Gear Talk: Vocal Capsules

In today’s world, we have quite the choice of capsules. Watch a lot of old concert recordings and everyone was using one of a just a few options. But today, there are multiple manufacturers making many different varieties of microphones. Whether your twisting the capsule onto a wireless transmitter or swapping out a wired mic, it certainly isn’t much easier to start tailoring your selection to your use case and hopefully to the person using it. That is the topic of this series. Over the next few weeks I’ll be taking a week and talking about how I use each capsule that I have. One of the downsides here is that I only own a subset of Shure capsules (with a few outliers). I’m not sponsored by Shure (though I’d like to be) and this isn’t meant to be a sales pitch but rather me sharing how I match capsules to singers on a weekly basis. If you are heavy Sennheisser, EV, DPA users, reach out to me via email ( and we can work together to write a few posts about a wider variety of choices (reach out quickly and we can get those integrated into this series). 

This week I’m going to talk about the standard: the SM58. I usually buy this head for every wireless mic I buy because it is the most multi-purpose mic that I know of. If you can’t at least make something or someone decent with a 58, you’re doing it wrong. For over 50 years this microphone has been used for so many popular artists like Martina McBride, Cheryl Crow, Rascal Flatts, Iggy Pop, Luke Bryan, and so many more. But what sets it apart? Most people cite it’s classic performance with a flat response except a brightened upper mid which gives it a warm but present feeling for most. But it’s also known to be so durable in fact that people seem ok letting it be run over by a truck. Shure claims each part, when received from their contractors, is put through military grade endurance testing and I believe them. I have held a broken or damaged SM58 microphones in my hand before but have never actually witnessed any just cease to even work a little bit. On top of that, you can get it for $99. So not only is it super durable and sound great with most people, it’s probably one of the most reasonably priced microphones on the market. 

So how do I use it? Let me tell you. For me, every mic is situational. Quite often new people walk across the stage I work with. Most of them are speakers so it’s a headset battle but with vocalists it becomes a full out capsule war. Now while I don’t usually start with an SM58, but if I’m just having a hard time matching the vocalist with a capsule, it will never be long before they’ll be singing into a 58 just to get back to a baseline. This is the biggest thing I use the mic for: getting a baseline. So many times I’m just not sure which direction to go with the rest of my arsenal so I’ll just pick up a 58 and use what I hear with this mic as guidance. How their voice sounds with little correction with an SM58 will guide the next step. On top of that, in at least most cases, the SM58 is not a bad choice for a microphone. But there are a few instances when I use it first. The biggest first use for me is for really quiet singers (or soft spoken speakers). The rejection for this mic is pretty great and it’s a cardioid microphone giving it a forgiving pickup pattern. As they gain confidence and in turn volume, I might switch off of it for a different capsule but maybe not. There is just something to be said about being consistent about using the same microphone with the same person so you can start building up memory about what you’ll need to do each time they sing. But, that boost in the upper mids can be detrimental, in my opinion, for female vocalists. Everything must be weighed when considering which capsule to use. 

Well, that’s about it for this week. But I want to hear back from you. In the comments below or on facebook, tell me the weirdest thing you’ve ever mic’d with an SM58. I want to know if my weird stories are the norm or an exception. Don’t forget to come back next week when I’ll talk about the capsule I use first on any male vocal and compare it back to our series baseline, the SM58. If you want to be emailed when any new content goes live, follow this link, and subscribe to my blog. You won’t regret it! See you next week!

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!