Telos Z/IPSTREAM R/2 User Manual

Z/IPSTREAM R/2

8 Channels of Z/IPStream, One Box.

USER’S MANUAL

Z/IPStream R/2 Manual for Software v1.01.35

Applies to: 2001-00441 through 2001-00454 and 2201-00034 through 2201-00036

Notices and Cautions

CAUTION:

The installation and service instructions in this manual are for use by qualiﬁed personnel only. To avoid electric shock, do not perform any servicing other than that contained in the operating instructions unless you are qualiﬁed to do so. Refer all servicing to qualiﬁed personnel.

This instrument has an autoranging line voltage input. Ensure the power voltage is within the speciﬁed range of 100-240v. The ~ symbol, if used, indicates an alternating current supply.

This symbol, wherever it appears, alerts you to the presence of uninsulated, dangerous voltage inside the enclosure – voltage which may be suﬃcient to constitute a risk of shock.

This symbol, wherever it appears, alerts you to important operating and maintenance instructions. Read the manual.

CAUTION: DOUBLE POLE/NEUTRAL FUSING

The instrument power supply incorporates an internal fuse. Hazardous voltages may still be present on some of the primary parts even when the fuse has blown. If fuse replacement is required, replace fuse only with same type and value for continued protection against ﬁre.

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WARNING:

The product’s power cord is the primary disconnect device. The socket outlet should be located near the device and easily accessible. The unit should not be located such that access to the power cord is impaired. If the unit is incorporated into an equipment rack, an easily accessible safety disconnect device should be included in the rack design.

To reduce the risk of electrical shock, do not expose this product to rain or moisture. This unit is for indoor use only.

This equipment requires the free ﬂow of air for adequate cooling. Do not block the ventilation openings in the top and sides of the unit. Failure to allow proper ventilation could damage the unit or create a ﬁre hazard. Do not place the units on a carpet, bedding, or other materials that could interfere with any panel ventilation openings.

If the equipment is used in a manner not speciﬁed by the manufacturer, the protection provided by the equipment may be impaired.

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WARNUNG:

Die Installations-und Serviceanleitung in diesem Handbuch ist für die Benutzung durch qualiﬁziertes Fachpersonal. Um Stromschläge zu vermeiden führen Sie keine andere Wartung durch als in dieser Betriebsanleitung aufgeführt, es sei denn Sie sind dafür qualiﬁziert. Überlassen Sie alle Reparaturarbeiten qualiﬁziertem Fachpersonal.

Dieses Gerät hat eine automatische Bereichseinstellung der Netzspannung.

Stellen sie sicher, dass die verwendete Netzspannung im Bereich von 100-240V liegt.

Das Symbol ~, falls verwendet, bezeichnet eine Wechselstromversorgung.

Dieses Symbol, wo immer es auftaucht, macht Sie auf nicht isolierte, gefährliche elektrische Spannung (ausreichend um einen Stromschlag hervorzurufen) innerhalb des Gehäuses aufmerksam. Spannungen.

Dieses Symbol, wo immer es auftaucht, weist Sie auf wichtige Bedienungs-und Wartungsanleitung hin. Lesen Sie die Bedienungsanleitung.

ACHTUNG: ZWEIPOLIGE ABSICHERUNG / NULLEITER ABSICHERUNG

Das Netzteil des Gerätes hat eine interne Sicherung eingebaut. Auch wenn die Sicherung durchgebrannt ist, können auf einigen primären Bauteilen noch gefährliche Spannungen vorhanden sein. Wenn ein Austausch der Sicherung erforderlich ist, ersetzen Sie die Sicherung nur mit gleicher Art und Wert für den kontinuierlichen Schutz gegen Feuer.

WARNUNG:

Das Gerätenetzkabel ist die Haupttrennvorrichtung. Die Steckdose sollte sich in der Nähe des Gerätes beﬁnden und leicht zugänglich sein. Das Gerät sollte nicht so angeordnet sein, dass der Zugang zum Netzkabel beeinträchtigt ist. Wird das Gerät in ein Rack eingebaut, sollte eine leicht zugängliche Sicherheitstrennvorrichtung in den Rack-Aufbau mit einbezogen werden.

Um die Gefahr von Stromschlägen zu verringern, darf dieses Produkt nicht Regen oder Feuchtigkeit ausgesetzt werden. Dieses Gerät ist nur für die Benützung im Innenbereich. Dieses Gerät erfordert freie Luftzirkulation für eine ausreichende Kühlung. Blockieren sie nicht die Lüftungsschlitze auf der Geräteoberseite und den Seiten des Gerätes. Unzureichende Belüftung kann das Gerät beschädigen oder Brandgefahr verursachen. Platzieren Sie das Gerät nicht auf einem Teppich, Poster oder andere Materialien welche die Lüftungsöﬀnungen beeinträchtigen könnten.

Wird das Gerät anders als in der, vom Hersteller angegebenen Weise verwendet, kann der, durch das Gerät gegebene Schutz beeinträchtigt werden.

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USA CLASS A COMPUTING DEVICE INFORMATION TO USER. WARNING:

This equipment generates, uses, and can radiate radio-frequency energy. If it is not installed and used as directed by this manual, it may cause interference to radio communication. This equipment complies with the limits for a class a computing device, as speciﬁed by fcc rules, part 15, subpart j, which are designed to provide reasonable protection against such interference when this type of equipment is operated in a commercial environment. Operation of this equipment in a residential area is likely to cause interference. If it does, the user will be required to eliminate the interference at the user’s expense. Note: objectionable interference to tv or radio reception can occur if other devices are connected to this device without the use of shielded interconnect cables. Fcc rules require the use of shielded cables.

CANADA WARNING:

“This digital apparatus does not exceed the class a limits for radio noise emissions set out in the radio interference regulations of the Canadian department of communications.”

“Le présent appareil numérique n’émet pas de bruits radioélectriques dépassant les limites applicables aux appareils numériques (de class a) prescrites dans le règlement sur le brouillage radioélectrique édicté par le ministère des communications du Canada.”

CE CONFORMANCE INFORMATION:

This device complies with the requirements of the EEC council directives:

♦ 93/68/EEC (CE MARKING)

♦ 73/23/EEC (SAFETY – LOW VOLTAGE DIRECTIVE)

♦ 89/336/EEC (ELECTROMAGNETIC COMPATIBILITY)

Conformity is declared to those standards: EN50081-1, EN50082-1.

Z/IPStream R/2 MANUAL

TRADEMARKS

Telos, Omnia, Axia, Z/IPStream, and the Z/IPStream logo are trademarks of TLS Corp.. All other trademarks are the property of their respective holders.

NOTICE

All versions, claims of compatibility, trademarks, etc. of hardware and software products not made by Telos which are mentioned in this manual or accompanying material are informational only. Telos makes no endorsement of any particular product for any purpose, nor claims any responsibility for operation or accuracy. We reserve the right to make improvements or changes in the products described in this manual which may aﬀect the product speciﬁcations, or to revise the manual without notice.

WARRANTY

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This product is covered by a ﬁve year limited warranty, the full text of which is included in this manual.

UPDATES

The operation of Z/IPStream R/2 is determined largely by software. We routinely release new versions to add features and ﬁx bugs. Check the Telos Alliance web site for the latest. We encourage you to sign-up for the email notiﬁcation service oﬀered on the site.

FEEDBACK

We welcome feedback on any aspect of Z/IPStream R/2, 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.

SERVICE

You must contact The Telos Alliance before returning any equipment for factory service. We will need your unit’s serial number, located on the back of the unit. The Telos Alliance will issue a return authorization number, which must be written on the exterior of your shipping container. Please do not include cables or accessories unless speciﬁcally requested by the Technical Support Engineer. Be sure to adequately insure your shipment for its replacement value. Packages without proper authorization may be refused. US customers, please contact Telos Technical Support at +1-216-622-0247. All other customers should contact local representative to make arrangements for service.

WE SUPPORT YOU...

BY PHONE / FAX:

♦ You may reach our 24/7 Support team anytime around the clock by calling +1-216-622-0247.

♦ For billing questions or other non-emergency technical questions, call +1-216-241-7225 between

9:30 am to 6:00 PM, USA Eastern time, Monday through Friday.

♦ Our Fax number is +1-216-241-4103.

BY E-MAIL:

♦ Technical support is available at [email protected]

♦ All other questions, please email [email protected]

VIA WORLD WIDE WEB:

The Telos Alliance web site has a variety of information which may be useful for product selection and support. The URL is www.TelosAlliance.com

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REGISTER YOUR PRODUCT

Did you know that all Telos Alliance products come with a 5-Year Warranty? Take a moment to activate your coverage online at http://telosalliance.com/product-registration/ .

The Telos Alliance

1241 Superior Avenue Cleveland, OH., 44114 USA

+1-216-241-7225 (phone)

+1-216-241-4103 (fax)

+1-216-622-0247 (24/7 technical support)

[email protected]

TABLE OF CONTENTS

1 A Letter from our CEO, Frank Foti VII

1 About This Manual (Or: Start Here…Really.) IX

1 So what is Z/IPStream R/2? 1

2 A few words about Livewire+ and AES67 2

3 Installation 5

4 TFront Panel Interface 10

5 IP Address Conguration 13

6 Web GUI Navigation 16

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7 Audio Source Conguration 22

8 Processing and Encoding 31

9 Omnia 3 Processing Setup 64

10 Metadata Sources and Filters 75

11 Event Scheduling 79

12 License Management 82

13 Omnia 3 Preset Management 84

14 System Options 86

15 Logging 98

16 Omnia.9 Processing Setup 100

17 Frequently Asked Questions (FAQ) 151

18 Specications 153

A1 Telos Alliance Limited Warranty 155

War of the Waves

Dear Valued Customer,

It’s with great pride and a tip of the hat to an incredible team that I congratulate you on your new Telos Alliance product. Everything we do here at the radio division of the Telos Alliance is with one end goal in mind: To help broadcasters declare victory in extremely competitive environments. By purchasing this product from us, in essence, you have declared war on your competition.

After all, the majority of Telos Alliance employees were broadcasters themselves once, and the products we’ve developed over the years have been designed as solutions to speciﬁc issues faced on the front lines of our industry. We’re right there in the trenches with you and have the weapons you need in your arsenal.

Telos Systems is a catalyst to out-of-this-world sound, with the most powerful and popular broadcast telephone systems in the industry; IP/ISDN codecs and transceivers; plus processing/encoding for streaming audio. We built an industry on the back of these amazing telephony systems, and they are still going strong.

While we at the Telos Alliance never forget our roots, we are also blazing trails in terms of new technologies like stream-encoding and AoIP, so that all types of broadcasters can excel in this ever-evolving digital world. Omnia Audio not only lets you stand out on the dial with your unique signature sound via legendary audio processors, audio codecs, and microphone processing, it lets you give your listeners a better streaming experience across devices with innovative stream encoding/processing software and hardware.

Axia Audio is a driving force behind the AES67 AoIP standard, and its networked AoIP radio consoles, audio interfaces, networked intercom, and software products continue to move AoIP adoption forward and help broadcasters streamline operations with cohesive, smart, and feature-rich AoIP ecosystems.

Last, but certainly not least, 25-Seven has traditionally been known for its audio delays, but its Voltair watermark monitor/processor has made a name for itself more recently as the disruptive product that helped broadcasters take back their ratings and harness the true power of their listening audiences.

You work so hard on your programming day-in and day-out, it deserves technology that will optimize sound and performance at every point in the airchain and online. Armed with Telos products, you have what you need to set your competition squarely in your crosshairs.

With that, I’ll leave you to prep your armaments. I hope that you will enjoy your Telos Alliance products for many years to come!

Sincerely,

Frank Foti

CEO, The Telos Alliance

About This Manual

Read Me. Please. (OK, at least read some of me.)

If you’re like many engineers or IT types, you’ve probably already put your brand new R/2 into the rack, connected power, connected the network, and perhaps even connected some audio. You may have even powered it up and ﬁgured out how to set an IP address to access the web GUI. If that’s you…Great! You’re a few steps ahead of the game and we hope this manual can provide some additional valuable information for you.

If that’s totally not you and you’re more of a “by the book” kind of person, that’s great too. We hope you’ve at least gone through the included Quick Start Guide and found it helpful. This manual will cover some of the same basics as the Quick Start Guide, and go into great detail about how to get your R/2 up and running.

The layout of this manual is pretty straightforward…It should be in the order you would normally perform installation and conﬁguration steps for the most part (though there may be a few little rabbit trails). The author has tried to take a “what would someone trying to set this up for the ﬁrst time need to know, and in what order” approach. Some of the chapters dealing with audio processing setup or other conﬁguration details will be a bit longer than others, but overall this manual should be a pretty quick read.

We realize reading manuals isn’t everyone’s idea of a great time, but we’ll try at least to keep it informative and somewhat entertaining—though the author has a habit of cracking some pretty bad jokes. If you aren’t asleep by the end of it, and have your R/2 up and running, this manual has accomplished its goal.

There are probably a lot of things you could be doing with your time besides reading a manual, and we appreciate that you’ve taken the time to read this one. If you REALLY don’t have time to read anything else, start with chapters 3 through 7. Chapter 8 is probably the longest in the manual, but covers how to actually make R/2 start doing stuﬀ. If you’re in even more of a hurry, jump straight to the Frequently Asked Questions at the end of this manual (Chapter 16).

If after reading this manual you have any suggestions for improvement (or if you see something we might have missed) send a note to [email protected].

As always, our support team is available 24/7 to assist by calling 216-622-0247 or e-mail [email protected].

The world is listening. Stream like you mean it.

