Gear Talk: Audio Distribution Part 5

Welcome to our final article in the audio distribution series.  Over the last 4 weeks we’ve covered the gamut from analog patch panels and analog splitters, to the older digital distribution formats, to the latest and greatest in audio networking in use today. Today we’ll focus on some of the more proprietary but well known manufacturer specific formats that are widely used today, most of which you will probably recognize. Buckle up and let’s get started!

First up, is Waves Soundgrid.  Anyone familiar with using Waves plugins live, has likely come across soundgrid in some form or another.  Soundgrid is Waves’ proprietary format and allows for up to 128 bidirectional channels to flow over a single ethernet cable.  Soundgrid can handle sample rates between 44.1K up to 96K. Soundgrid is a layer 2 network technology that uses one of three applications to route audio between devices.  Waves MultiRack which, when coupled with Waves processing servers allows the end user to process additional audio tools through a live console in real time with little to no latency.  MultiRack can act as a soundgrid router in addition to its duties as a plugin host. Waves LV1 is an actual software audio mixer with 64 stereo channels of audio and can hold up to 16 different soundgrid compatible devices.  Waves LV1 can be used as a traditional console or used as a giant audio router for soundgrid, or both. LV1 also allows end users to use Waves plugins directly in the mixer allowing for very flexible and detailed mixes. The last application, Waves Soundgrid Studio is primarily designed for DAW use and allows for additional processing power by offloading plugin processing to an outside PC instead of using the DAWs processor.  This too, can act as a soundgrid router. Waves Soundgrid is a fairly robust platform with options of integrating with MADI, AES50, Dante, and AES3, as well as having a driver for computer audio to be added to the Soundgrid network for recording or multitrack playback. Waves also works with many different manufacturers to make option cards for direct connectivity to many different audio consoles. The only possible downside to Waves Soundgrid is that a host program like LV1, MultiRack, or Soundgrid Studio have to be actively up and running for audio to properly flow and stay routed. You are also completely reliant on perfect performance from all parts for the system to work. 

Aviom’s ANET is another format still in use, while older and mostly replaced with Dante it can handle a 64 channel stream.  ANET uses a proprietary Layer 2 network technology to shift 48K audio around to different devices. It uses proprietary hardware and software to make and control routes.  Aviom’s ANET is most popular for its 16 channel personal mixers. Aviom made ANET cards for many console manufacturers and also analog to ANET converters that allowed end users to convert analog line level signals to ANET signals to be used with their personal mixing systems.  ANET was also utilized with limited success in large venues for it’s channel count and routing capabilities.

To help connect multiple consoles, DigiCo developed Optocore.  Optocore is a Ring distribution system that can handle up to 512 channels of audio at 96K.  Optocore can be used in a star topology but loses some of its redundancy. One nice feature with Optocore is that outputs are automatically calculated by the number of inputs being assigned so a lot of the math is done automatically for the end user.  Optocore can also be interfaced with a number of other manufacturers via option cards or via a MADI converter.

For major broadcast consoles there are a couple options for users to choose.  Calrec’s Hydra and Hydra 2 networks and Lawo’s NOVA networks. Each of these are basically giant routers with thousands of inputs and outputs also capable of handling hundreds of processing channels.  Both formats have redundancy for both DSP and control. Both use a computer to make routes but are not dependant on the computer for the routes to be maintained. Both Calrec and Lawo can integrate with almost any audio standard and have options for both analog and digital IO.  These consoles and their router frames are intricate pieces of technology and can act as a hub to a television studio or an OB sports or entertainment truck. Both utilize remote I/O boxes so that they can be networked through an entire installation and can easily be scaled up or down based on the needs of the current client or production.

Allen and Heath has built a new 96K protocol for the dLive and SQ Platforms called GigaAce.  GigaAce is capable of handling over 300 channels of bidirectional audio and control down a single ethernet cable.  GigaAce is a layer 2 format and primarily point to point to point with the ability for redundancy. One useful feature is that GigaAce can carry control data for multiple A&H consoles down the line so one A&H console with other A&H consoles connected to it via GigaAce can bridge control networks off a single ethernet cable.  Allowing an end user to easily control A&H consoles spread out at an installation.

StageTec uses a proprietary network protocol called Nexus to handle audio networks of 4096 by 4096 when properly configured.  Like Lawo and Calrec it can handle consoles with hundreds of processing channels. StageTec is seen mostly in large theatre type installations but is making inroads into broadcast setups.  StageTec consoles are also capable of distributed I/O boxes and can easily be scaled based on the needs of the client or production.

