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Being a reporter in times of lockdown = challenge accepted!

Being a reporter in times of lockdown = challenge accepted!

BEING A REPORTER IN TIMES OF LOCKDOWN = CHALLENGE ACCEPTED!

Remote broadcasting – Audio-over-IP

A smart trick and flawless broadcasts from home thanks to Digigram’s IQOYA Guest Preview.

Basically, reporters’ job is to “report”, therefore they need to be on the field (preferably with an IQOYA Talk) to meet real situations and craft the news. But, in times of nation-wide lockdowns, when social distancing is a golden rule, how do reporters… Report?

Bypassing this new (and hopefully, temporary) normal and reporting from a comfy couch or cluttered garage is a thing: countless images have flourished over the web, depicting journalists happily working from home, with a ton of equipment imported from their usual studio.

However, shipping equipment to each journalist’s place and ensure a smooth news service is sometimes beyond possible. 

At Digigram, we have come up with a ridiculously easy trick – simply hijacking our IQOYA Guest Preview service and turning it into remote reporters’ best companion (beside the coffee shot)! Basically, that hands-down service was designed to host remote guests during a show or broadcast. Whether using a smartphone, tablet or being interviewed from a computer, the guest needs absolute no knowledge of any radio-specific tools, tunings and features.

All it takes is a reliable internet connexion and a web browser. Sound quality for the stream transmission is top notch (you know how much we care about sound quality) and the service has the simplest configuration.

Turning IQOYA Guest preview into a work sidekick is an easy-peasy operation that requires only minimal equipment. This lightweight solution takes no more than 4 steps:

#1 Choose the best place to broadcast

Is it your sofa? Your desk? The kitchen counter? You may roam your home to find the most convenient setting for your computer or smartphone as a scenery for your upcoming broadcast.

#2 Fix yourself a warm drink

Carefully choose the mug for your home made cappuccino – if you were to make a selfie while broadcasting, you’d certainly want to look at your best. So, discard that good ol’ chipped mug you received for your 12th birthday.

#3 Clear the place

Kick the kids or the cat out of the place (but #stayhome anyway, ok?) – This is your workplace, after all. And you are about to broadcast your regular show.

#4 Open your favourite browser

Launch IQOYA Guest. Log in. Sip some coffee. Click on CALL to connect to the studio. Take another sip. Mic on. You are on air.

Discover IQOYA GUEST and broadcast from home (or wherever you want) today!

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IQOYA SERV/LINK: the most powerful AoIP codec in the world

IQOYA SERV/LINK: the most powerful AoIP codec in the world

IQOYA SERV/LINK: The most powerful AoIP codec in the world

Distribution – remote broadcasting – Infrastructure – Audio-over-IP

s

ince 1985, Digigram has innovated in the development of sound processing software and hardware. From the famous PCX3 to its solutions for SIP infrastructures, discover 35 years of history at the heart of this company on the border between IT and audio.

SERV/LINK: one Rack Unit. No compromise.

In 2009, Digigram invented the first stereo audio codec dedicated to transport over IP networks which natively integrated the redundant streaming function on two separate networks.

Always at the forefront of research, Digigram sensed the need to go further in the design of its codecs. In order to develop a tool that would bring an indisputable advantage to the broadcaster, Digigram put the “user” above everything else. 

While radios already offered several simultaneous audio programs, Digigram chose to propose a codec with high channel density in only 1U. This codec had to manage talk-back channels and commentaries via IP. It had to support remote broadcasting of programs between remote studios and generate multiple WEB radio streams for CDNs. This led to the creation of IQOYA SERV/LINK, an extremely reliable and compact codec. 

This new generation codec was the first IP audio codec in the market to offer as many channels in 1U. IQOYA SERV/LINK accommodates Digigram’s professional audio cards and relies on the Fluid IP™ streaming engine developed by the brand. It is available in several versions with different format compatibility: MADI / AES67 / AES / EBU / DANTE / Analog.

The functional scope of the IQOYA SERV/LINK was then extended to support MPEG-TS streaming, followed by applications using DVB Audio transport and then for simultaneous live remote broadcasting applications.

In the “remote broadcasting” configuration, it is the only IP audio codec in the market capable of managing up to 64 mono or stereo full-duplex IP streams, regardless of the audio encoding format and the audio connectivity.