1 So what is Z/IPStream R/2?

A Bit of History

It started with the telephone…

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For over 30 years, Telos has led the industry in telephony and audio encoding technology for broadcast. When Steve Church developed the ﬁrst digital telephone hybrid for broadcast in 1984, it was the beginning of a revolution in how the industry would get audio from the ﬁeld on the air. Telephone calls had never sounded so good.

MP3 and ISDN

This revolution continued with the introduction of the Telos Zephyr ISDN codec in 1993. It was the ﬁrst codec to make use of the (then relatively new) MPEG Layer III codec to send full 20 kHz mono audio across a single 56 or 64 kbps data channel. Broadcasters could ﬁnally bring full ﬁdelity audio to their listeners from the ﬁeld (or between facilities) without dedicated equalized program loops or RF links. The Zephyr quickly became the de-facto industry standard for ISDN codecs, used by studios around the globe for transporting audio.

SO WHAT IS Z/IPSTREAM R/2?

Streaming…It begins.

While ISDN was becoming popular for broadcasters to transport audio from the ﬁeld or between facilities, the next frontier was exploring new methods for getting audio to the listeners. The Internet was still in its infancy, but it had developed to the point where streaming audio (even over dialup) was becoming viable. In 1997, Telos introduced the very ﬁrst dedicated hardware encoder for MP3 streaming over the Internet, simplifying what was a bit of a cumbersome process and dramatically improving the audio quality.

Omnia: A Legend in Audio Processing for Broadcast

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During this same period of time, Frank Foti was working tirelessly to improve the sound of audio processing for broadcast. His dial-dominating innovations in audio processing at Z100 were the foundation of what would ultimately become Omnia. The audio processing developments didn’t stop at the FM dial, however…It quickly became apparent that there was considerable beneﬁt to be gained from specialized audio processing for streaming as well.

Telos Encoding + Omnia Processing = Z/IPStream

As streaming continued to gain in popularity, the demand for an “all in one” solution to handle both processing and encoding for streams grew tremendously. In 2009, Omnia launched the A/XE software package, combining the best of Telos encoding technology with 3-band Omnia audio processing tailored speciﬁcally for streaming. In 2012, Telos introduced R/1, a new hardware appliance designed speciﬁcally for stream processing and encoding without a dedicated PC.

SO WHAT IS Z/IPSTREAM R/2?

The Z/IPStream family gained yet another powerful option with the availability of acclaimed Omnia.9 processing in 9X/2 streaming software. Z/IPStream represents the latest in encoding and audio processing developments from The Telos Alliance, and supports a wide variety of streaming technologies including adaptive bitrate streaming for a seamless listener experience regardless of network conditions, or listening platform.

About R/2

Up to 8 Channels of Z/IPStream, one box

The R/2 is a dedicated multichannel processor/encoder appliance for streaming, designed for applications demanding the ultimate in reliability, with the highest possible audio quality and channel density per rack space.

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Features

Z/IPStream R/2 supports up to 8 simultaneous audio channels, including the latest generation of licensed MP3 and AAC codecs for streaming, with signature Omnia 3-band or powerful Omnia.9 processing. Flexible conﬁguration allows multiple processing and/or encoding instances per audio channel.

Audio I/O includes AES/EBU, Livewire Audio over IP, and multicast or unicast RTP. Supported server platforms include HTTP, Icecast, Shoutcast, Wowza, Adobe RTMP, and others.

Triton Digital Integration

R/2 includes direct support for the Triton Digital streaming platform, which allows users to take full advantage of their content distribution and dynamic ad replacement technologies, while streamlining hardware and software implementation.

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2 A Few Words about Livewire+ and AES67

What is Livewire+™ ?

Livewire+ is the pioneering technology invented by the Telos Alliance to convey low-delay, high-reliability audio over switched Ethernet.

Introduced as Livewire™ in 2003, today’s Livewire+ is AES67-compliant. That means that it complies fully with the AES67-2013 Interoperability Standard, allowing AES67 devices to connect directly to Livewire+ networks and exchange audio streams. Livewire+ is also extensible, able to incorporate future AES standards when they are ratiﬁed. Livewire+ is also backward-compatible with the RAVENNA™ networking protocol.

With Livewire+, a single Ethernet cable carries real-time uncompressed digital audio, device control messages, program associated data, and even routine network traﬃc. An entire facility can be wired in hours, instead of weeks. All Axia Audio studio products, and most products from other Telos Alliance brands, utilize Livewire+ networking technology. Expanding or modifying your system is simple thanks to the inherent scalability and modularity of Livewire+.

Livewire+ oﬀers a revolutionary change in how studios can be built. But at the same time, it’s a natural continuation of general trends and what you already know.

A FEW WORDS ABOUT LIVEWIRE+ AND AES67

How Livewire+ works

Livewire+ has an audio advertising system. Every source has a text name and numeric ID. These are transmitted from source devices to the network. Devices that play audio build lists of all available sources from which users can select.

UsingxNode audio interfaces, you enter the names of your input sources via any PC with a web browser. Withplayout PCs attached to the network, you open a conﬁguration window.

Livewire+ networks employ two types of audio streams.Livestreamshave small, frequent packets optimized for live audio that requires very low (circa1 ms.) delay, for microphones and headphone audio.Standard Streamsare also real-time streams, but with bigger packets, and are used for audio streams which don’t require super-low latency - like audio from CD players, or that exchanged with automation system PCs. Devices that connect to Axia networks can transmit and receive both stream types; the user selects which type to generate when a device is initially conﬁgured.

A sophisticated phase-locked loop clocking system allows Livewire+ to use very small buﬀers for least latency and ensures that audio channels remain time-aligned (as needed for multiple mics in a studio or for TV surround-sound mixing.)

Converged Networks

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An Ethernet network used for Livewire+ audio can also be shared with other data transmissions, such as ﬁle transfers and web browsing. An Ethernet system with a switch at the center may have a mix of audio nodes and normal servers, PCs, etc., because the Ethernet switch directs traﬃc only to where it is needed.

Even on a single link, traﬃc can be mixed because we use modern Ethernet’s priority mechanism to be sure audio packets have ﬁrst call on the link’s bandwidth. A studio audio delivery system can use this capability to download an audio ﬁle from a server, for example, while simultaneously playing another audio ﬁle live.

Livewire+ maximizes the beneﬁts of converged networking in the broadcast facility. Many stations using Livewire+ have computer data, telephone, audio, and control on a single network that uses computer industry standard wiring, spurring cost-eﬃciencies throughout the plant.

Audio Quality

A Livewire+ network is a controlled, high-speed environment, with no risk of audio drop-outs from network problems and plenty of bandwidth for many channels of high-quality uncompressed audio. We use studio-grade 48kHz/24-bit PCM encoding. Axia digital xNode audio adapters deliver 138dB of dynamic range, with less than

0.0002% THD. Even analog xNodes have 100dB dynamic range, < 0.005% THD, and headroom to +24dBu.

Livewire+ is standards-based

Since the very beginning, The Telos Alliance has based its AoIP networking technologies on standards. IP (Internet Protocol), the networking standard that is the underpinning of nearly all critical business networks (and of the Internet itself ) is the basis for Livewire+ AoIP.

As charter members of the AES X.192 Working Group, we helped deﬁne the AES67 standard — and became the ﬁrst broadcast manufacturer to become AES67 compliant.

Livewire+ is so standards-based, in fact, that your audio can even be played by PC media playersthat support standard protocols and uncompressed PCM audio. The Internet’s IP standard for streaming media, called RTP/IP, is used for standard audio streams. RTP stands for Real-Time Protocol. It’s the Internet’s standard way to transport streaming audio and video, just as TCP/IP is the standard for general data.

A FEW WORDS ABOUT LIVEWIRE+ AND AES67

The Gold Standard

In the decade since the introduction of Livewire+, broadcasters have adopted it at an exponential rate; AoIP has become the new standard for broadcast facilities.

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Consider these facts:

♦ There are over 6,000 studios worldwide built with Livewire+.

♦ More than 5,500 networked Axia consoles are at work daily.

♦ Over 80 Livewire+ Partner companies provide compatible hardware and software products.

♦ There are more than 60,000 individual Livewire+ devices in the ﬁeld.

♦ Livewire+ is fully compliant with the AES67 Interoperability Standard.

Impressive, no? But there are even more exciting things in the future. The Telos Alliance, with one of the largest R&D groups in broadcast, is fully committed to AoIP interoperability. We’ve been proponents of open standards since Day 1, freely sharing our technology with interested Hardware and Software Partners. We were charter, supporting member of the X.192 Working Group that deﬁned the AES67 standard, and as founding members of the Media Networking Allliance, are actively engaged in work to promote and enhance standards-based AoIP networking.

3 Installation

Physical Considerations

Rack Mounting

R/2 includes a pair of rack rails designed to secure the unit to the rear of the equipment rack. It is highly recommended that you install these rails if possible due to the depth of the unit. Mounting by the front panel alone may cause excessive strain on the chassis and is not recommended. The supplied rails should ﬁt in most standard equipment racks, but if installing the rails is not possible, you may wish to consider placing an upside-down rack shelf underneath the unit for additional physical support.

Airow

As with most rackmount server chassis, the airﬂow on R/2 is from front to back. While it is not absolutely necessary to leave an additional rack space above and below the unit, it certainly is not a bad idea if the space is available. Do not block the front panel of R/2 behind a solid rack door.

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Wiring Management

Since airﬂow is from front to back, it is important to keep the rear of the unit free from obstructions to allow for hot air exhaust. Install the wiring in such a way that it allows for a reasonable service loop and does not block airﬂow. Do not allow wiring from other equipment in the rack to block airﬂow from the rear panel of R/2.

Rear Panel

INSTALLATION

Power

The R/2 is equipped with dual power supplies (120V – 240V, 50 – 60Hz) to allow for facilities with redundant power feeds to the equipment racks from separate sources. An LED on the supply will indicate the presence of AC power and proper operation. If power is lost on one of the AC inputs or one of the supplies should fail, the R/2 will sound an alarm. The only way to mute this alarm is to connect power to both supplies or (in the case of power supply failure) remove the failed supply.

RS-232

Not currently implemented. Inserting stream metadata via RS-232 will be supported in a future release. See chapter 10 (Metadata Sources and Filters) for more details about stream metadata.

“Control” Ethernet Port

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Note:

If only one of the two power supplies is connected, an alarm will sound. Connect both supplies to AC power, even if you don’t have redundant AC power feeds to the equipment rack.

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This port (if populated) is intended for out-of-band system management. Not currently implemented.

USB Ports

Future expansion or factory service. Not currently implemented.

NIC 1 and NIC 2

The R/2 oﬀers dual network adapters. Typically, one adapter (NIC1) would be used for Audio over IP, while the second (NIC2) would be dedicated to streaming audio traﬃc. Either interface can be used for management via the web GUI. Connect your network(s) to NIC 1 and/or NIC 2. Do not connect both interfaces to the same network. Doing so may cause routing and connectivity issues.

Note:

Do not connect both interfaces to the same network simultaneously. Routing and connectivity issues may result. Ensure each interface is connected to a dierent IP network, (dierent VLAN, physical switch, IP address block, etc.) as necessary. Consult your IT department or Telos technical support if you are not sure how the interfaces should be congured. Only one interface may be needed in some rare cases.

INSTALLATION

VGA

Factory service only. Not currently implemented.

Audio (AES/EBU)

Eight AES/EBU audio connections are available through two high-density 26-pin D-sub connectors. Connect the provided breakout cable(s) to the D-sub connectors. Use the thumbscrews to securely attach the cables in place. Sample rate conversion is provided on all inputs. The BNC Word Clock connectors are not currently implemented. Each audio channel can be conﬁgured to generate a stream at 44.1 or 48 kHz (see Chapter 7: Audio Source Conﬁguration).

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AES Breakout Cable Layout

INSTALLATION

Audio Card Pinout

Port A, Jumpers

Pin Signal Pin Signal Pin Signal

1 Clock Gnd 10 Clock In

2 OUT4 Cold 11 OUT4 Hot 19 OUT4 Gnd

3 OUT3 Gnd 12 OUT3 Cold 20 OUT3 Hot

4 OUT2 Cold 13 OUT2 Hot 21 OUT2 Gnd

5 OUT1 Gnd 14 OUT1 Cold 22 OUT1 Hot

6 IN4 Cold 15 IN4 Hot 23 IN4 Gnd

7 IN3 Gnd 16 IN3 Cold 24 IN3 Hot

8 IN2 Cold 17 IN2 Hot 25 IN2 Gnd

9 IN1 Gnd 18 IN1 Cold 26 IN1 Hot

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Port B, Jumpers

1 Clock Gnd 10 Clock Out

2 OUT8 Cold 11 OUT8 Hot 19 OUT8 Gnd

3 OUT7 Gnd 12 OUT7 Cold 20 OUT7 Hot

4 OUT6 Cold 13 OUT6 Hot 21 OUT6 Gnd

5 OUT5 Gnd 14 OUT5 Cold 22 OUT5 Hot

6 IN8 Cold 15 IN8 Hot 23 IN8 Gnd

7 IN7 Gnd 16 IN7 Cold 24 IN7 Hot

8 IN6 Cold 17 IN6 Hot 25 IN6 Gnd

9 IN5 Gnd 18 IN5 Cold 26 IN5 Hot

Power Up

Press the “On/Oﬀ” button located on the right half of the front panel to power up the unit (See Chapter 4 for a full front panel description). During the power up process, the front panel will light up and display a splash screen. If the display remains dark and fans do not spin up, verify that AC power is present at both power supply inputs.

Pin Signal Pin Signal Pin Signal

If the fans spin up but the display remains dark or does not show a splash screen, contact support.