Yamaha has developed a new network technology called TWINLANe.  In addition to their heavy use of Dante in their CL, QL, and TF mixers Yamah has also developed TWINLANe as a method for audio transport in their PM Series consoles.  TWINLANe allows for 400 channels of audio to be distributed to consoles or devices on the network.

SDI, or Serial Digital Interface is the de-facto standard of video transport in use today.  Almost all professional video devices use SDI to carry audio and video. While SDI is not necessarily an audio format I did think it important to cover because HD-SDI has the ability to carry up to 16 channels of 48K audio down one stream.  This means you can send full HD video and 16 channels of audio down one cable. This can be extremely beneficial to installations that already use SDI to transport video because those SDI lines can also carry audio and can be used to distribute audio where no additional audio cabling may be run.  Keep in mind however, that not all SDI devices are created equal and that some can only see two, four, or eight, channels. If you’re looking to utilize the audio channels inside an SDI stream make sure the devices involved (including any SDI routers involved) can handle the amount of audio needed.

While there are many other audio formats in use around the world today I wanted this article series to cover the main ones in use and some of the original formats that grew the audio industry.  Understanding the basics of digital audio and digital audio networking will be crucial as audio consoles merge ever more with IP protocols and turn more and more into network devices. As capabilities and inter-connectivity grows, it’s crucial to understand the ability of your console to connect with other devices. We know that these last few weeks of audio distribution may have been a bit of drinking from the fire hose so please feel free to ask questions below, drop a thought on Facebook, or email us at As always, if you want to be emailed when a new post is live on the site, sign up at this link! See you all next week!

Gear Talk: Audio Distribution Part 4

Welcome back to our series on audio distribution.  In the past series we have been talking about analog audio distribution, digital audio clocking and sample rates, older digital audio transport formats, and now this week we dive into true audio networking.  These days we have a lot of options for getting digital audio devices to be able to communicate and the world of actually networking multiple consoles together on a single network has emerged. The two dominant methods currently on the market are Dante and AVB.  Both have their advantages and disadvantages and today we will dive into both.

Up until now we’ve been dealing with formats that are direct connections or layer two networks.  What is a layer two network you ask? A Layer two network is the simplest of an IP network and EtherSound and CobraNet use it.  A layer two network is simply a collection of devices with addresses that send data back and forth. Now, there is a limit to the amount of devices and information because layer two networks have no logic to assign IP addresses, they have no advanced routing capability and no ability to handle some of the higher end functions that make larger networks run faster or even run at all.  Layer three networks can handle the distribution of IP addresses and they have advanced routing features that allow packets and other data flow efficiently on the network. Dante and AVB are both layer three formats that allow for these advanced features and as such, can handle tremendous amounts of audio data at once. One of the reasons Dante and AVB can work in a networked format and allow for such flexibility is their use of PTP network clocking.  PTP, or Precision Time Protocol, is a way of network time synchronization that utilizes an oscillating clock in all devices on the network. These clocks all keep time at a consistent similar rate. When connected to a network they all share time data and align themselves accordingly. This is where having a “Master Clock” can come into play in Dante and AVB networks. 

Dante is owned by a company called Audinate and licenses chips and software to any company who would like to use them.  Because of this, Dante has become very pervasive in the industry. They have a range of different chips which can be integrated into any device the end user requires.  Dante works by setting up “flows.” A Dante flow is a collection of up to four channels, in Dante parlance there are both transmit and receive flows. These flows can be sent to one or multiple devices at once.  To route various devices in a dante network a standalone program is needed to make routes and configure settings. It is not plug and play. As with other digital audio transport technologies and digital audio networks, clocking in Dante is key.  In Dante networks one device must be the clock master, just as when using MADI, AES50, or other formats. This can be controlled with the Dante Controller application by selecting your “preferred master.” Most Dante devices have the ability to become clock master and the Dante network can determine this automatically based on a variety of factors if no master is chosen manually. This is particularly useful when using a mixing console that is the primary device in the network.  Thankfully Dante allows the ability for devices to derive their clock from an outside source and then allow that source to be the network clock master. This is good for audio consoles that have a Dante Card but use an outside clock or their own internal clock. The Dante card can pass that clock off to the rest of the network. This keeps all devices on the network synced. If the Master Clock should drop off the network or become corrupted a new device will be elevated as the master clock automatically with little to no interruption of audio.  This makes Dante very stable as no one device can totally destroy the network.

