, which encode and decode the CobraNet signal, can be expensive.

Transmission

Simple block diagram of an audio system employing CobraNet technology. Red lines indicate
analog audio signals, while green lines indicate standard Ethernet signals.

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 CobraNet is transmitted using standard Ethernet packets. Instead of using TCP/IP packets,
CobraNet transfers data using data link layer packets, which travel quickly through hubs, bridges
and switches, and are not as susceptible to the latency and QoS problems commonly found in
streaming protocols using a higher transport layer. However, since Cobra Net does not use IP
protocol, its packets cannot travel through routers, and therefore it is limited to use on a LAN;
CobraNet cannot be used over the Internet. The network over which CobraNet is transmitted must
be able to operate at a minimum of 100 Mbit/s. All CobraNet packets are identified with a unique
Ethernet protocol identifier (0x8819) assigned to Cirrus Logic.

CobraNet is not designed to work over wireless networks. Bandwidth and reliability issues
associated with typical 802.11 wireless networks tend to cause frequent dropouts and errors.
However, wireless communication of CobraNet data can be accomplished reliably using lasers.

Channels and bundles

CobraNet data is organized into channels and bundles. A typical CobraNet signal can
contain up to 4 bundles of audio travelling in each direction, for a total of 8 bundles per device.
Each bundle houses up to 8 channels of 48 kHz, 20-bit audio, for a total capacity of 64 channels.
CobraNet is somewhat scalable, in that channel capacity increases when 16-bit audio is used, and
channel capacity decreases when 24-bit audio is used. The number of channels allowed per bundle
is limited by the 1,500-byte Ethernet MTU.

There are three types of bundles: multicast, unicast, and private:

 Multicast bundles are sent from one CobraNet device to all other CobraNet devices in the
network using Ethernet multicast addressing. Each CobraNet device individually
determines if it will use the bundle or discard it. Therefore, multicast bundles are more
bandwidth-intensive than other bundle types. Bundle numbers 1–255 are reserved for
multicast bundles.
 Unicast bundles are sent from one CobraNet device to any other device or devices
configured to receive the bundle number. Unicast bundles are much more efficient because
network switches route them only to devices which actually want to receive them. Despite
their name, unicast bundles may still be sent to multiple devices, either by transmitting

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 multiple copies of the audio data or using multicast addressing. Bundle numbers 256–
65279 are reserved for unicast bundles.
 Private bundles may be sent with unicast or multicast addressing. Bundle numbers 65280–
65535 are reserved for private bundles. Private bundle numbers are paired with the MAC
address of the device that transmits them. To receive a private bundle, both the bundle
number and the MAC address of the transmitter must be specified. Because 256 private
bundles available to each transmitter, there is no limit on the total number private bundles
on a network.

As long as multicast bundles are used sparingly, it is virtually impossible to exceed the bandwidth
of a 100 Mbit network with CobraNet data. However, there are limitations to the maximum number
of bundles that can be sent on a network, since the conductor must include data in its beat packets
for every bundle on the network, and the beat packet is limited to 1,500 bytes. If each device is
transmitting one bundle, there may be up to 184 transmitters active simultaneously (for a total of
184 bundles). If each device is transmitting four bundles, then only 105 transmitters can be active,
although they would be producing a total of 421 active bundles. The use of private bundles does
not require any additional data in the beat packet, so these network limitations can be sidestepped
by using private bundles.

Synchronization

The CobraNet network is synchronized to a single CobraNet device known as the conductor. A
conductor priority can be configured to influence selection of the conductor. Among devices with
the same conductor priority, the first to establish itself on the network becomes is elected
conductor. All other devices are known as performers. In the event that the conductor fails, another
CobraNet device will be chosen to become the conductor within milliseconds. CobraNet cannot
function without a conductor.

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Packets

Four main types of packet are used in the transmission and synchronization of CobraNet

 Beat packets – the conductor outputs a beat packet to all other CobraNet devices on the
network at a rate of 750 packets per second. All other CobraNet devices on the network
synchronize their audio clock and their data transmissions to the beat packet. The beat
packet contains network operating parameters, clock data and transmission permissions for
multicast and unicast bundles.
 Audio packets – also known as isochronous data packets, these packets are sent out by all
CobraNet devices after they receive a beat packet. At standard latency settings, one audio
packet is sent for each beat packet received, and each audio packet includes 64 samples of
audio data per channel. At lower latency settings, audio packets may be sent twice or four
times for each beat packet received. Bundles do not share packets; separate packets are sent
in sequence for each bundle transmitted from the same device.
 Reservation packets – these packets are transmitted as needed or typically once per
second at minimum. Their function is to control bandwidth allocation, initiate connections
between CobraNet devices, and monitor the status of CobraNet devices.
 Serial bridge packets – asynchronous serial data may be sent between CobraNet devices
on the same network. Many standard asynchronous serial formats are supported, including
RS-232, RS-422, RS-485 and MIDI.