INSTALLATION

Fans

The fans in R/2 are thermostatically controlled. As the unit powers up, the fans will start at full speed. They should spin down to a much slower speed after a short period of time. If they do not return to low speed, it may indicate a faulty fan or other hardware issue. During normal operation, these fans may increase in speed if the unit’s internal temperature rises. If the fans remain at full speed during operation and the environmental temperature of the unit appears normal, contact support.

Input Level Meters

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Note:

The fans will run at full speed for a brief period of time during boot. They should return to a relatively moderate speed within a short period of time. If the fans remain at full speed, it may indicate a hardware fault. Slightly higher fan speed during operation is normal if the internal temperature of the unit increases.

Once the unit has fully booted, the front panel LCD will display the input audio level meter screen. As of software version 1.0.34, these meters can display either AES or Livewire input levels by using the up and down navigation arrows on the front panel.

Livewire and other Audio over IP input sources (as well as output level metering and processing metering) can also be metered via the unit’s web GUI or via NfRemote for Omnia.9 processing (See Chapter 16 for Omnia.9 processing information).

Note:

Select between Livewire or AES input levels by using the up and down navigation arrows from this screen (see chapter 7 for information about dening audio sources). Metering can also be accessed from within the web GUI or NfRemote software (if using Omnia.9 processing).

4 Front Panel Interface

Overview

This chapter describes the front panel controls and indicators of R/2.

Status LEDs

A column of four LEDs is located directly to the left of the main LCD. The top LED illuminates when the system is operating. The others are not currently implemented.

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Main LCD

The main LCD provides unit status and conﬁguration. If this LCD fails to display the Z/IPStream R/2 splash screen at startup, or remains blank during operation, please contact technical support.

Navigation Cluster

A six button navigation cluster is used for initial IP address conﬁguration and navigating the front panel LCD. The controls are relatively self explanatory. Up and down will increment or decrement values, or scroll between parameter settings. Left and right will move between digits in IP address entry, or move between screens on the front panel. The red “X” button will cancel an entry, the green “” (OK button) will conﬁrm the current setting or select a parameter for modiﬁcation.

FRONT PANEL INTERFACE

System Status LEDs

A set of ﬁve LEDs next to the power and reset buttons indicates system status.

Info

Used for alert indication or system identiﬁcation.

♦ Solid red: The system has overheated. This may be caused by reduced airﬂow or high ambient tempera-

ture in the rack. Ensure there is nothing obstructing the front or back of the unit and that the temperature in the rack is normal.

♦ Blinking red (1 second intervals): Fan failure. Contact technical support.

♦ Blinking red (1/4 second intervals): Power supply failure. Check for a failed power supply, contact

technical support.

♦ Blue (solid or blinking): Local or remote Unit ID function—Not currently implemented.

Net 2

Indicates NIC 2 link and activity. Flashes during activity. This LED will remain dark if NIC 2 is not connected.

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Net 1

Indicates NIC 1 link and activity. Flashes during activity. This LED will remain dark if NIC 1 is not connected.

SSD

Indicates activity of the internal Solid State Drive. Flashes intermittently during normal operation.

Pwr

This is the main system power LED and will illuminate whenever the system is powered up.

System Controls

Reset

Performs a “hard” reset on R/2 to reboot. Should not be used during normal operation, but can be used if the R/2 becomes unresponsive for some reason. Should this occur, contact technical support.

On/O

Performs an orderly power up or shut down. A slight delay before shut down after pressing this button is normal. The SSD indicator will ﬂash rapidly for a short period of time during shut down.

FRONT PANEL INTERFACE

Front Panel USB Port

This port is used for software upgrades, as well as factory conﬁguration and troubleshooting. Contact technical support for additional information.

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5 IP Address Conguration

Overview

After physical installation of R/2 has been completed, at least one of the two network interfaces must be assigned an IP address to allow further conﬁguration and normal operation.

IP Conguration

Each of the two network interfaces on R/2 can be assigned a static IP address, or obtain an IP address automatically via DHCP (Dynamic Host Conﬁguration Protocol). In most applications, R/2 would be assigned static IP addresses, but it may also be helpful to use DHCP with reserved assignments tied to the MAC addresses of R/2. Contact your network administrator if you are unsure of what your network requires.

Metering Screen

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After the unit boots, it should be displaying the audio input level metering screen. This is the normal “home” condition for the R/2 display. The up/down arrows will select between AES and Livewire input levels (see Chapter

7). “X” or the left/right arrows will allow you to return to this point from other areas of the front panel screens.

NIC Selection

From the metering screen, use the right arrow to move to the “Interfaces” screen. Select “NIC 1” or “NIC 2” by using the up/down arrow keys. The display arrow will move to indicate the current selection. Press the “” key to select the desired interface.

Network Mode

IP ADDRESS CONFIGURATION

Once you have selected the desired interface (displayed here in the upper right hand corner of the LCD), you can select which mode that interface should operate in. By default, R/2 ships with both interfaces conﬁgured for DHCP.

Verify Current Interface Parameters

To verify the current IP address and mode, from the “Network Mode” screen, press the right arrow. In the example above, NIC 1 is set for DHCP and has an address of 192.168.123.154 with a subnet mask of 255.255.255.0 and a gateway of 192.168.123.253. Press the left arrow or “X” to return to the “Network Mode” screen.

Note:

If the network cable is not connected to an interface, one of two conditions may occur…A “self-assigned” IP address of 169.x.x.x or all zeros (even if a static IP has been assigned). This may also occur if the unit is congured for DHCP but not able to reach a DHCP server.

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Conguring a Static IP

From the “Network Mode” screen, press the “” key. You will be asked to switch to static mode. Note the interface number displayed in the upper-right hand corner (NIC 1 in this example).

Use the up arrow key to select “Yes”, then press the “” key. The “Static IP” conﬁguration screen will be displayed.

IP ADDRESS CONFIGURATION

To modify the static IP conﬁguration:

♦ Use the up/down arrow keys to select the ﬁeld you want to change (IP address, Netmask, or Gateway).

♦ Press the “” key to begin editing a ﬁeld.

♦ While editing a ﬁeld, use the left/right arrow keys to select the digit you wish to change, then the up/

down arrow keys to change the value.

♦ When done editing, press the “” key.

When ﬁnished entering all values, press the right arrow key. The display will prompt: “Apply Changes?”

Use the up arrow key to select Yes, then press the “” key to select. After a few moments, you will be returned to the “Network Mode” screen.

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To conﬁgure the second NIC, return to the “Interfaces” screen, select “NIC 2”, and repeat the conﬁguration process.

6 Web GUI Navigation

Overview

Beyond initial IP address settings from the front panel, all R/2 conﬁguration is performed through a web browser (or through NfRemote software for Omnia.9 processing—See Chapter 16). This chapter will provide an overview of the R/2 web GUI.

On a computer with access to the same network as R/2, enter the R/2’s IP address (conﬁgured from the front panel in the previous chapter) in a web browser. The unit can also be located by the hostname: R2-XXXXXX, replacing XXXXXX with the serial number of the unit. For example, unit 74-1001 will be R2-741001.

Note:

Most modern browsers are supported (including Chrome, Firefox, Safari, Edge, and others that support HTML 5). No Java or Flash is used in the R/2 web GUI.

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Login

Enter user as the user name. There is no default password. click the Submit button. You should now be logged into Z/IPStream R/2.

WEB GUI NAVIGATION

Control Panel

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The main “home” display is called the “Control Panel” This is the screen where the status of all processing and encoding instances is displayed, and new processing or encoding instances are deﬁned (Chapter 8). All other conﬁguration functions can be accessed through the sections listed at the top this screen. Click the “Z/IPStream R/2” logo at the upper left corner or “Main” to return here at any time. “Help” in sub-menus will download a copy of this manual, and “Logout” will return to the login screen.

WEB GUI NAVIGATION

Audio Sources (Chapter 7)

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R/2 is licensed by number of audio sources. These sources must be deﬁned before processing and encoding instances can be conﬁgured. The “Audio Sources” screen deﬁnes available audio channels for processing and encoding. These sources can be AES inputs from the internal audio card, Livewire channels from an Audio over IP network, or other RTP audio stream (Unicast or Multicast).

Metadata (Chapter 10)

Web streams frequently have “Now Playing” metadata associated with them for display on websites and other streaming clients. The “Metadata” section of the interface is where these metadata sources are deﬁned. Incoming data via TCP, UDP, is supported. RS-232 will be supported in a future version. Custom XML “Metadata Filters” allow considerable ﬂexibility in data formatting and translation.

WEB GUI NAVIGATION

Schedule Manager (Chapter 11)

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R/2 supports timed scripting of events for various functions. This could include switching to an alternate audio source on a stream, changing a processing preset, or blacking out a sports broadcast or other content for which you do not have streaming rights.

Licenses (Chapter 12)

R/2 is a modular platform which can be licensed for various numbers of channels (from 2 up to 8) and types of audio processing (Omnia 3-band or Omnia.9). This screen allows management of the licenses available on the system.

WEB GUI NAVIGATION

Processing Presets (Chapter 13)

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The “Preset Editing” screen allows management of audio processing presets for the standard Omnia 3 processing. Presets can be added (copied from an existing preset), uploaded from a compatible preset ﬁle, or deleted from the unit. Omnia.9 processing presets are managed through NfRemote software (see Chapter 16).

Options (Chapter 14)

Includes global conﬁguration options for R/2, including management of background software services, as well as software updates, and backup and restore of system conﬁguration. Network parameters can also be conﬁgured from this section.

WEB GUI NAVIGATION

Logs (Chapter 15)

Comprehensive logs are kept for diagnostic and other purposes. This screen allows viewing or downloading logs from R/2.

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7 Audio Source Conguration

Overview

This chapter will cover deﬁning the audio sources available for processing and encoding in R/2—one of the most critical steps in conﬁguration after setting the IP addresses and accessing the web GUI.

R/2 is licensed by number of audio sources. Two audio sources are standard with R/2, up to eight can be licensed per unit. Each of these audio sources is available for multiple instances of processing and/or encoding. 3-band Omnia processing is standard, with optional Omnia.9 processing available.

Audio Sources

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This screen will display the audio sources currently available on the system, and allow you to add new or remove existing sources. To add a new audio source, click on “Add audio source”. To remove an existing source, select it in the list, then click “Remove”. Ensure that the source is no longer in use by any encoding or processing instances before removing it.

“Add Audio source” menu

Select the type of audio source you wish to add.

AUDIO SOURCE CONFIGURATION

Wave: Local audio card

Allows input from a local AES/EBU audio source or Livewire audio channel. Selecting this option will open the “Add Wave Audio Source” panel.

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Friendly name

Since many audio source names on the system can be very similar, give this audio source a “Friendly” name that will be easier to remember when conﬁguring processing and encoding instances (for example, Livewire 1, AES 1, Classical, Jazz, or whatever else will make it easier to identify in R/2).

Sampling rate

This is the sample rate that the streams from this audio channel will be generated at. Available options are 44.1 kHz or 48 kHz. Choose whichever sample rate best suits the streaming application.

AUDIO SOURCE CONFIGURATION

Source device

Displays a list of available source devices on the system. There are up to 24 possible Livewire inputs available (deﬁned via the “Livewire Driver” menu) and 8 AES inputs available.

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Notes:

Available Livewire I/O channels are mapped via the “Livewire Driver” menu.

Audio card port “A” carries channels 01 through 08 (In 1-4) while port “B” carries channels 09 through 16 (In 5-8).

AUDIO SOURCE CONFIGURATION

AES Breakout Cable Layout

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RTP: Local port

Allows input via a unicast RTP stream on a user-deﬁned IP port. Selecting this option will open the “Add RTP Audio Source” panel.

AUDIO SOURCE CONFIGURATION

Friendly name

Since many audio source names on the system can be very similar, give this audio source a “Friendly” name that will be easier to remember when conﬁguring processing and encoding instances (for example, Livewire 1, AES 1, Classical, Jazz, or whatever else will make it easier to identify in R/2).

Sampling rate

This is the sample rate that the streams from this audio channel will be generated at. Available options are 44.1 kHz or 48 kHz. Choose whichever sample rate best suits the streaming application.

RTP Port

Enter the port that will receive this RTP audio stream.

Receive Buﬀer Size (Packets)

Allows precise adjustment of the incoming RTP audio stream buﬀer for time alignment, or other purposes.

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Use PTP timestamp

Speciﬁes use of an IEEE 1588 PTP (Precision Time Protocol) timestamp for stream synchronization purposes.

Timestamp oﬀset (ns)

Allows an oﬀset to the incoming IEEE 1588 PTP timestamp to be speciﬁed.

RTP: Multicast

Allows input via a multicast RTP stream (including Livewire+/AES67). Selecting this option will open the “Add RTP Multicast Audio Source” panel.

AUDIO SOURCE CONFIGURATION

Friendly name

Since many audio source names on the system can be very similar, give this audio source a “Friendly” name that will be easier to remember when conﬁguring processing and encoding instances (for example, Livewire 1, AES 1, Classical, Jazz, or whatever else will make it easier to identify in R/2).

Sampling rate

This is the sample rate that the streams from this audio channel will be generated at. Available options are 44.1 kHz or 48 kHz. Choose whichever sample rate best suits the streaming application.

Channel number

Speciﬁes the Livewire or other Audio over IP network channel number associated with the desired RTP multicast stream.

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Browse

Displays a list of multicast channels and streams available on the network. Select a channel from this list to populate the “Channel number” ﬁeld. Select “Reload channels list” to refresh this list of available channels.

Note:

Both this method and the “Livewire Driver” method described later in this chapter can be used to dene Livewire channels as available audio sources.