In the “program distribution” configuration, it is the only IP audio codec in the market which supports up to 128 mono, multiple-format, input and output channels with the possibility to simultaneously encode, decode and transcode IP audio streams.

Two redundant hot-swappable power batteries and the 1 + 1 redundancy functionality make the IQOYA SERV / LINK a very secure and reliable, program distribution (WEB radios, DVB Audio, FM) and remote broadcasting (commentaries, talk-back channels, inter-studio links) codec.

For more than 10 years, Digigram’s IP codecs have been used for the distribution of radio programs. More than 10,000 IQOYA are currently in operation 24/7 all over the planet.

They are known for their reliability and versatility, as well as for the resilience of the audio connections they enable. Digigram has put all its know-how and passion to perfect its codecs IQOYA X/LINK and IQOYA SERV/LINK ranges.

Bridging LAN Audio to external WAN: Security is key

Remote broadcasting, on-field coverage, live game commentary, audio distribution to transmitters or web radios – the audio production environment has multiple forms. These streams traditionally travel through dedicated analog/AES-EBU audio infrastructures. Those legacy infrastructures use audio codecs to interconnect the audio world with the Wide Area IP network – among which is Internet. Within a traditional company IT infrastructure, codecs are designed to remain in the so-called DMZ, the first security layer and farthest possible ‘exclusion zone’ from the company’s sensitive data – the latter being lodged into a ‘privileged zone’, a deep layer heavily protected by firewalls.

Simplifying and Reinforcing a DVB multiplex generation workflow with only 2 AoIP codecs

RCF is a network of Christian associative radios, which includes 64 stations in France and Belgium, followed by 630,000 listeners every day. It is the 1st independent generalist radio network of proximity in France. RCF offers local and national programs, implemented by 300 employees and 3000 volunteers, and is available in 5 broadcasting modes (FM, Mobile, Internet, Podcast and DAB +).The national structure, based in Lyon, generates the national program (20h of program / day) and the local radios broadcast 4h of their own program, according to a schedule common to all the radios.

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At last – a simple solution for MPX signal transport over IP!

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At last – a simple solution for MPX signal transport over IP!

Distribution – Infrastructure – Multiplex – Transmitter

D

igigram has developed a solution to transport composite MPX signals from the studio over an IP network with the same quality and reliability as a point-to-point link.

MPX (multiplex) is the composite signal dedicated to the transmitter site. It is the only signal accepted by the transmitter (exciter) for FM diffusion, and it is traditionally delivered via a point-to-point link.

What are the main differences for an MPX signal delivery, between an IP wireless point-to point-link, and an IP network link?

There are two distinct advantages of an MPX-over-IP transport solution for broadcasters:

The first is a significant gain in flexibility and scalability; and the second is a reduction in equipment to be maintained and deployed on the transmitter sites. However, to ensure the same levels of quality and reliability over an IP network as for a point-to-point link, attention needs to be paid to IP transport and jitter mastering.

IP mastering

Built on the universally acclaimed IQOYA audio-over-IP platform, Digigram has developed a highly reliable solution allowing broadcasters to deliver MPX signals over IP networks.   All the mechanisms that have helped to make IQOYA a worldwide success for rock-solid audio-over-IP streaming have been deployed for MPX transport over IP. That includes FEC, dual streaming with time diversity, network jitter mastering, and smart synchronisation on the incoming stream. Plus, additional smart features have been added, such as decoder synchronisation for MFN, and auxiliary data tunnelling.

Jitter mastering

Why is it important to master the jitter? Mastering the jitter guarantees the same reliability as a point-to-point link. That means that transporting MPX signal can be achieved either via point-to-point or over an IP network depending on the broadcaster’s requirements.

Data rate mastering

MPX is a composite signal, hence smart codec design requires far less dynamic range than baseband audio. For instance for Audio stereo + RDS, 144 kHz sample rate @12 bits encoding with analog transmission leads to a very acceptable 2 Mb/s instead of the usual 3.3 Mb/s or more. With AES192 digital MPX transport 2.6 Mb/s can in addition transport an auxiliary subcarrier.

Conclusion:

The IQOYA hardware codec is a highly versatile codec for program delivery. It delivers analog or digital baseband audio or MPX through point-to-point or IP networks links. These baseband audio links are distributed to FM transmitter sites, web radio CDN, and/or DVB multiplexers for satellite, and DAB encoders for digital radio.

Digigram is proud to be one of the few in the world able to transport MPX over IP network with a very high jitter resilience and acceptable bandwidth.