Latency

The buffering and transmission of audio data in Ethernet packets typically incurs a delay of
256 samples or 5 1⁄3 milliseconds. Additional delays are introduced through A-D and D-A
conversion. Latency can be reduced by sending smaller packets more often. In most cases, the
programmer can choose the desired CobraNet latency for a particular CobraNet device (5 1⁄3,
2 2⁄3, or 1 1⁄3 milliseconds). However, reducing audio latency has consequences:

 Reducing latency requires more processing by the CobraNet interface and may reduce
channel capacity.
 Reducing latency places additional demands on network performance, and may not be
possible in some network configurations if the forwarding delay is too high.

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  Since reducing latency means sending smaller packets more often, higher resolution (i.e.
96 kHz, 24-bit) audio channels can be sent per bundle without exceeding the 1,500-byte
payload limit for Ethernet packets.

Channels per bundle

Latency 16-bit, 20-bit, 24-bit, 16-bit, 20-bit, 24-bit,
48 kHz 48 kHz 48 kHz 96 kHz 96 kHz 96 kHz
 5 1⁄3 ms 8 8 7 5 4 3
 2 ⁄3 ms
2
8 8 8 8 8 7
 1 ⁄3 ms
1
8 8 8 8 8 8

It may seem from the Latency vs. Channels per bundle table that more information can be sent at
a lower latency. However, that is not the case. More channels can be sent per bundle, but fewer
bundles can be processed simultaneously by one device. So, while eight 24-bit, 96 kHz channels
can be sent in one bundle at  1 1⁄3 ms latency, due to processing constraints, the CobraNet device
may only be able to send and receive one bundle instead of the usual four. The bundle capacity of
CobraNet devices are unique to the particular device, and are not always the same. The Channels
per bundle vs. test case latencies table illustrates the bundle capacity for a Biamp AudiaFLEX-CM
DSP device. The Rx and Tx columns indicate the absolute maximum number of channels that can
be received or transmitted. The Rx/Tx column represents the maximum number of channels that
can be received and transmitted simultaneously.

 1 1⁄3 ms latency  2 2⁄3 ms latency  5 1⁄3 ms latency

Channels per bundle
Rx Tx Rx/Tx Rx Tx Rx/Tx Rx Tx Rx/Tx
8 32 32 32/32 32 32 32/32 32 32 16/16
7 32 32 32/32 32 32 29/29 28 32 14/15
6 32 32 32/32 32 32 29/29 24 32 12/13
5 32 32 32/32 32 32 25/27 21 32 12/13
4 32 32 32/32 32 32 24/24 20 28 12/12
3 32 32 32/32 32 32 20/21 15 24 9/11
2 32 32 28/29 27 32 16/16 12 18 6/7
1 16 16 16/16 16 16 9/10 7 10 4/4

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Hardware and software

CobraNet interfaces come in several varieties, some of which can support more channels
than others. Additionally, CobraNet interfaces have two Ethernet ports labelled "primary" and
"secondary". Only the primary Ethernet port needs to be connected, but if both ports are connected
the latter acts as a fail-safe. Careful network design and topology which takes advantage of this
feature can provide extremely high reliability in critical applications.

The typical CobraNet interfaces provided by Cirrus Logic are the CM-1 and the CM-2

 CM-1 – the standard CobraNet card, provides 32 in and 32 out audio channels.
 CM-2 – compact, low-power, lower cost design provides 8 or 16 audio channels.
 Both cards are designed to be added to audio products by the manufacturer.