Use PTP timestamp

Speciﬁes use of an IEEE 1588 PTP (Precision Time Protocol) timestamp for stream synchronization purposes.

Timestamp oﬀset (ns)

Allows an oﬀset to the incoming IEEE 1588 PTP timestamp to be speciﬁed.

AUDIO SOURCE CONFIGURATION

Triton Digital Services

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R/2 oﬀers a direct streamlined interface to Triton Digital Services for streaming and ad replacement. Stations can be added to R/2 via the “Triton Digital Services” screen. This screen also allows stations to be removed from R/2, or toggled on and oﬀ.

Adding a Triton Digital Services Station

Enter the Triton Digital Services login and password for the station you wish to add and click “Save”.

Click “Remove” to remove a station from the list, or “Toggle” to enable and disable a station.

Click “Main” to return to the main R/2 “Control Panel” screen, or “Audio Sources” to return to audio source conﬁguration.

AUDIO SOURCE CONFIGURATION

Livewire Driver

The “Livewire Driver Conﬁguration” screen allows conﬁguration of the integrated Livewire IP Audio Driver and provides a direct mechanism for deﬁning the Livewire channel names and numbers used by R/2. Up to 24 Livewire channels are available and correspond to the 24 Livewire channels available via the “Source Device” menu under “Add Wave Audio Source”.

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Network Adapter

Select the desired network interface for to be used for Livewire traﬃc. The IP addresses listed for these two interfaces should correspond to the ones assigned for NIC 1 and NIC 2.

Discovered channels (DST)

Displays a list of Livewire channels available on the network. This list is for reference only. Selecting channels in this list will not directly populate the channel number ﬁelds. Select “Reload channels list” to refresh this list of available channels.

Note:

If no channels are displayed in this list, rst verify that one of the NIC interfaces is connected to the Livewire network and properly congured. Second, attempt to restart the R/2 service (not the entire R/2 unit, just the R/2 “service” via the “Services” menu under “Options”—Refer to Chapter 14 for details)

AUDIO SOURCE CONFIGURATION

Name

Enter a name to uniquely identify this channel on the Livewire network. It will appear on other Livewire devices as

name@ZIPStreamR2 (see the above channel list for an example).

SRC (Send)

Enter a unique Livewire network channel number from 1 to 32767 that each IP audio driver channel will SEND to as a SOURCE. Enter “0” to disable sending to the Livewire network from a channel.

Note:

Each device MUST use unique channel numbers when sending audio to the Livewire network. It is recommended that you develop a channel numbering scheme to ensure that every device on the network is always assigned a unique block of channel numbers. For example, the last octet of the IP address with the channel number appended is one such example. In this case, the channel numbers assigned to this particular R/2 would be 22201-22224.

DST (Receive)

This section allows R/2 to RECEIVE audio as a DESTINATION FROM the Livewire network. Using the “Discovered Channels” list, enter a valid channel number from the Livewire network to assign it to an audio driver channel. Audio from this network channel will appear on the corresponding Livewire driver channel in the “Source Device” menu under “Add Wave Audio Source”.

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Click “Main” to return to the main R/2 “Control Panel” screen, or “Audio Sources” to return to audio source conﬁguration.

8 Processing and Encoding

Overview

Once the audio sources have been deﬁned, the next step is to conﬁgure the processing and encoding instances that will use those sources. Each audio source can be used for audio processing only, encoding only, or any combination of the two. Available encoding modes include AAC, MP3, and adaptive AAC. Multiple codecs, bitrates and encoding destinations are supported for each audio source. This is potentially one of the most complex yet extremely powerful conﬁguration sections within R/2.

Dening Processing and Encoding Instances

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From the Main “Control Panel” window, click “New”.

PROCESSING AND ENCODING

Create New Program

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The “Create New Program” window allows selection of the “program type” to be deﬁned. Click on the program type that you wish to create.

Note:

Multiple processing and/or encoding instances (program types) are supported for each of the audio sources dened previously (see Chapter 7). Encoder instances can also contain multiple destinations and bitrates for encoding.

Audio processing instance

PROCESSING AND ENCODING

While R/2 is designed primarily for streaming applications, there may be cases where you wish to process an audio signal without encoding a stream for a server. The “Audio Processing Instance” program allows processing an audio I/O without encoding. This can be useful for “utility” processing of audio around a facility, or as “backup” to a main audio processor, particularly when using the optional Omnia.9 processing.

Name

Enter a “friendly name” for this processing instance to easily identify it for status and control from the main “control panel” screen (i.e. Kxxx-Backup Processing).

Audio Source

Select one of the previously deﬁned audio sources as the input to this processing instance (see Chapter 7). Note that these sources can be used in multiple processing or encoding instances simultaneously.

Processing

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Selects between Omnia 3 and optional Omnia.9 processing (or no processing at all, should you simply want to route audio through this instance without processing). Up to 8 channels of Omnia.9 processing can be licensed per R/2. When you have reached the maximum number of Omnia.9 licenses available (or if Omnia.9 processing is not licensed) this option will not be displayed. See Chapter 16 for more information on Omnia.9 processing.

Processing preset

Selects the processing preset to be used for this instance. When Omnia 3 processing is selected, the list of available presets will appear. These presets are not format speciﬁc, though there are bitrate speciﬁc presets for “Music” and “Talk”. Experiment with various presets to ﬁnd one that works best for your programming format and application. Presets can be copied and modiﬁed to create a sound customized exactly to your needs. See Chapter 9 to learn how to “ﬁne tune” the Omnia 3 processing, and Chapter 13 for more information on Omnia 3 preset management.

PROCESSING AND ENCODING

If optional Omnia.9 processing is enabled, the preset must be selected via NfRemote software due to the powerful processing tools and test instrumentation available through Omnia.9. See Chapter 16 for more details.

Output device

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The available output destinations for audio from this instance are listed here (up to 24 Livewire outputs, and 8 local AES outputs). Select the output destination for this processing instance. Livewire channels are conﬁgured through the “Livewire Driver Conﬁguration” interface. See Chapter 7 for more details.

Sync buffer (Samples, Seconds)

These values will allow adjustment of the synchronization buﬀer between input and output. It is recommended that you leave these at the defaults unless advised by technical support. Improper adjustment can cause buﬀering issues.

Cancel/Save

Once you have ﬁnished conﬁguring the processing instance, click “Save” to save it, or “Cancel” to return to the Control Panel screen without saving changes.

PROCESSING AND ENCODING

Standard AAC Encoder Instance

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The Standard AAC Encoder instance accepts audio from a source, encodes it to AAC, and then passes the encoded stream along to one or more servers that you specify. Remember that that the encoded stream can be sent simultaneously to multiple servers, which will then replicate the stream to the client. Multiple AAC bitrates, formats, or processing parameters are supported from the same audio source by creating multiple AAC encoder instances.

Name

Enter a “friendly name” for this encoder instance to easily identify it for status and control from the main “control panel” screen (i.e. Kxxx-AAC Stream 1).

Audio Source

Select one of the previously deﬁned audio sources as the input to this processing instance (see Chapter 7). Note that these sources can be used in multiple processing or encoding instances simultaneously.

Processing

Selects between Omnia 3 and optional Omnia.9 processing (or no processing at all, should you simply want to route audio through this instance without processing). Up to 8 channels of Omnia.9 processing can be licensed per R/2. When you have reached the maximum number of Omnia.9 licenses available (or if Omnia.9 processing is not licensed) this option will not be displayed. See Chapter 16 for more information on Omnia.9 processing.

PROCESSING AND ENCODING

Processing preset

Selects the processing preset to be used for this instance. When Omnia 3 processing is selected, the list of available presets will appear. These presets are not format speciﬁc, though there are bitrate speciﬁc presets for “Music” and “Talk”. Experiment with various presets to ﬁnd one that works best for your programming format and application. Presets can be copied and modiﬁed to create a sound customized exactly to your needs. See Chapter 9 to learn how to “ﬁne tune” the Omnia 3 processing, and Chapter 13 for more information on Omnia 3 preset management.

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If optional Omnia.9 processing is enabled, the preset must be selected via NfRemote software due to the powerful processing tools and test instrumentation available through Omnia.9. See Chapter 16 for more details.

AAC Format

There are several diﬀerent “ﬂavors” of AAC. The most common form (and the default for this instance) is AAC-LC (Advanced Audio Coding-Low Complexity). HE-AAC is “High Eﬃciency” AAC. It is designed for bitrates between 24 and 96 kbps. An “improved” version of HE-AAC (HE-AAC v2) was developed for lower bitrates between 14 and 56 kbps. HE-AAC v2 also oﬀers improved stereo performance at lower bitrates. It is worth noting while HE-AAC v2 is widely supported, it may not be compatible with some players.

Note:

If in doubt, choose AAC-LC. This is the most common form of AAC encoding, though the HE (High Eciency) variants are becoming more common for lower bitrate applications.

PROCESSING AND ENCODING

Mono/Stereo

Select whether this stream will be encoded in Mono or Stereo. Mono should provide a mono sum of the incoming audio. This setting will be locked to stereo for HE-AAC v2.

Bitrate

Enter the bitrate for this stream. The valid bitrate range will change depending on which AAC format is selected.

Stream Output Format

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Selects the output ﬁle format for the stream. Even though the streams are not a “ﬁle” per-se, this setting controls how the AAC encoded audio is encapsulated. The default is generally suﬃcient for most applications, but other settings may be required for diﬀerent server platforms (such as Wowza or other RTMP server). Although multiple server destinations are supported per instance, only one output format can be selected. Use another instance if you need to send a diﬀerent format to another server (i.e. HTTP server and Wowza/RTMP simultaneously).

♦ RAW – Send unformatted (raw encoded) audio to your stream. Use this option with RTMP streams.

♦ ADIF - Audio Data Interchange Format, a ﬁle format to exchange AAC data.

♦ ADTS - Audio Data Transport Stream, a format, used by all other stream types.

♦ RAW+ESBRSIG – Send unformatted audio along to your stream that includes explicit backwards

compatible signaling method. This option can be used with RTMP streams.

PROCESSING AND ENCODING

Adding Streams

Multiple server destinations can be speciﬁed for each encoder instance. To add a destination, click on “Select stream type to add”. Supported server destinations include HTTP, Icecast, SHOUTcast, Wowza, or other RTMP (Adobe Flash Media Server, etc.)

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From the dropdown menu, select the server type that you wish to send an output from this encoder instance to.

HTTP Server

The HTTP option will send the output to the HTTP stream server built into R/2. Using this server, you can monitor the encoded stream directly from R/2. This is often helpful in troubleshooting server connections, or just as a quick way to test that everything is working as expected. The HTTP server is also useful when setting up audio process-

PROCESSING AND ENCODING

ing, as the delay is much shorter than that experienced with an external server.

Friendly Name

Specify an easy to remember “Friendly” name to identify this stream output in the list of streams that have been deﬁned (i.e. Kxxx – Internal Test Stream).

Stream resource ID

This resource ID will become part of the URL that you use to reach this stream. For example, if the stream server is active on port 8888, and you use a resource ID of “TestStream” then you would listen to this stream by entering http://ip_address_of_r2:8888/TestStream in your browser or media player.

Full stream URL

Displays the full URL of this HTTP stream (including the port and Stream resource ID).

Note:

As of this writing, R/2 currently displays “localhost” rather than the actual IP address of R/2. Replace “localhost” in the URL with the IP address or hostname of R/2.

When ﬁnished, click “Save” to save and return to the “AAC Encoder” screen or click “Cancel” to cancel any changes.

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Send to ICEcast Server

The ICEcast Server option allows sending a stream to an ICEcast (or compatible) server. ICEcast is an open source streaming server available as a free download for various platforms. A separate option is provided for SHOUTcast compatible servers.

Friendly name

Specify an easy to remember “Friendly” name to identify this stream output in the list of streams that have been deﬁned (i.e. Kxxx – Icecast Stream).

PROCESSING AND ENCODING

Metadata source

Select a metadata input source to be used with this encoding instance (if desired). See Chapter 10 for more details.

Server address

Enter the IP address and port of the ICEcast server.

Mount point

Each ICEcast server may contain multiple broadcasts (or mount points) each containing a separate stream of content. Thus, ‘mount point’ is just a unique name that identiﬁes a particular stream. The mount point needs to be conﬁgured properly in order for the stream to work. Mount points should not contain any spaces or odd characters.

Username

Enter a user name here for access to the audio stream. This should be the same username conﬁgured on your ICEcast server for sources. The default username is “source”.

Password

Enter the password conﬁgured on your ICEcast server for encoding sources.

(You DID change it, didn’t you? Please tell me it isn’t still set to “hackme”)

Admin user

Enter the “admin” username for your ICEcast server. The default is “admin”.

This user will be used for the metadata functions of ICEcast.

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Admin password

Enter the administrator password conﬁgured on your ICEcast server here.

(Again, I REALLY hope it isn’t still set to “hackme”)

Stream name

Enter a name for your stream. This will be used for stream directory listings (if enabled) and displayed in the stream metadata.

Genre

Enter the format of your stream or station here, for example rock, country or urban. ICEcast will display this information in the metadata ﬁeld.

Stream URL

Enter a URL to be displayed with the stream metadata. This can be a station’s website, or other URL associated with the stream.

Check the Public checkbox to list this stream in public directories, or leave it un-checked if you do not want the stream listed.

Note:

Even with the “Public” checkbox un-checked, it may still be possible for stream “aggregators” to nd and list your stream as they crawl the Internet searching for publicly available audio streams. There are very few practical ways to avoid this, unfortunately.

When ﬁnished, click “Save” to save and return to the “AAC Encoder” screen or click “Cancel” to cancel any changes.