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Five benefits of using a SaaS application for radio outside broadcasting

Five benefits of using a SaaS application for radio outside broadcasting

Five benefits of using a SaaS application for radio outside broadcasting

Contribution – Infrastructure – Application

The high expectations of today’s audiences are driving up content quality. Listener interests, rather than technical constraints, now dictate the nature of content being produced. Looking at these trends as a whole, broadcasters are seeking more flexible outside broadcasting (OB) solutions capable of supporting enriched content creation. Software as a service (SaaS) applications are becoming increasingly popular because they bring a high ease of use while reducing overall costs. Here are five other good reasons for using SaaS for OB.

1. Less money spent on equipment

Thanks to the SaaS model, there is no need to invest in costly equipment or infrastructure. In addition to enabling users to generate a broadcast-quality interview from any connected device, certain SaaS applications are available on a pay-as-you-go model that further reduces costs.

2. More time for content, less for setup

Accessible directly from a web browser and available after a few clicks, a SaaS application can eliminate the need for a technician to set up interviews. Because technical requirements are reduced to their minimum, journalist or producer staff have the freedom to focus on producing quality content instead of sweating over technical matters.

3. Upgrade or downgrade as needed

The best broadcasting SaaS applications give users the ability to extend the number of contributors to a program, adding and removing contributors with ease. This makes a SaaS OB application the perfect tool for radio stations’ editorial managers!

4. Maximum flexibility

Functions such as master control room (MCR) codec operation may be distributed to each studio or centralized according to the workflow habits of each radio station. Accessible from any connected browser on a local PC or a nomadic device, the simplified operation enabled by the SaaS model can allow the technical team to be more adaptable and agile.

5. Automatic synchronous update to the latest versions

A SaaS application for audio OB productions is a time-saver. Technical setup time is minimized because it is simply not needed anymore. Furthermore, certain SaaS applications dedicated to radio broadcasting provide automatic updates and allow users to benefit simultaneously from all the latest versions and options.

Conclusion:

Broadcast professionals are looking for simple, reliable solutions that will help them to create enriched audio content. Where they previously needed a mobile studio, technicians, and expensive or bulky equipment, they now need only a laptop, the internet and a professional SaaS application in order to maintain broadcast quality, performance and continuity of service.

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Three Tips for Setting Up and Managing a Standard and AES67 Network

Three Tips for Setting Up and Managing a Standard and AES67 Network

Three Tips for Setting Up and Managing a Standard and AES67 Network

Contribution – Infrastructure – Studio

A lthough large radio stations and groups typically have the resources to develop dedicated networks for AES67 (IP audio) and for corporate traffic, small and mid-sized stations often need to pursue a less-expensive approach. Here we provide three recommendations that help smaller radio broadcasters ensure the peaceful coexistence of AES67 audio-over-IP (AoIP) traffic and standard network traffic on a single network.

Seamless Handling of AoIP and Standard Traffic on One Network

The AES67 standard makes AoIP protocols such as Dante, Livewire and Ravenna interoperable, in turn simplifying connectivity and reducing hardware and overall clutter. To identify the requirements of handling and optimizing AoIP traffic and standard traffic — web, video transfers, and corporate data — on one network, we examined the interoperability of AES67-compatible products from four different manufacturers and then created three recommendations for small and mid-sized radio broadcasters in setting up and managing AES67 (IP audio) on an existing network.

In short, we recommend the use of Internet Group Management Protocol (IGMP) snooping protocol to distribute predictable bandwidth on a high number of streams; installation of PTP (Precision Time Protocol)-enabled switches; and activation of quality of service (QoS) mechanisms to limit disruption and avoid audio glitches.

Making It Happen: Three Keys to AES67/AoIP on a Standard Network

Recommendation 1: Enable IGMP

First step: Activate the IGMP. By realizing select communications, IGMP enables the management of subscriptions to the multicast addresses. It manages the distribution of network and audio packets, in turn preventing saturation of bandwidth and reducing clutter on the network.

Recommendation 2: Consider PTP

The switch used to support this single-network model can be PTP-enabled or not. If it is, it facilitates better synchronization of the network and is less sensitive to the disruptive effects that can be generated by the corporate network. Although a PTP-enabled switch is more expensive than the alternative, its benefits often are worth the extra cost. When the switch is not PTP-enabled, synchronization or jitter problems may occur during heavy use of the network. (One way to avoid this issue is to configure the QoS present on the switches.) Without PTP support, clock jitter on AES67 traffic becomes quite high above 100 audio channels.