Software

Cirrus Logic provides a software application known as CobraCAD, which assists in the design of
the network on which the CobraNet system will run. It helps to identify if there are too many
routers between two CobraNet devices, if a certain latency is possible given the network
configuration, and other tasks. However, Cirrus Logic does not provide software to manipulate
their hardware. In fact, in the simplest of cases, no software is required by the end user. For
instance, a simple breakout box which converts a CobraNet signal to eight analog audio signals
would require little or no configuration by the end user apart from possibly selecting the bundle
number. If configuration is required (for example, in a DSP box with integrated CobraNet I/O),
then the manufacturer of the device typically supplies proprietary software for that purpose

Devices

One type of device that integrates CobraNet is the audio DSP. As self-powered speakers became
more common, Cobranet was frequently used to distribute the audio signal from the DSP. These
devices typically receive audio from CobraNet (and often from other digital or analog sources
simultaneously), and process the audio using digital filters and effects (for example, volume
control, EQ, compression, delay, crossovers, etc.) and then output the audio via CobraNet (or other
digital or analog outputs). Some DSPs even have an integral telephone hybrid, and can incorporate

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CobraNet and other sources into a teleconferencing application. Amplifiers with integrated
CobraNet help keep the signal chain digital for a longer span. Amplifiers with CobraNet inputs
may also have limited DSP and network monitoring capabilities.

Loudspeakers with integrated CobraNet help keep the signal chain digital for an even longer span.
In a typical unpowered speaker application, the amplifier would be housed far away from the
speaker, and a long speaker cable (analog) would be run between the speaker and the amplifier.
The speaker cable would be subject to interference and signal loss from electrical resistance.
However, a powered speaker, powered by an electrical cable and fitted with integrated CobraNet
inputs, eliminates the speaker cable and replaces it with a network cable. Since a speaker will only
use one audio channel out of the bundle, many speakers with CobraNet will also have a number
of analog outputs for the rest of the channels in the bundle, which is useful in speaker cluster
applications. Many digital mixing consoles are available with optional CobraNet interfaces for
increased channel capacity and reduced cabling.

Manufacturers

Manufacturers who wish to integrate CobraNet connectivity into their devices either license the
technology or purchase CobraNet interface modules or chips from Cirrus Logic. Many audio
equipment manufacturers have included CobraNet in their products. Below is a partial list:

 Biamp Systems
 Bose Corporation
 dbx
 Crest Audio
 Crown International
 D&R Electronica
 Dolby Laboratories
 EAW
 Electro-Voice
 JBL
 Lab.gruppen
 Mackie

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  Midas Consoles
 Peavey MediaMatrix
 QSC Audio Products
 Rane
 Renkus-Heinz
 Soundcraft

III.DANTE

Dante (Digital Audio Network Through Ethernet) is a combination of software, hardware, and
network protocols that deliver uncompressed, multi-channel, low-latency digital audio over a
standard Ethernet network using Layer 3 IP packets. Developed in 2006 by a Sydney-based
company named Audinate, Dante builds and improves on previous audio over Ethernet
technologies, such as CobraNet and EtherSound.

Like most other audio over Ethernet technologies, Dante is primarily for professional, commercial
applications. Most often, it is used in applications where a large number of audio channels must
be transmitted over relatively long distances or to multiple locations.

Digital audio provides several advantages over traditional analog audio distribution. Audio
transmitted over analog cables can be adversely affected by signal degradation due to
electromagnetic interference, high-frequency attenuation, and voltage drop over long cable runs.
Thanks to digital multiplexing, the cabling requirements for digital audio distribution are almost
always reduced when compared to analog audio. Dante also provides specific advantages over
first-generation audio over Ethernet technologies, such as CobraNet and EtherSound.
Technological advancements include native gigabit support, higher channel count, lower latency,
and automatic configuration. Dante audio networking utilizes standard IP networks to transmit
high-quality, uncompressed audio with near-zero latency. It's the most economical, versatile, and
easy-to-use audio networking solution, and is scalable from simple installations to large-capacity

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networks running thousands of audio channels. Dante can replace multiple analog or multicore
cables with a single affordable Ethernet cable to transmit high-quality multi-channel audio safely
and reliably. With Dante software, the network can be easily expanded and reconfigured with just
a few mouse clicks. Dante is the audio networking choice of nearly all professional audio
manufacturers, with hundreds of Dante-enabled audio products now available.