PROCESSING AND ENCODING

Send to SHOUTcast Server

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SHOUTcast DNAS (Distributed Network Audio Software) is a media server developed by Nullsoft, now distributed by Radionomy. SHOUTcast is available in multiple levels from from a basic (free) version, through paid options that allow monetization. Currently it is available for Windows, Linux and Mac OSX. Cloud-hosted options are also available.

R/2 replaces the DSP encoding plugin for Winamp normally used for broadcasting to a SHOUTcast server.

Friendly name

Specify an easy to remember “Friendly” name to identify this stream output in the list of streams that have been deﬁned (i.e. Kxxx – SHOUTcast Stream).

Metadata source

Select a metadata input source to be used with this encoding instance (if desired). See Chapter 10 for more details.

Server address

Enter the IP address and port of the SHOUTcast server.

Password

Enter the password conﬁgured on your SHOUTcast server for encoding sources.

Stream name

Enter a name for your stream. This will be used for stream directory listings (if enabled) and displayed in the stream metadata.

Genre

Enter the format of your stream or station here, for example rock, country or urban. SHOUTcast will display this information in the metadata ﬁeld.

Stream URL

Enter a URL to be displayed with the stream metadata. This can be a station’s website, or other URL associated with the stream.

PROCESSING AND ENCODING

Check the Public checkbox to list this stream in public directories, or leave it un-checked if you do not want the stream listed.

Note:

Even with the “Public” checkbox un-checked, it may still be possible for stream “aggregators” to nd and list your stream as they crawl the Internet searching for publicly available audio streams. There are very few practical ways to avoid this, unfortunately.

Protocol Version

Choose between SHOUTcast Version 1 or Version 2, depending on the requirements of your SHOUTcast server platform.

Stream ID

Specify the numerical identiﬁer of the stream for control or referencing the stream conﬁguration. This can only be a numeric value from 1 to 2147483647.

When ﬁnished, click “Save” to save and return to the “AAC Encoder” screen or click “Cancel” to cancel any changes.

Send to Wowza server

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R/2 oﬀers direct integration with Wowza servers for easy conﬁguration of encoding parameters. Conﬁguration ﬁles can be loaded into R/2 as an XML ﬁle downloaded from the Wowza server, XML conﬁguration copy/paste, or manually conﬁgured.

Note:

In order to send AAC encoded audio to a Wowza server, you must use “Raw” for the stream le output format, as described earlier in this chapter. Other stream le formats cannot be used simultaneously in the same instance (i.e. ADTS for the internal HTTP server or other servers). Create a new instance if you wish to use another stream le format simultaneously.

Although it can be disabled, Wowza is set by default to use authentication for RTMP encoding sources such as R/2, and it is highly recommended. You will need to deﬁne at least one valid “Publisher” account in the “Server” setup for Wowza.

In the Wowza Streaming Engine Manager console, Navigate to “Server”, then to “Source Authentication”. Click on “Add Source” and specify a username and password. This is necessary for both XML-based and manual conﬁguration of a Wowza stream in R/2.

PROCESSING AND ENCODING

Conﬁguring R/2 for a Wowza stream

Click on the “Options” dropdown to choose a method for conﬁguring a Wowza connection.

Load as XML

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Allows loading an XML conﬁguration for a Wowza connection either from a ﬁle, or by copy/paste.

This ﬁle can be downloaded from the Wowza server by clicking on the desired streaming application in Wowza (live, for example) then clicking on “Incoming Publishers”. Select “The Telos Alliance” from the list of available encoder manufacturers. The appropriate conﬁguration parameters will appear, along with a button to download an XML ﬁle containing the conﬁguration for R/2. You will also need to deﬁne an account for the incoming source.

Friendly name

Specify an easy to remember “Friendly” name to identify this stream output in the list of streams that have been deﬁned (i.e. Kxxx – Wowza Stream).

Metadata source

Select a metadata input source to be used with this encoding instance (if desired). See Chapter 10 for more details.

XML data

Copy and paste the XML conﬁguration data for this connection from the Wowza server here.

PROCESSING AND ENCODING

Load from ﬁle

Click here to load Wowza XML conﬁguration data from a ﬁle.

When ﬁnished, click “Save” to save and return to the “AAC Encoder” screen or click “Cancel” to cancel any changes.

Send to Wowza Server

Allows manual conﬁguration of a stream pushed to a Wowza server.

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Friendly name

Specify an easy to remember “Friendly” name to identify this stream output in the list of streams that have been deﬁned (i.e. Kxxx – Wowza Stream).

Metadata source

Select a metadata input source to be used with this encoding instance (if desired). See Chapter 10 for more details.

Server address

Enter the IP address of the Wowza server.

Server port

Enter the port of the Wowza server. The default is 1935.

Publish path

Enter the path to the desired streaming application on your Wowza server (live, for example).

Stream

Enter the name of the stream you wish to create on your Wowza server. This stream will be created under the speciﬁed path (application).

Use Authentication

Check this box to use authentication when sending a stream to the Wowza server. Authentication for RTMP publisher sources such as R/2 is enabled by default in Wowza and will likely need to be enabled here as well.

Username

Enter the username speciﬁed as a publisher in the Wowza server.

Password

Enter the password associated with the publisher username in the Wowza server.

PROCESSING AND ENCODING

When ﬁnished, click “Save” to save and return to the “AAC Encoder” screen or click “Cancel” to cancel any changes.

Send to RTMP Server

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RTMP (Real Time Messaging Protocol) is a TCP-based protocol initially developed by Macromedia to deliver streams to the Flash player. The speciﬁcation is now available (albeit in incomplete form) for public use. A number of servers support the RTMP protocol, including Adobe Flash Media Server, Wowza and others.

Note:

In order to send AAC encoded audio to an RTMP server, you must use “Raw” for the stream le output format, as described earlier in this chapter. Other stream le formats cannot be used simultaneously in the same instance (i.e. ADTS for the internal HTTP server or other servers). Create a new instance if you wish to use another stream le format simultaneously.

Friendly name

Specify an easy to remember “Friendly” name to identify this stream output in the list of streams that have been deﬁned (i.e. Kxxx – RTMP Stream).

Metadata source

Select a metadata input source to be used with this encoding instance (if desired). See Chapter 10 for more details.

Server address

Enter the IP address of the RTMP server.

Server port

Enter the port of the RTMP server. The default is 1935.

RTMP Publish path

Enter the publishing path for the RTMP server—generally provided by your CDN, and may also be referred to as an “entry point”.

RTMP Stream

Enter the name of the RTMP stream name. The stream name may be ﬁxed or chosen, it depends on your CDN.

PROCESSING AND ENCODING

Some providers, such as Akamai, require the stream name to be very speciﬁc, and they provide the format that must be used.

Use RTMP Authentication

Check this box to use authentication when sending a stream to the RTMP server.

Username

Enter the username for encoding to your RTMP server.

Password

Enter the password for encoding to your RTMP server.

When ﬁnished, click “Save” to save and return to the “AAC Encoder” screen or click “Cancel” to cancel any changes.

MP3 Encoder Instance

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An MP3 Encoder instance accepts data from an audio source, processes it, encodes it to MPEG Layer-3, and makes it available to media servers. The encoded stream can also be made available through the HTTP stream server built into the application. The internal HTTP server is intended for monitoring the stream by a very small number of listeners. It is not designed to feed a large number of clients.

Name

Enter a “friendly name” for this encoder instance to easily identify it for status and control from the main “control panel” screen (i.e. Kxxx-MP3 Stream 1).

Audio Source

Select one of the previously deﬁned audio sources as the input to this processing instance (see Chapter 7). Note that these sources can be used in multiple processing or encoding instances simultaneously.

PROCESSING AND ENCODING

Processing

Selects between Omnia 3 and optional Omnia.9 processing (or no processing at all, should you simply want to route audio through this instance without processing). Up to 8 channels of Omnia.9 processing can be licensed per R/2. When you have reached the maximum number of Omnia.9 licenses available (or if Omnia.9 processing is not licensed) this option will not be displayed. See Chapter 16 for more information on Omnia.9 processing.

Processing preset

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Selects the processing preset to be used for this instance. When Omnia 3 processing is selected, the list of available presets will appear. These presets are not format speciﬁc, though there are bitrate speciﬁc presets for “Music” and “Talk”. Experiment with various presets to ﬁnd one that works best for your programming format and application. Presets can be copied and modiﬁed to create a sound customized exactly to your needs. See Chapter 9 to learn how to “ﬁne tune” the Omnia 3 processing, and Chapter 13 for more information on Omnia 3 preset management.

If optional Omnia.9 processing is enabled, the preset must be selected via NfRemote software due to the powerful processing tools and test instrumentation available through Omnia.9. See Chapter 16 for more details.

MP3 Format

Select whether this stream will be encoded in Mono or Stereo. Mono should provide a mono sum of the incoming audio.

PROCESSING AND ENCODING

Bitrate

Enter the bitrate for this stream. Valid MP3 bitrates are 20 kbps to 320 kbps.

Adding Streams

Multiple server destinations can be speciﬁed for each encoder instance. To add a destination, click on “Select stream type to add”. Supported server destinations include HTTP, Icecast, SHOUTcast, Wowza, or other RTMP (Adobe Flash Media Server, etc.)

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From the dropdown menu, select the server type that you wish to send an output from this encoder instance to.

HTTP Server

The HTTP option will send the output to the HTTP stream server built into R/2. Using this server, you can monitor the encoded stream directly from R/2. This is often helpful in troubleshooting server connections, or just as a quick way to test that everything is working as expected. The HTTP server is also useful when setting up audio processing, as the delay is much shorter than that experienced with an external server.

PROCESSING AND ENCODING

Friendly Name

Specify an easy to remember “Friendly” name to identify this stream output in the list of streams that have been deﬁned (i.e. Kxxx – Internal Test Stream).

Stream resource ID

This resource ID will become part of the URL that you use to reach this stream. For example, if the stream server is active on port 8888, and you use a resource ID of “TestStream” then you would listen to this stream by entering http://ip_address_of_r2:8888/TestStream in your browser or media player.

Full stream URL

Displays the full URL of this HTTP stream (including the port and Stream resource ID).

Note:

As of this writing, R/2 currently displays “localhost” rather than the actual IP address of R/2. Replace “localhost” in the URL with the IP address or hostname of R/2.

When ﬁnished, click “Save” to save and return to the “MP3 Encoder” screen or click “Cancel” to cancel any changes.

Send to ICEcast Server

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The ICECast Server option allows sending a stream to an ICEcast (or compatible) server. ICECast is an open source streaming server available as a free download for various platforms. A separate option is provided for SHOUTcast compatible servers.

PROCESSING AND ENCODING

Friendly name

Specify an easy to remember “Friendly” name to identify this stream output in the list of streams that have been deﬁned (i.e. Kxxx – Icecast Stream).

Metadata source

Select a metadata input source to be used with this encoding instance (if desired). See Chapter 10 for more details.

Server address

Enter the IP address and port of the ICEcast server.

Mount point

Each ICEcast server may contain multiple broadcasts (or mount points) each containing a separate stream of content. Thus, ‘mount point’ is just a unique name that identiﬁes a particular stream. The mount point needs to be conﬁgured properly in order for the stream to work. Mount points should not contain any spaces or odd characters.

Username

Enter a user name here for access to the audio stream. This should be the same username conﬁgured on your ICECast server for sources. The default username is “source”.

Password

Enter the password conﬁgured on your ICECast server for encoding sources.

(You DID change it, didn’t you? Please tell me it isn’t still set to “hackme”)

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Admin user

Enter the “admin” username for your ICECast server. The default is “admin”.

This user will be used for the metadata functions of ICEcast.

Admin password

Enter the administrator password conﬁgured on your ICECast server here.

(Again, I REALLY hope it isn’t still set to “hackme”)

Stream name

Enter a name for your stream. This will be used for stream directory listings (if enabled) and displayed in the stream metadata.

Genre

Enter the format of your stream or station here, for example rock, country or urban. ICECast will display this information in the metadata ﬁeld.

Stream URL

Enter a URL to be displayed with the stream metadata. This can be a station’s website, or other URL associated with the stream.

Check the Public checkbox to list this stream in public directories, or leave it un-checked if you do not want the stream listed.

Note:

Even with the “Public” checkbox un-checked, it may still be possible for stream “aggregators” to nd and list your stream as they crawl the Internet searching for publicly available audio streams. There are very few practical ways to avoid this, unfortunately.

PROCESSING AND ENCODING

When ﬁnished, click “Save” to save and return to the “MP3 Encoder” screen or click “Cancel” to cancel any changes.

Send to SHOUTcast Server

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SHOUTcast DNAS (Distributed Network Audio Software) is a media server developed by Nullsoft, now distributed by Radionomy. SHOUTcast is available in multiple levels from from a basic (free) version through paid options that allow monetization. Currently it is available for Windows, Linux and Mac OSX. Cloud-hosted options are also available.

R/2 replaces the DSP encoding plugin for Winamp normally used for broadcasting to a SHOUTcast server.

Friendly name

Specify an easy to remember “Friendly” name to identify this stream output in the list of streams that have been deﬁned (i.e. Kxxx – SHOUTcast Stream).

Metadata source

Select a metadata input source to be used with this encoding instance (if desired). See Chapter 10 for more details.

Server address

Enter the IP address and port of the SHOUTcast server.

Password

Enter the password conﬁgured on your SHOUTcast server for encoding sources.

Stream name

Enter a name for your stream. This will be used for stream directory listings (if enabled) and displayed in the stream metadata.

Genre

Enter the format of your stream or station here, for example rock, country or urban. SHOUTcast will display this information in the metadata ﬁeld.