One of the main advantages to Dante is that it has the ability to work on almost any network infrastructure so long as that infrastructure can scale with the traffic.  Also, because it is layer three, it uses more advanced identification methods that allow devices to be very flexible. Patches made in Dante Controller can be kept no matter what Dante network connected devices are on and are not dependent on Dante Controller being left on, nor are the patches tied to a specific instance of Dante Controller.  This is due to the ability of all Dante devices being layer three. The devices do not necessarily need a specific IP address to work. Another advantage to Dante networks is the ability to run Dante alongside regular network traffic (with the implementation of QoS). Now, this takes some higher level of networking knowledge and if it is going to be deployed in a large network it is definitely wise to bring in a qualified networking engineer to help make sure you’re not affecting other network traffic and that your Dante network is running smoothly. Most people however use Dante in it’s own enclosed network.  This makes it very easy to network Dante devices. As you don’t have to worry about outside traffic interrupting your Dante audio streams or your Dante audio streams messing with your outside network traffic. Another great aspect of Dante, is it’s ability to use redundant topology. What this means is that most Dante devices have the ability to have Primary and Secondary ports and can stream audio over two completely separate networks. When set up properly this can isolate the network from failures such as bad or accidentally disconnected cables, or a network switch that has malfunctioned. The key here is to make sure your secondary and primary networks NEVER interact.  They should always be physically separated or at least separated by VLANs.

AVB or, Audio Video Bridging, is another prominent audio network widely in use today.  It works very similarly to dante in that it utilizes layer three networking technology to pass audio.  AVB uses PTP much like Dante, except that it utilizes a more advanced version with better sync abilities for better timing across the network.  Like Dante, AVB uses a separate control program for routing and Clock Master selection. AVB has three different device types; a talker which sends audio,  listener which receives audio, and a controller, which can both send, receive and route or just route. This means that some AVB devices do not need a standalone application to make routes, they can route themselves.  AVB uses “streams” of up to 8 channels. These streams are not bidirectional and only flow in one direction. So you can have a stream of inputs to a device, but would then utilize another stream for outputs on that same device.   One main area AVB differs from Dante is that it has a comprehensive certification platform for devices and network switches. This means that using an AVB switch guarantees a certain amount of network functionality without having to worry about latency or other issues.  AVB has done the work to make sure anything with an AVB label can handle a certain amount of AVB traffic. This makes it very “plug and play” and is the main advantage over Dante. The one downside is that AVB does not work alongside normal network traffic. The end user is very strongly pushed towards using AVB certified products and does not have the flexibility to use already existing network infrastructure.  The main advantage to this though is that little to no networking experience is needed to setup a working AVB network. Another key advantage of AVB over Dante is its ability to manage network bandwidth. Both Dante and AVB utilize more network bandwidth as the end user adds flows or streams. AVB can track this and keep the end user from routing too much audio through a certified switch. This again, makes it very appealing to end users with little to no network experience.

As with anything, Dante and AVB are tools.  Both have their strengths and weaknesses and both can be used very effectively when deployed to the right situations. If you have any questions, please feel free to leave a comment below or on facebook. Come back next week for our last post in the audio distribution primer when we cover some of the more popular manufacturer audio distribution formats. If you’d like to be notified when a new post is published here at Studio.Stage.Live subscribe at this link and we will email you. See you next week!

Gear Talk: Audio Distribution Part 3

Welcome to week three of our audio distribution primer. Up to this point we’ve covered the analog ways we distribute audio signal. This week  we’ll discuss some of the most common digital audio transport formats currently in use and discuss items to consider when using them.

The first digital audio format I’d like to discuss is called AES3 and it’s one of the older and still very versatile audio formats in use today.  The AES3 format or more often just referred to as AES was created in 1985 and refined as late as 2003. AES3 is basically a two channel format that allows two devices to communicate digitally between them.  It typically uses a single XLR or BNC cable to connect the devices. AES3 can run from 44.1K to 192K depending on the device. Most newer consoles and devices can also reformat the bitrate so that a 96K device can talk to a 48K device well enough.  Keep in mind when connecting to AES devices, that both devices have the same bitrate or that one or both can change their bitrate at need. AES has its advantages over analog because it is less susceptible to hums and buzzes and also keeps everything digital and keeps you from having to deal with gain staging as there are no preamps to deal with once in the digital realm. Many digital consoles for sale today will include a few AES ports on it’s local I/O, go take a look and see if you’ve got one!