About Dante Controller

Dante Controller is a software application provided by Audinate which allows users to configure
and route audio around Dante networks. It is available for Windows and OS X (see Minimum
System Requirements for more details). Once you install Dante Controller on your PC or Mac and
connect it to a Dante network, you can use Dante Controller to:
 View all Dante-enabled audio devices and their channels on the network
 View Dante-enabled device clock and network settings
 Route audio on these devices, and view the state of existing audio routes
 Lock and unlock Dante devices
 Change the labels of audio channels from numbers to names that suit you
 Customize the receive latency (latency before play out)
 Save audio routing presets
 Apply previously saved presets
 Edit presets offline, and apply as configurations for new network deployments
 View and set per-device configuration options, including:
Changing the device name
Changing sample rate and clock settings
Viewing detailed network information
Access the device web page to upgrade firmware and license information (where
supported) Identify a device for example by flashing LEDs (where supported)
 View network status information, including:
 Multicast bandwidth across the network
 Transmit and receive bandwidth for each device
 View device performance information, including latency statistics and packet errors
View clock status information for each device, including frequency offset history and clock
event logs

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 Routing Audio
Routing Terminology
Device: A device means a Dante-enabled device, and more specifically that component of the
audio equipment that implements the Dante interface. A Dante device typically has Tx and Rx
channels and other routing-related properties.
Transmit (Tx) channel: A transmit channel transmits audio from the audio hardware onto the
network.
Receive (Rx) channel: A receive channel receives audio from the network and sends it to the
audio hardware.
Flow: Dante audio routing creates flows. Each flow carries several channels of audio from a
transmitter to one or more receivers. Unicast routing creates flows to single receivers. Multicast
routing creates flows that can be received by multiple receivers. Multicast flows are assigned IDs
enabling them to be identified in Dante Controller.
Unicast routing: Unicast flows are point-to-point from a single transmitter to a single receiver.
Unicast flows typically have room for 4 channels of audio.
Multicast routing: Multicast flows are one-to-many from a single transmitter to any number of
receivers. Use Dante Controller to choose which channels are to be multicast. Unlike unicast
routing, multicast flows consume network bandwidth even if there are no receivers, but do not
require additional bandwidth to add more receivers.
Subscription: A subscription configures a receive channel to receive audio from a transmit
channel on another Dante device.
Subscription status: For a receive channel, subscription status indicates whether it is
subscribed, Whether it is receiving unicast or multicast audio, whether the subscription is OK, or
whether an error has occurred.

Subscription

Dante routing is performed by associating a receiving (Rx) channel with a transmitting

(Tx) channel. This is called subscription.
Example: Route Tx channels 1 and 2 (labeled “Audio L” and “Audio R”) on the device labeled

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“Source” to Rx channels 3 and 4 on the device labeled “Dest”.

Dante will perform the necessary audio routing to deliver the audio from the Tx channels to the
Rx channels.

Redundancy
Many Dante devices support redundant audio routing. These devices have two network
interfaces, labeled primary and secondary. Primary interfaces should be connected to one physical
network. If redundancy is being used, secondary interfaces should be connected to a second
separate network. Secondary interfaces cannot communicate with primary interfaces network.

If the secondary network is connected to a device that supports redundancy, it is enabled

automatically.
The same audio data is transmitted on both the primary and secondary networks simultaneously.
In the event of a failure on one network, audio will continue to flow via the other network.

Clock Synchronization
All Dante-enabled devices use the IEEE 1588 Precision Time Protocol (PTP) across the network
to synchronize their local clocks to a master clock, providing sample-accurate time alignment
throughout the network.
One Dante device will be elected as the PTP Master Clock for the network; all other Dante devices
act as a PTP Slave Clocks to the elected master clock. Although many Dante devices may be

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capable of becoming PTP Master Clock, only one device will win the election. Devices with clock
inputs (e.g. Word Clock or AES3) will be preferred in the election process. A gigabit connected
device is preferred over a device connected via 100Mbps. A tie-breaker rule of the lowest MAC
address is used if several equivalent candidate master clocks are available. The election process
may be overridden by manually setting 'Preferred Master' on a device.

Latency
In Dante, variation in latency in the network is compensated for at the receiver. Each
receiver has a device latency setting. This setting defines the latency between the timestamps on
the incoming audio samples and when those samples are played out.
The typical default latency for a Dante device is 1 msec. This is sufficient for a very large network,
Consisting of a Gigabit network core (with up to 10 hops between edge switches) and 100 megabit
links to Dante devices. Smaller, Gigabit-only networks can use lower values of latency (down to
150 μsec for very fast devices, such as PCIe cards). Recommended latency settings are displayed
in Dante Controller, and may also be found in the documentation accompanying the product.