Stream URL

Enter a URL to be displayed with the stream metadata. This can be a station’s website, or other URL associated with the stream.

PROCESSING AND ENCODING

Check the Public checkbox to list this stream in public directories, or leave it un-checked if you do not want the stream listed.

Note:

Even with the “Public” checkbox un-checked, it may still be possible for stream “aggregators” to nd and list your stream as they crawl the Internet searching for publicly available audio streams. There are very few practical ways to avoid this, unfortunately.

Protocol Version

Choose between SHOUTcast Version 1 or Version 2, depending on the requirements of your SHOUTcast server platform.

Stream ID

Specify the numerical identiﬁer of the stream for control or referencing the stream conﬁguration. This can only be a numeric value from 1 to 2147483647.

When ﬁnished, click “Save” to save and return to the “MP3 Encoder” screen or click “Cancel” to cancel any changes.

Send to Wowza server

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R/2 oﬀers direct integration with Wowza servers for easy conﬁguration of encoding parameters. Conﬁguration ﬁles can be loaded into R/2 as an XML ﬁle downloaded from the Wowza server, XML conﬁguration copy/paste, or manually conﬁguration.

Although it can be disabled, Wowza is set by default to use authentication for RTMP encoding sources such as R/2, and it is highly recommended. You will need to deﬁne at least one valid “Publisher” account in the “Server” setup for Wowza.

In the Wowza Streaming Engine Manager console, Navigate to “Server”, then to “Source Authentication”. Click on “Add Source” and specify a username and password. This is necessary for both XML-based and manual conﬁguration of a Wowza stream in R/2.

Conﬁguring R/2 for a Wowza stream

Click on the “Options” dropdown to choose a method for conﬁguring a Wowza connection.

PROCESSING AND ENCODING

Load as XML

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Allows loading an XML conﬁguration for a Wowza connection either from a ﬁle, or by copy/paste.

This ﬁle can be downloaded from the Wowza server by clicking on the desired streaming application in Wowza (live, for example) then clicking on “Incoming Publishers”. Select “The Telos Alliance” from the list of available encoder manufacturers. The appropriate conﬁguration parameters will appear, along with a button to download an XML ﬁle containing the conﬁguration for R/2. You will also need to deﬁne an account for the incoming source.

Friendly name

Specify an easy to remember “Friendly” name to identify this stream output in the list of streams that have been deﬁned (i.e. Kxxx – Wowza Stream).

Metadata source

Select a metadata input source to be used with this encoding instance (if desired). See Chapter 10 for more details.

XML data

Copy and paste the XML conﬁguration data for this connection from the Wowza server here.

Load from ﬁle

Click here to load Wowza XML conﬁguration data from a ﬁle.

When ﬁnished, click “Save” to save and return to the “MP3 Encoder” screen or click “Cancel” to cancel any changes.

PROCESSING AND ENCODING

Send to Wowza Server

Allows manual conﬁguration of a stream pushed to a Wowza server.

Friendly name

Specify an easy to remember “Friendly” name to identify this stream output in the list of streams that have been deﬁned (i.e. Kxxx – Wowza Stream).

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Metadata source

Select a metadata input source to be used with this encoding instance (if desired). See Chapter 10 for more details.

Server address

Enter the IP address of the Wowza server.

Server port

Enter the port of the Wowza server. The default is 1935.

Publish path

Enter the path to the desired streaming application on your Wowza server (live, for example).

Stream

Enter the name of the stream you wish to create on your Wowza server. This stream will be created under the speciﬁed path (application).

Use Authentication

Check this box to use authentication when sending a stream to the Wowza server. Authentication for RTMP publisher sources such as R/2 is enabled by default in Wowza and will likely need to be enabled here as well.

Username

Enter the username speciﬁed as a publisher in the Wowza server.

Password

Enter the password associated with the publisher username in the Wowza server.

When ﬁnished, click “Save” to save and return to the “MP3 Encoder” screen or click “Cancel” to cancel any changes.

PROCESSING AND ENCODING

Send to RTMP Server

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RTMP (Real Time Messaging Protocol) is a TCP-based protocol initially developed by Macromedia to deliver streams to the Flash player. The speciﬁcation is now available (albeit in incomplete form) for public use. A number of servers support the RTMP protocol, including Adobe Flash Media Server, Wowza and others.

Friendly name

Specify an easy to remember “Friendly” name to identify this stream output in the list of streams that have been deﬁned (i.e. Kxxx – RTMP Stream).

Metadata source

Select a metadata input source to be used with this encoding instance (if desired). See Chapter 10 for more details.

Server address

Enter the IP address of the RTMP server.

Server port

Enter the port of the RTMP server. The default is 1935.

RTMP Publish path

Enter the publishing path for the RTMP server—generally provided by your CDN, and may also be referred to as an “entry point”.

RTMP Stream

Enter the name of the RTMP stream name. The stream name may be ﬁxed or chosen, it depends on your CDN. Some providers, such as Akamai, require the stream name to be very speciﬁc, and they provide the format that must be used.

Use RTMP Authentication

Check this box to use authentication when sending a stream to the RTMP server.

Username

Enter the username for encoding to your RTMP server.

PROCESSING AND ENCODING

Password

Enter the password for encoding to your RTMP server.

When ﬁnished, click “Save” to save and return to the “MP3 Encoder” screen or click “Cancel” to cancel any changes.

Adaptive AAC Encoder

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As more audio content is being streamed to mobile devices, adaptive encoders take on increased importance. An adaptive encoder sends an audio stream in multiple bitrates to the server. The client device then automatically chooses the best one based on available bandwidth. As more or less bandwidth becomes available, the mobile device may request higher or lower bitrate audio, so it is able to adapt to network conditions dynamically.

Name

Enter a “friendly name” for this encoder instance to easily identify it for status and control from the main “control panel” screen (i.e. Kxxx-Adaptive Stream 1).

Audio Source

Select one of the previously deﬁned audio sources as the input to this processing instance (see Chapter 7). Note that these sources can be used in multiple processing or encoding instances simultaneously.

Processing

Selects between Omnia 3 and optional Omnia.9 processing (or no processing at all, should you simply want to route audio through this instance without processing). Up to 8 channels of Omnia.9 processing can be licensed per R/2. When you have reached the maximum number of Omnia.9 licenses available (or if Omnia.9 processing is not licensed) this option will not be displayed. See Chapter 16 for more information on Omnia.9 processing.

PROCESSING AND ENCODING

Processing preset

Selects the processing preset to be used for this instance. When Omnia 3 processing is selected, the list of available presets will appear. These presets are not format speciﬁc, though there are bitrate speciﬁc presets for “Music” and “Talk”. Experiment with various presets to ﬁnd one that works best for your programming format and application. Presets can be copied and modiﬁed to create a sound customized exactly to your needs. See Chapter 9 to learn how to “ﬁne tune” the Omnia 3 processing, and Chapter 13 for more information on Omnia 3 preset management.

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If optional Omnia.9 processing is enabled, the preset must be selected via NfRemote software due to the powerful processing tools and test instrumentation available through Omnia.9. See Chapter 16 for more details.

Fragment duration (milliseconds)

Conﬁgures the fragment length. Adaptive streaming splits audio into fragments which are then (usually by HTTP) downloaded by the client device. The longer the fragments, the more latency in the stream as the player will need to keep at least one fragment buﬀered before downloading the next one

Synchronize stream start to fragment duration aligned steps

This setting will synchronize fragment start between primary and backup servers. This option should be used together with an RTP source using absolute PTP time so adaptive streams will be started only on fragment length boundaries and fragments from primary and backup servers will be synchronized. If only one encoder server is being used, then this option should be left unchecked.

Transport Format

Select the desired transport format for this Adaptive AAC Encoding Instance. This will depend on the server platform and application. Multiple streams and bitrates can be deﬁned in a single Adaptive AAC instance as necessary.

PROCESSING AND ENCODING

Note:

ADIF and ADTS CRC should NOT be used with Smooth Streaming or HLS unless there are very specic server requirements.

RAW (for Smooth Streaming)

Smooth Streaming is an IIS Media Services extension that facilitates adaptive streaming to clients via HTTP. SDKs are available from Microsoft to enable Smooth Streaming over Silverlight and Windows Phone 7. It may also be ported to Apple iOS, Android and Linux

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Friendly name

Specify an easy to remember “Friendly” name to identify this stream output in the list of streams that have been deﬁned (i.e. Kxxx – Adaptive Stream).

Metadata source

Metadata is not yet available for Smooth Streaming but should be available for custom CDN servers in the future.

Stream Name

If multiple streams use the same mount point, then Stream Name can be used to separate them. The default Stream Name is “Encoder”.

Mount point

Each Smooth Streaming server may contain multiple broadcasts (or mount points) each containing a separate stream of content. Thus, ‘mount point’ is just a unique name that identiﬁes a particular stream. The mount point needs to be conﬁgured properly in order for the stream to work. Mount point names should not contain any spaces or odd characters.

When ﬁnished, click “Save” to save and return to the “Adaptive AAC Encoder” screen or click “Cancel” to cancel any changes.

PROCESSING AND ENCODING

ADTS (for HLS)

Apple’s HTTP Live Streaming protocol allows you to create multiple bitrate adaptive streams, which allow the client’s device to request the needed bitrate stream depending on the available network bandwidth.

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Friendly name

Specify an easy to remember “Friendly” name to identify this stream output in the list of streams that have been deﬁned (i.e. Kxxx – HLS Stream).

Metadata source

Select a metadata input source to be used with this encoding instance (if desired). See Chapter 10 for more details.

Stream name

This name will be used in HLS ﬁlenames and folders. Please use only letters, numbers, dashes and underscores.

Storage

Select “Local Folder” or “FTP Server”. The “Local Folder” setting will make ﬁles available via an HTTP share on R/2 in the format: http://r2address/streams/HLS/streamname/streamname.m3u8

Destination

When set to “FTP Server”, this ﬁeld speciﬁes the FTP server where HLS ﬁle chunks should be uploaded. This ﬁeld requires a full FTP path i.e. “ftp://user:password@hostname:21/folder/” to specify a username, password, and port, or “ftp://hostname/folder/” if no authentication or port number is needed. If the HLS stream is transferred to an FTP server, the server must have its own web service to serve the ﬁles to the listeners.

Delete existing ﬁles

Removes all ﬁles from the folder or FTP server when the stream is started. Use this option with caution as the ﬁles will be permanently deleted!

Archive size

Set number of fragments that need to be stored. The old fragments will be deleted. If this variable is set to 0, all the fragments will be stored.

When ﬁnished, click “Save” to save and return to the “Adaptive AAC Encoder” screen or click “Cancel” to cancel any changes.

PROCESSING AND ENCODING

Conﬁguring Available Bitrates

In addition to deﬁning the other adaptive streaming parameters, you will need to conﬁgure the bitrates and AAC encoding formats contained in those streams.

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Click “Add bitrate” to add an entry to the list.

AAC Format

There are several diﬀerent “ﬂavors” of AAC. The most common form (and the default for this instance) is AAC-LC (Advanced Audio Coding-Low Complexity). HE-AAC is “High Eﬃciency” AAC. It is designed for bitrates between 24 and 96 kbps. An “improved” version of HE-AAC (HE-AAC v2) was developed for lower bitrates between 14 and 56 kbps. HE-AAC v2 also oﬀers improved stereo performance at lower bitrates. It is worth noting while HE-AAC v2 is widely supported, it may not be compatible with some players.

Note:

If in doubt, choose AAC-LC. This is the most common form of AAC encoding, though the HE (High Eciency) variants are becoming more common for lower bitrate applications.

PROCESSING AND ENCODING

Channels

Select whether this stream will be encoded in Mono or Stereo. Mono should provide a mono sum of the incoming audio. This setting will be locked to stereo for HE-AAC v2.

Bitrate

Enter the bitrate for this stream. The valid bitrate range will change depending on which AAC format is selected.

When ﬁnished, click “Save” to save and return to the “Adaptive AAC Encoder” screen or click “Cancel” to cancel any changes.

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Managing Processing and Encoding Instances

PROCESSING AND ENCODING

Click “Main” or the Z/IPStream R/2 logo to return to the main “Control Panel” screen. All processing and encoding instances that have been deﬁned will be displayed, along with their conﬁgured parameters and status. Click the “triangle” on the “Status” line to display additional details about the individual streams contained within each instance.

Instance Type

The oval at the top right of each instance will display the type of instance that has been deﬁned.

Indicates an audio processing instance

Indicates a standard AAC encoding instance

Indicates an MP3 encoding instance

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Indicates a multiple bitrate (adaptive) AAC encoding instance

Instance Controls

Edit the parameters of an instance. Some parameters may not be changed once an instance has been deﬁned, or while an instance is running.

Delete an instance. This is a permanent action. Deleted instances cannot be recovered and will need to be re-created.

Starts an instance that is currently stopped. Instances that are not currently running will be “greyed out”.

PROCESSING AND ENCODING

Stops an instance that is currently started.

Metering and Processing Control

I/O Metering is provided for each instance. Audio processing presets for each instance can be modiﬁed by clicking the “pencil” icon next to the preset name (see Chapter 9 for more details on Omnia 3 processing). Instances using Omnia.9 processing can be accessed via NfRemote for more detailed monitoring and control of the audio processing (see Chapter 16).

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9 Omnia 3 Processing Setup

Overview

Standard audio processing for Z/IPStream R/2 is the Omnia 3-band processor. Omnia 3 is intuitive and easy to use. Applying audio will activate movement on the bar graphs, and this is conﬁrmation that processing is occurring.

All audio processing adjustments are performed in the Preset Editor. A “preset” is simply a large table of values representing all of the control values for every control listed in the menu tree. When you choose a preset, the control values are loaded into the processing structure, reconﬁguring Omnia 3.