The second digital audio format is AES10, more commonly known as MADI.  MADI is a 64 channel bidirectional format often found in broadcast applications for its ease of use and high channel count.  MADI, like most other digital formats relies heavily on having the correct bitrate and having a single master clock. MADI is nice because along with it’s high channel count its fairly easy to diagnose problems.  So long as you have the same bitrate between your two consoles (or other audio devices) and have the cables connected correctly, MADI tends to just work. There are no address to assign like with some other protocols.  It’s a 1 to 1 connection so there’s very little to mess up. MADI typically uses BNC cables. One for sending and another for receiving. There are some CAT5 MADI devices but BNC and optical fiber are the most prominent in use today.  Using fiber, MADI Can be run for thousands of feet or even several miles. MADI is a 48K native format but can be run at high bitrates with doubled cables and special cards or a halved channel count. There are also devices that can bridge MADI devices together without having to have a single clock master which allows for even greater flexibility, a lot of manufacturers make USB to MADI interfaces (i.e. – Madiface XT) which also allows for easy recording.  Several console manufacturers use MADI as their default method of communication between stageboxes and consoles.  Because of this MADI also has a 56 channel bidirectional mode where control commands for stage boxes can be sent using the bandwidth for the last 8 channels.

Another popular audio format still in heavy use is AES50.  This is primarily used by Music Group companies; Midas, Behringer, and Klark Teknik.  AES50 is another point to point network like AES3 and MADI, with the ability to carry 48 channels bidirectionally.  AES50, like MADI and AES is dependent on clocking and sample rates. AES50 runs over a single cat5 cable and can be run up to 300ft or can be converted to fiber and run much longer.  AES50 is a 48K native format but can be used at 96K with a halved channel count or doubled cable run. Midas PRO series consoles use AES50 in 96K mode and use a second cable to handle the higher channel count.  AES50 like MADI, is point to point and doesn’t deal with addressing or extra computer programs for control. Also like MADI, AES50 is simple and easy to use for its ease of troubleshooting. There are multiple options of converting AES50 into other formats or even bridging AES50 networks but for the most part it is similar to MADI and is a great simple sound connection.  The addition of AES50 to the Behringer X32 makes it especially powerful for its’ price range as the console can route two seperate AES50 streams through itself to other AES50 devices making for easy digital splits between consoles and other devices.

The next audio format for discussion is CobraNet.  CobraNet was the first audio format to market audio over ethernet to the masses successfully.  Starting in 1996 it offered the ability to send between 8 or 64 channels bidirectionally over cat5 cables at 48K uncompressed audio.  At first it was limited to a bit depth of 16bits but eventually was able to offer bit depths of up to 24bit and sample rates of 48K and eventually 96K.  CobraNet also had the ability to be networked but had many constraints. CobraNet is the progenitor to most modern networking formats today and was placed in many theaters studios and large event spaces where low channel counts needed to be delivered long distances (think delays speakers at festivals or opera houses).  CobraNet works by assigning channel numbers to a sending device and then assigning channel numbers to a receiving device. This is typically done with software inside the device or with dip switches. This made it somewhat flexible and allowed for limited routing possibilities. Another feature of CobraNet was its ability to run over fiber via network switches so it could be run over very long distances and also be routed to various places without any of the usual pitfalls when running audio over long distances with copper wire.

The last format is a little lesser known and is not in much use anymore.  EtherSound is a 64 channel network of 48K audio over a single cat5 cable developed in 2001.  EtherSound is a proprietary format started by DigiGram and used by Yamaha for their LS9 and M7CL consoles and can be added to their other consoles by an option card.  While many manufacturers made option cards and devices for EtherSound it was a little too ahead of its’ time. It could be networked on layer two switches and used a special program to patch channels similarly to the way Dante and AVB patch today.  Unfortunately EtherSound didn’t gain a ton of popularity and is not in wide use in the wild. Even with the help of Yamaha’s pervasive console sales EtherSound never caught on in the way that newer digital network formats have today. Personally, I think that it was just a little ahead of its’ time and implemented in a clunky way that scared too many technicians early on.  I’ve worked with this technology and it’s actually pretty cool but it can be finicky for the end user.

Both EtherSound and CobraNet work by dividing the audio into data packets and sending them out (just like your computer does when sending files over the internet) and then separating the packets at the other end and converting them to another digital format or to analog audio depending on the application desired.  These are two (CobraNet and Ethersound) “true” audio networking standards where AES3, MADI, and AES50 are all still point to point connections. Next week we will continue to dive into audio networking as we discuss the Dante and AVB formats which are the most widely used formats in use today. If you want to be sure not to miss it, subscribe at this link, and we’ll be sure to send you an email when next week’s post goes live! If you have any questions about what was discussed today, be sure to drop a comment below or hit us up on facebook! See you next week!