Latency is set on the receiver. However, when a subscription is made, there is an automatic
negotiation process between the receiver and the transmitter, to ensure that the latency for the
subscription is high enough to prevent packet loss.
For example, Ultimo devices support a minimum of 1ms latency. If a faster device (such as a PCIe
card) is set to 0.25ms latency, and is then subscribed to an Ultimo transmitter, the latency used for
the msubscription will be 1ms, which is the minimum supported latency for the subscription.
Subscriptions to other devices (such as a Brooklyn II device) will be set at 1ms (or whatever
latency the receiver is set to). This effectively makes the device latency setting a 'default' latency,
which is used unless the transmitter doesn't support it.

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 Block diagram of Dante network

3) Comparison
AVB COBRANET DANTE

Network Network Compatibility Network Compatibility

Compatibility
AVB is Switchable Cobra net is Switchable Dante is Switchable

AVB is not Routable Cobra net is not Dante is Routable

Routable
Ethernet data rates is : Ethernet data rates is : Ethernet data rates is :
100Mb and fast Ethernet fast Ethernet, Gigabit
1/10/40/100GB Ethernet
Fast Ethernet, Gigabit
Ethernet, 10 Gigabit
Ethernet
Audio Specifications Audio Specifications Audio Specification

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 Minimum latency is Minimum latency is Minimum latency is
0.25 ms 1 1⁄3 ms 150 us

Maximum channels Maximum channels Maximum channels

per link per link is per link is 1024
64(32x32) (512x512)

Maximum sampling Maximum sampling Maximum sampling

rate rate rate is
96 kHz 192 kHz
Maximum bit depth Maximum bit depth Maximum bit depth is
24 bits 32 bits
Manufacturer Info Manufacturer Info Manufacturer Info
IEEE, AVnu Cirrus Logic Audinate Pty. Ltd.
E-POWER  Biamp Systems

CROWN BY  Bose EXTRON

HARMAN Corporation ELECTRONICS
 dbx
 Crest Audio
 Crown
International
 D&R
Electronica
 Dolby
Laboratories
 EAW

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  Electro-Voice
 JBL
 Lab.gruppen
 Mackie
 Midas Consoles
 Peavey Media
Matrix
 QSC Audio
Products
 Rane

TD(TECH Renkus-Heinz CROWN BY

DECISION) HARMAN
TESIRA  Soundcraft YAMAHA

 AMPETRONIC

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 TECHNICAL SPECIFICATION

AVB

Transport type: Switched network

Layer: Layer 2
Supported sample rates: No restriction
Supported bit depths: No restriction
Channel packetization: Streams (1 to 400 channels)
Latency: 2 ms standard, may be reduced in manufacturer’s implemetation
Maximum channel count: 400 in (single stream), 400 out (single stream)
Gigabit, 48 kHz/ 24 bit
As number of streams increases, channel count decreases
Channel count lower than 512 x 512 due to the benefit of bandwidth
reservation – 75% for media, 25% for non-media
Maximum CAT-5e cable length: 100 metres (328 feet)

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Cirrus CobraNet

Transport type: Switched network

Layer: Layer 2

Supported sample rates: 48 kHz, 96 kHz

Supported bit depths: 16, 20, 24

Channel packetization: Bundles of 2 or 8 channels

Latency: 5 1/3 ms (default), 2 2/3 ms, 1 1/3 ms

Maximum channel count: 32 in, 32 out (device implementation specific)

Bundle channel capacity varies as below:

Maximum CAT-5 / CAT-5e cable length: 100 metres (328 feet) 18

Audinate Dante

Transport type: Switched network

Layer: Layer 3 (IP routable)

Supported sample rates: 48 kHz, 96 kHz, 192 kHz

Supported bit depths: 16, 20, 24

Channel packetization: IP packets

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Latency: Variable from 83.3μs (device direct to device, not over network)

Increases with higher channels per IP packet

Maximum channel count: 512 in (single IP packet), 512 out (single IP

packet)

Gigabit, 48 kHz/ 24 bit

Maximum cable length: 100 meters (328 feet)

Harman BLU link

Transport type: Proprietary ring

Supported sample rates: 48 kHz, 96 kHz

Supported bit depths: 24-bit

Maximum channel count: 256 at 48 kHz, 24-bit

128 at 96 kHz, 24-bit

Channel packetization: N/A

Latency: 11 samples from transmitting device to first receiving device, 4

samples from device to device thereafter Maximum cable length: 100
metres (328 feet)

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