We recommend exploring the provided factory presets. Then, once you ﬁnd the preset that most closely matches the sound you’re after, you can customize it to your liking. More information about Omnia 3 preset management can be found in Chapter 13 of this manual.

Understanding the Omnia 3 Bar Graphs and Indicators

The bar graphs are capable of indicating more than just level information. The texture and density of the audio signal can be observed, based upon the dynamic action of the bar graphs, and peak-responding “pills”. Of interest are the “pills” at the end of the input and output meters, as they indicate peak level. The bar section represents the RMS average of the signal. Wide dynamic range will display a separation between the pills and the bar, whereas signal with little peak information will cause the pills to ride on the crest of the bar graph. The bar graphs can indicate up to 25 dB of gain reduction.

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The bar graphs provide a wealth of information about the processing control in Omnia 3. The algorithms automatically adapt the style of compression/limiting control being employed on a moment-by-moment basis. This can be deduced if the metering is studied over time. During normal operation, the indicators will have a dynamic “bounce” that you will be able to get a feel for. Every now and then, you will see the processor react quickly and show a larger amount of gain reduction. Notice that the response is very fast. This action will recover very slowly, and return to “rest” with the main bar graph. This action will be easily noticed on material that is very dynamic in texture. More on this below.

Another feature unique to Omnia 3 is processor “Hold”. During brief pauses in audio, the bar graphs will “freeze” and the GATE indicators will show the processor has entered the Hold mode of the algorithm. This is most easily noticed when there is a “dry” voice being processed.

The Input/Output level meters will change from green to red whenever the level exceeds –12dBfs. The pills also change color on the Input/Output displays. In the dynamic processing section, the pills maintain the same color.

OMNIA 3 PROCESSING SETUP

Note:

The input and output levels are displayed relative to 0dB full scale (0dBfs). The 0 indicator on the input/output bar graphs means that every available bit of signal level is being used at that time! There is nothing more in the level department, except to create distortion… nasty sounding distortion!

Interpreting the Gain Reduction Meter Displays

Through careful observation of the processing bar graphs, signiﬁcant information can be acquired and analyzed about the audio signal on a moment-by-moment basis. The bar shows the average value of the gain reduction, while the ﬂoating pill indicates the peak value of gain reduction.

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Since the processing algorithms in Omnia 3 ‘adapt’ to the audio, it is capable of performing many diﬀerent processing functions at diﬀerent times. The processing display bar graphs may appear to indicate diﬀerent kinds of operation with diﬀerent program material. This behavior is primarily based on dynamic range diﬀerences in the applied program material. For program material that is already processed or lacking dynamic range, the bar graphs will indicate diﬀerently than with material that possesses a high degree of dynamic range. This is because audio signals that lack dynamic range, naturally or by previous processing, will possess a lower peak to average ratio. Conversely, audio signals with a wide dynamic range possess a higher peak to average ratio. The Omnia 3 adapts diﬀerently to each case.

For program material that has low dynamic range (or high RMS and low peak levels), there will be more activity in the WB-AGC sections and less activity in the Multiband Processors. This behavior is due to the WB-AGC response to high RMS energy, while the Multiband section is reacting to lower peak energy. Sometimes the Multiband section may not indicate any action at all. For Omnia 3, this is completely normal! Examples of this behavior might be seen with heavily processed commercials or music, or with music passages of sustained level. A good example is the Aerosmith album “Pump.” This recording has very little dynamic range. Try any up-tempo selection from this recording and you will see the Wideband-AGC section make initial adjustments along with small amounts of limiting. Once adjusted, the Wideband-AGC section will stay deep in gain reduction, although with little movement, and action within the limiters will be minimal.

When the programming has wide dynamic range (low RMS and high peak levels), the opposite will occur. The Multiband section becomes active, while the WB-AGC section will appear not to respond as much. During these events, the Multiband sections could be working aggressively, while the WB-AGC section indicates little activity. Good examples of programming able to cause this behavior: vocal passages, live voice, classical music, and passages with high transient levels. Good test examples: almost anything classical, as well as almost anything by Steely Dan!

Also keep in mind that the WB-AGC section is designed to operate much more slowly than the Multiband sections, primarily because of the nature of each function. Remember that the WB-AGC section operates on the program’s RMS energy. During gain calculations, the incoming program’s “average” level is established, and gain adjustments, if needed, are made based on those calculations. This is why the WB-AGC sections will appear to move slower as it makes changes over relatively long time periods.

OMNIA 3 PROCESSING SETUP

The intent of the Multiband section is to normalize the spectral balance and provide control of the peak levels. Peak energy must be detected and adjusted in a quick and accurate manner while, at the same time, not interfering with the sonic integrity of the audio signal. For this reason, the Multiband sections operate faster, with special background instructions to govern their behavior, and strictly on an as-needed basis.

Since the processing displays are capable of providing a wide range of information, we do not recommend setting up the system based on any speciﬁed meter indications. Instead, we recommend setting up the processing by using your ears to judge the sound. We’ve provided the meters only to analyze the signals and aid you in adjusting the speciﬁc parameters needed to achieve your desired sonic results. Like a speedometer, the meters are a guide, not the road!

The red ticks that appear above the WB (Wideband AGC), L (Low-Band AGC), M (Mid- Band AGC), and H (High-Band AGC) bar graphs are the GATE indicators. They display GATE activity. The GATE function is described above in the section titled “Understanding the Bar graphs and Indicators”

Editing Presets

Once a at least processing or processing/encoding instance has been deﬁned, you are ready to begin the editing process. From the main “Control Panel” screen, click on the pencil icon to the right of the preset name in the desired instance, and you will be presented with a block diagram of the Omnia 3’s audio path. Clicking on any of the blocks takes you to the controls available for that section. Once you’ve made your adjustments, click Save or Cancel, depending on whether or not you like what you hear.

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Input

Clicking on the Input block displays the input section sub-menu. There are adjustments for the individual Left/ Right channels. Drag the slider-bar to adjust the respective level. There is a 30dB range of adjustment between –20dB and +10dB.

Using normal program audio, a correct gain setting results in peak indications (the“bouncing balls”) hitting -12 dBFS (where the level meters turn red) or a little higher. This corresponds to system headroom of 12 dB. You may adjust for more headroom if you wish (lower indications), but setting the input level for less headroom (higher meter indications) is strongly frowned upon.

OMNIA 3 PROCESSING SETUP

Wide Band AGC

A very ﬂexible wideband leveler section provides smooth, transparent control of the input program. This is achieved through two signiﬁcant Omnia innovations: a dual referenced release gate and a hidden, intelligent “makeup” gain algorithm. The dual gate reference is a unique process that correlates the dynamics of the audio input signal to a “rolling” reference level, and from that information makes conditional decisions aﬀecting the character of the release function.

A hidden, Make-Up Gain, control signal determines when the amplitude of the input program suddenly falls to a reduced level. It then adjusts the side chain gain in order to “ﬁll in” the softer program passages so that the average level is increased. This allows the AGC function to operate with slower time constants, while signiﬁcantly increasing the average audio level. These slower overall time constants yield lower intermodulation distortion, contributing to Omnia’s trademark sound.

Using Classical music for an example, the orchestra often plays forte, and then enters abruptly into a quiet passage. Conventional AGC algorithms would hold the softer passage down until it was able to slowly recover at the static release time setting, making such passages nearly inaudible at normal listening levels. Omnia 3’s makeup gain function allows a hidden, faster time constant to provide quick recovery, but only during the softer passage. As soon as the orchestra starts to play louder, the “makeup” time constant yields control back to the primary AGC circuit, returning gain to the previous platform level. This sophistication preserves the dynamic integrity of the signal while greatly enhancing the listenability of the program.

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This is the ﬁrst processing section within the Omnia 3, so the controls in this section aﬀect the overall signal, its density, and hence aﬀect every other processing section following it. The job of the Wideband AGC is to erase long-term audio level ﬂuctuations, while doing so in a very unobtrusive manner. This sub-menu provides a few parameters that tailor the action of this section.

OMNIA 3 PROCESSING SETUP

Drive

The sets the audio level that enters the WB-AGC. Increasing the Drive will produce more compression. This control is calibrated between –6dB and +6dB, and adjusts the signal level into the Wideband AGC. It should be adjusted to net approximately 12dB of gain reduction with typical programming. Too little gain reduction can defeat the “leveling” function of the Wideband AGC. Too much gain reduction has little additional beneﬁt. Nominal gain reduction values for the Wideband AGC are between 10 and 15dB.

Gate

When set to ON, it will freeze the gain at the last level of processing action that occurred before the audio signal fell below the threshold of operation. This control helps to minimize “pumping”, and the increase of background noise during pauses in programming.

Release

Controls the speed of recovery for any given amount of gain reduction. Faster action yields less dynamic range and the presence of more density to the audio. The Release control sets the time constants in relative terms using Slow, Medium, and Fast. Because the time constants are program-dependent, calibrating these controls in absolute time values (ms/dB) would be useless—therefore we use simple terminology.

Clicking Save after adjusting any of the parameters in this section will save the change and close the sub-menu.

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Bass

This section contains the Bass EQ controls. In the the unique Omnia 3 signal processing architecture, these controls are placed in their optimal position — just before the multiband processor sections. Each of the Deep Bass and Phat Bass controls can boost the level up to 12 dB. Be careful here not to overdrive the following sections or over emphasize these lower frequency ranges. When used properly this specialized low frequency enhancement tools can deliver the thunderous bass and warmth that the Omnia is known for, and it can do so without making the sound muddy.

OMNIA 3 PROCESSING SETUP

Phat Bass

A unique enhancement that adds ﬁltered harmonics of the lower registers to the upper bass frequencies. The algorithm extracts information contained in the initial attack to do its work, and low frequency texture is therefore emphasized. Older recordings sound fuller (or phatter) with the added illusion of loudness. Phat Bass EQ also helps devices with small speakers sound like they have more bass than they actually do. A little of this eﬀect goes a long way so be careful not to apply too much boost. Be sure to listen on several diﬀerent types of both ﬁxed and mobile media devices to arrive at a setting that works best for all of them.

Deep Bass

For those who demand thunderous bass, the Omnia 3 has it! Up to 6 dB of “thunder” can be dialed in. And we’re not talking about a simple bass boost, but a rather sophisticated concept that takes into consideration the time alignment of the low frequencies as they pass through the entire Omnia system. It allows a loud, clean low end, with absolutely no sacriﬁce to the overall loudness of your signal. We won’t tell you where the control is in the signal chain. But we will tell you that Deep Bass function is a shelf boost at 90 Hz, utilizing a phase compensated 12dB/octave slope to emulate the EQ function.

X-over

This section contains no adjustable parameters, but here’s a bit of information about why we did what we did. Most multiband audio processors make a compromise in the crossover area in order to provide a ﬂat dynamic frequency response. This is done to avoid audible peaks or dips in the recombined frequency response as the individual band gains change during processing. In an eﬀort to minimize this problem, most processors use a ﬁxed phase-oﬀset in the crossovers. While such treatment helps to minimize frequency response peaks and dips, it also results in a loss of phase linearity, usually causing an increase in “smearing”, as well as reducing musical clarity.

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In Omnia 3’s implementation, the crossover network is carefully time-aligned so that the recombined spectrum remains ﬂat, regardless of the amount of gain control being applied within any band. This true phase linear response assures that harmonic overtones are not displaced in time. The result: the truer, more natural, and more musical Omnia sound.

OMNIA 3 PROCESSING SETUP

Multiband AGC Sections

Similar in character to the wideband AGC described above, but expanded to three bands, this section has the ability to signiﬁcantly add power and loudness to the audio while tailoring and pre-processing it for delivery to the ﬁnal limiter section. The system uses diﬀerent algorithms for each of the bands: The low and mid bands use a feedback conﬁguration, which produces a larger, warmer sound on lower frequencies. The high band utilizes a feedforward design that maintains a more open, natural and musical texture on higher frequencies.

As with any multiband processor, improper adjustment can exaggerate noise at high frequencies, particularly with older recordings that possess a good deal of tape hiss. This can especially occur if the highest AGC band is driven so that signiﬁcant gain reduction occurs, and the band’s gating control is set to the lower end of the scale. During audio segments that have little or no HF content, the AGC will try to increase its gain in an eﬀort to restore treble balance. Even during periods when the gate function is in the default Freeze mode, the problem might still exist with some programming.

Since the user parameters are the same for all three bands, only the LF Band will be detailed here.

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Drive

Sets the audio level that enters the AGC. Increasing the Drive will produce more compression. This control is calibrated between –6dB and +6dB, and adjusts the signal level into the AGC. It should be adjusted to net approximately 12dB of gain reduction with typical programming. Too little gain reduction can defeat the “leveling” function of the Wideband AGC. Too much gain reduction has little additional beneﬁt. Nominal gain reduction values for the AGCs are between 10 and 15dB.

Release

Controls the speed of recovery for any given amount of gain reduction. Faster action yields less dynamic range and the presence of more density to the audio. The Release control sets the time constants in relative terms using Slow, Medium, and Fast. Because the time constants are program-dependent, calibrating these controls in absolute time values (ms/dB) would be useless—therefore we use simple terminology.

OMNIA 3 PROCESSING SETUP

Gate

When set to ON, it will freeze the gain at the last level of processing action that occurred before the audio signal fell below the threshold of operation. This control helps to minimize “pumping”, and the increase of background noise during pauses in programming.