Recommendation 3: Configure Packet Priority via QoS

QoS is used to manage the priority of packets over the network, and it improves the network capabilities of a switch that does not contain PTP. The AES67 standard imposes rules on manufacturers regarding QoS eligibility. It integrates the management of the priorities of the IP packets and to which class of service they are associated. The equipment and the network must follow the AES67 recommendations to ensure a uniform understanding of priorities.

Optimizing the Single-Network Model

Enabling the IGMP is the most important step radio broadcasters can take to avoid saturation of the audio network, regardless of whether or not they are using a PTP-enabled switch. That said, the bottom line is that the combination of PTP and Qos makes it possible to increase the quality and quantity of available audio streams. Through our tests, we determined that the combination of a non-PTP-enabled switch with QoS enabled made it possible to exchange 120 full-duplex channels on the network without loss of packets and without any latency problems.

Conclusion: Creating an AES67 Network is an Option for All

Network audio competence is increasingly essential for radio broadcasters, but today’s AES67-compatible IP-based products for AoIP — and a few helpful tips for implementation — can go a long way in enabling small and mid-sized operations to take advantage of IP audio and its many benefits.

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AES67 for TV applications

AES67 for TV applications

AES67 for TV applications

audio essence will no longer “follow” video; it will be produced independently and dynamically assembled with metadata in the delivery of the final content, leading to new productivity schemes

B y its nature, the delivery of audio over IP infrastructure enables a distributed approach to handling audio streams. Without the need to rely on specific pieces of equipment, connected in a point-to-point model, broadcast and A/V facilities can realize much greater flexibility in routing, scheduling and managing audio streams. At the same time, audio-over-IP (AoIP) technologies simplify operations by allowing users to maintain synchronized content within complex multiple-source, multiple-destination workflows, and to manage metadata more effectively in terms of end-to-end content management and overall operations.

The AES67 standard

The AES67 standard for AoIP interoperability has evolved to the point that its performance is roughly comparable to that of MADI (AES10). With the industry increasingly focusing on system approaches, technology suppliers are addressing the “discovery gap” — which was deliberately omitted by the AES67 Working Group — by bridging stream discovery at the system level or in equipment. At the same time, vendors are pushing forward control and monitoring specifications, such as AES70, NMOS, and others, that further enhance system implementation of IP-based applications.

Audio essence will no longer “follow” video

The value of AoIP to mission critical IP-based audio distribution applications has been widely demonstrated, firstly outside the studio with the ACIP standard and now inside with AES67, and video broadcasters now are considering integrating AES67 into video-over-IP environments. In fact, through the Alliance for IP Media Solutions (AIMS), broadcast equipment and solution suppliers have come together to ensure an easier transition to IP by supporting VSF TR-03/-04, SMPTE 2022-6, and AES67 standards. With such an approach, audio essence will no longer “follow” video; it will be produced independently and dynamically assembled with metadata in the delivery of the final content, leading to new productivity schemes.
When broadcasters and A/V facilities take full advantage of IP infrastructure’s potential to increase their workflow productivity and flexibility far beyond simple gains such as reduced installation or transmission costs, they realize the optimal cost-benefit ratio. To do so, they must undertake solid IP infrastructure engineering, taking care to establish the multicast routing and PTP clock synchronization capabilities essential to mission critical broadcast operation.

Because audio also needs to be transmitted between facilities, several audio manufacturers propose “Media Gateway” solutions to bridge the LAN-synchronous and ultra-low latency AES67 to outside WANs, using the ACIP standard (EBU Tech 3326 and EBU Tech 3368). As an example, for countrywide studio-to-studio transport of AES/EBU, bit transparency provides for the highest PCM audio quality, Dolby E® and user bits transport. This application benefits from a dedicated QoS priority to transport PTP (Precision Time Protocol) clock reference at reasonable jitter on a managed WAN to accurately synchronize and phase audio at the sample level.

Another important use case consists of quickly installing a high-performance audio link between an AES67-enabled OB and an AES67-enabled venue, such as a theater or a stadium.

Since AES67 avoids both dedicated audio lines and costly hardware routing, all applications requiring flexibility, high performance (high channel count, precise time-alignment, low latency…) and low capex could benefit from AES67 interoperability.

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