Mix

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This is where the three multiband processors are mixed together. Use care in adjusting this section, as too much level from one particular band may cause an excessive amount of emphasis to that range of frequencies. Such adjustment may also drive the ﬁnal limiter bands in that range of frequencies too hard, causing the sound to become unnatural, dense, and squashed. As explained earlier, if your desire is to “EQ” the sound, you should perform that function using the drive levels in the multiband section. The mixer is primarily intended for minor “EQ” changes to the overall sound. There is a +/-6dB range of adjustment for each band.

OMNIA 3 PROCESSING SETUP

Final Limiter

Omnia 3 employs a lookahead limiter to provide absolute and precision peak control. This limiter has been designed to minimize processing side-aﬀects like IMD, which are usually associated with limiters of this type. Using an innovative design that cancels intermodulation products before they develop, allows this limiter to sound extremely transparent. The limiting function is derived using numerous control signals that monitor one another.

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This type of peak controller is used instead of a clipper because it does not generate the same high levels of THD as a Clipper does. THD causes added diﬃculties in a coded audio system, as the harmonics generated from the clipping action, create added artifacts in the encoder. These are especially annoying at high frequencies.

On the other hand, a lookahead limiter, yields extremely low levels of THD, although it will create some IMD component, and this allows the audio coder to operate with minimal sonic artifacts. There are tradeoﬀs in how each of these peak controllers sound when they are set to produce added loudness. When a clipper is pushed, the audio may appear edgier. This is from the added harmonic content. In contrast, the lookahead limiter will appear busier, or dense as the action of the control signal may be heard, when more limiting is applied.

Limiter Drive

This is where the loudness versus quality is most evident! This limiter’s adjustment range is +/-3 dB, in 0.1dB steps. It is advisable to make minor changes, primarily as the “loudness ﬁne tuner”. Be careful, there is a lot of power here! A little bit goes a long way!

OMNIA 3 PROCESSING SETUP

Output

The peak output level adjustment is done using the individual Left/Right output controls. The control range is from –26dB to +6dB of gain. This level is normally set to the maximum input level, or just below it, of the device that the Omnia 3 is connected to.

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Output Filter

For webcasting applications, especially at lower bitrates, it is sometimes desirable to reduce the audio bandwidth. This is particularly useful if you are using a low cost sound card to feed and external encoder as the ﬁltering in many sound cards in inadequate.

A pull-down menu provides eight diﬀerent ﬁlter response curves that can reduce the spectrum down to 4kHz.

OMNIA 3 PROCESSING SETUP

Bypass

The processing can be bypassed simply by clicking on the BYPASS button located in the lower right hand corner of the processing panels. This is a simple and fast method to compare the before and after aﬀects of the processing.

To save changes to the current processing preset and return to the main “Control Panel” screen, click “Save”. Click “Cancel” to return without saving changes.

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10 Metadata Sources and Filters

Overview

The metadata subsystem gives you the option to tag your audio stream with “now playing” text, album covers and other graphic information.

The metadata is received from an external system over TCP/IP or UDP, then passed to a metadata ﬁlter for parsing. RS-232 serial metadata will be supported in a future version. Once the ﬁlter detects a complete record, it sends the tagging information to the Stream Sender which includes it in the outgoing stream data.

Metadata Sources

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From the main “Control Panel” window, click on “Metadata”. The Metadata window allows you to view currently deﬁned metadata sources, deﬁne new ones, and manage metadata “ﬁlters”.

After you deﬁne metadata sources, they will appear in the metadata dropdown list in the encoder conﬁguration dialog. Make sure to create the metadata source ﬁrst, or go back and edit any of your encoders to add metadata support to the streams.

To add a new metadata source, click on “Add Metadata Source”. Metadata via TCP, or UDP is supported. RS-232 serial is not currently implemented.

METADATA SOURCES AND FILTERS

TCP on specic port

This selection accepts metadata via a speciﬁed TCP port and processes it according to the selected Metadata ﬁlter.

Friendly name

Enter an easily identiﬁable “friendly” name for this metadata source (i.e. Kxxx Automation Metadata).

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Local port to accept TCP connection

Enter the TCP port number where your automation or metadata management system will connect to send metadata to R/2.

Metadata ﬁlter

Select from the list of available “metadata ﬁlters”. The factory supplied metadata ﬁlters should cover a number of common applications. Contact technical support for additional information on creating or modifying metadata ﬁlters.

Click Save to save this entry, or Cancel to return to the Metadata screen.

METADATA SOURCES AND FILTERS

UDP on specic port

This selection accepts metadata via a speciﬁed UDP port and processes it according to the selected Metadata ﬁlter.

Friendly name

Enter an easily identiﬁable “friendly” name for this metadata source (i.e. Kxxx Automation Metadata).

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Local port to accept UDP connection

Enter the UDP port number where your automation or metadata management system will connect to send metadata to R/2.

Metadata ﬁlter

Select from the list of available “metadata ﬁlters”. The factory supplied metadata ﬁlters should cover a number of common applications. Contact technical support for additional information on creating or modifying metadata ﬁlters.

Click Save to save this entry, or Cancel to return to the Metadata screen.

METADATA SOURCES AND FILTERS

Add Serial Metadata source

RS-232 serial metadata is not currently implemented in R/2.

Click Save to save this entry, or Cancel to return to the Metadata screen.

Manage Filters

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R/2 is extremely ﬂexible and can accept stream metadata from a number of diﬀerent sources using metadata “ﬁlters”. These ﬁlters deﬁne the format of the incoming data and reformat it accordingly for stream now playing information. The “Manage Filters” screen allows you to delete existing ﬁlters, or upload new metadata ﬁlters to R/2.

TECH NOTE:

Metadata “lters” are small mini-programs that translate the data received from an external source to tags that the stream sender can understand. Each lter is a separate le stored on R/2. These mini-programs are written using the Lua programming language. Technical Support can assist with developing custom metadata lters. You can nd out more about the Lua programming language at http://www.lua.org.

11 Event Scheduler

Overview

The scheduler is able to start, stop audio processor instances or encoders and change an instance’s processing preset at speciﬁed times. This is useful if you want to encode and stream only certain shows. If you need to process one show diﬀerently than another, you can use the scheduler to change audio processing presets.

Schedule Manager

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Click “Scheduler” from the main “Control Panel” screen to access the Schedule Manager.

The Schedule Events list displays all the events you have created so far. When you select an event from the list, the event commands are displayed in the Commands area. Click the “Add...” button to create a new event.

EVENT SCHEDULER

Add Weekly Event

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Event Name

Enter a name for the event. Keep the names descriptive but use only letters, digits, underscore and space characters. Do not use other special characters.

Time (24 hour, HH:MM:SS)

Enter the time for the event (in 24-hour, HH:MM:SS format)

Days

Click the check boxes for the days of the week when the event should be active.

Commands

The scheduler works by executing commands you enter in the Commands text box. As of program version 1.01.29, R/2 recognizes the following commands:

InstanceStart(“ instance_name”)

Use this command to start the instance named instance_name.

InstanceStop(“instance_name”)

This command will stop the instance named instance_name.

EVENT SCHEDULER

InstanceLoadPreset(“instance_name”,”preset_name”)

Use this command to load the preset named preset_name into the instance named instance_name.

You may add multiple commands by entering each command on a separate line. For example, you could start an instance then load a certain preset for it. Or you could start or stop multiple instances at the same time.

Click “Cancel” to cancel and return to the scheduler window, or click “Save” to save changes and return. The newly added event should appear in the list.

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12 License Management

Overview

The licenses determine the number of audio inputs that can be created in R/2, and the type of audio processing available. R/2 should come licensed from the factory for the conﬁguration that was ordered, but in some cases it may be necessary to enter additional license keys. Examples would be adding additional audio channels, or enabling Omnia.9 processing.

Licenses

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Displays the current license key(s) on the system. Click “Add” to add a new license key.

NOTE:

Licenses are veried with a Telos licensing server. Make sure R/2 is connected to the Internet when you enter the license codes.

LICENSE MANAGEMENT

Add…

Enter the license key. Click “Save” to save or “Cancel” to return to the Licenses screen without adding the license key.

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The code you entered will appear in the Licenses list on the left. Next to the license code, you will see additional text describing the state of the license. The text will initially say “Pending activation”. After 10-15 seconds it should change to “OK”. To reload the license status, use the “Refresh” button. Add each license code you have received individually. If all license codes have an “OK” status, then they are active and the additional features (audio channels or Omnia.9 processing) may be used. See Chapter 7 for conﬁguring additional audio sources, Chapter 8 for conﬁguring processing/encoding instances (including Omnia.9 if licensed) and Chapter 16 for conﬁguring Omnia.9 processing.

If you see an error message next to the license code, please check the license string to make sure it matches the one you received. If the error is “Activation failed”, or similar, then contact customer support at+1 (216) 6220247or[email protected]to learn how to activate your license manually.

13 Omnia 3 Preset Management

Overview

R/2 is equipped with Omnia 3 processing on all audio channels. The “Presets” screen allows creating new presets and managing existing Omnia 3 presets on the system. More information on editing Omnia 3 processing and presets can be found in Chapter 9.

Note:

Omnia.9 presets are managed through NfRemote software (see Chapter 16).

Preset Editing

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Click “Add” to create a new preset based on an existing factory preset, or click “Upload” to upload a compatible preset ﬁle. See Chapter 9 for more details on modifying Omnia 3 processing.

LICENSE MANAGEMENT

Add Preset

Name

Enter a “Friendly Name” for this preset (i.e. Kxxx – HOT HOT HOT) so it can be easily located in the list of presets.

Copy from

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Select a factory preset as a starting point for this new preset.

Click “Cancel” to return to the presets screen without saving changes, or “Save” to create the new preset and return to the presets screen.

14 System Options

Overview

This chapter will cover global conﬁguration options including:

♦ Setting administrative username and password

♦ Modifying web interface and stream server ports

♦ Conﬁguring logging

♦ E-Mail alerts for error conditions

♦ Backup and restore of system conﬁguration

♦ Management of services

♦ Date and Time/NTP

♦ Network interface conﬁguration

♦ Software update

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Options

SYSTEM OPTIONS

Administrative username

The default administrative login is “user”. Although only one administrative login username is supported, this username can be changed if desired.

Administrative password

By default, this password is blank. It is highly recommended that this password be changed, particularly if the R/2 web interface will be accessible from the public Internet. Enter the password again in the “repeat password” ﬁeld.

Web interface HTTP port

Modiﬁes the port for the internal web server for security, port conﬂicts, or other reasons. Port 80 is the default, as with any other standard HTTP server.

Web interface HTTPS port

Modiﬁes the port for the internal HTTPS (secure) web server for security, port conﬂicts, or other reasons. Port 443 is the default, as with any other standard HTTPS server.

HTTP stream server port

Modiﬁes the port for the internal “test” HTTP stream server. The default is port 8888.

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Note:

Normally the default values for these ports are sucient. Only change them if you have a specic reason to do so. Changes to port values will take eect after restarting R/2 or the R/2 service.

Stream server reconnect interval

Sets the interval for reconnect attempts to the stream server. The default value of 10 seconds should be suﬃcient in most cases.

Syslog server address

Syslog is a protocol designed for central logging of various events occurring on equipment within a network. If you wish to log R/2 events to a syslog server, enter the IP address for that server here.

Days to keep log les

Specify the length of time that logs should be kept on R/2, up to 360 days. The default is 30 days.

Local streams path

Speciﬁes where HLS and other streaming ﬁles should be generated when stored locally. There is no local access to the R/2 ﬁle system, and generally this value should not be changed from the default. See the HLS streaming conﬁguration examples from Chapter 8 for more details.

SYSTEM OPTIONS

Email on error

When this box is checked, it enables additional settings for conﬁguring e-mail alerts in the event of an error condition.

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Email address for error notication

Enter an email address in this ﬁeld. While emails may be sent directly to e-mail recipients, they can also be sent as text messages to mobile phones. If you are not familiar with how to do this, see the tutorial at http://sms411.net/

how-to-send-email-to-a-phone/.

Sender name

The default is ‘Z/IPStream R/2 application’. It will be useful to change this, if you have more than one R/2 running in your facility).

Sender email address

Enter the account username.

SMTP server address

You may enter either an IP address in the ###.###.###.### numeric format, or a name address (e.g. smtp.gmail. com). In order to resolve the name address, a working DNS server must be already conﬁgured on R/2 (see network conﬁguration details later in this chapter).

SMTP server port

The default is 25, but you may change it if necessary. Many providers are now blocking SMTP on port 25. An alternative is port 587, which is also supported by most SMTP servers for non-secure SMTP traﬃc. Ideally, you will want to point to an SMTP server on your own network (or your ISP’s network) to prevent issues with relaying.

Use SSL connection

Checking this box will ensure that your SMTP communications are encrypted and secure from hackers but it will work only if your email service provider can accept SSL connections. A number of providers have switched from open SMTP servers (port 25) to SSL and authentication in order to cut down on SPAM.

SYSTEM OPTIONS

Email send interval

Determines how frequently Z/IPStream sends you email alerts about a problem. The default is 3600 seconds (one hour). We suggest not setting it lower. You want to have a value large enough to give you enough time to resolve the situation before the problem repeats. Otherwise, in case of a persistent problem, you may end up having your email ﬂooded with too many messages.

SMTP username

Enter the user name for your SMTP server here (if applicable).

SMTP password

Enter the password for your SMTP server here (if applicable).

Click the Save button to make the changes permanent. You will then be returned to the Control Panel page.

Backup and Restore

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From the “Options” screen, click on “Backup” to access the Backup and Restore screen. This allows you to perform and restore complete backups of the R/2 conﬁguration (including presets).

Refresh

Refreshes the displayed list of backups currently on the system.

Backup

Performs a backup of the R/2 conﬁguration.

Upload

Uploads a previously downloaded backup.