Author: Justin Fugett

Posted on

Gear Talk: Audio Distribution Part 2

Welcome back to our Gear Talk series about audio distribution.  Last week, we talked about the various types of splitters, patch panels, and snake connectors, but this week we’re taking a moment to stop and understand how digital audio consoles process their audio and how that affects connecting audio consoles to each other over digital formats.  Even if you only have two consoles to connect understanding the various factors involved is vital.

One of the things I often come across when working in broadcast land is the need to bridge multiple audio consoles (or just audio devices) through digital networks.  In broadcast applications with consoles that can support multiple formats and thousands of inputs and outputs it is something I have become fairly familiar with and I have grown to understand most of the popular digital audio formats in use today.  When connecting multiple digital audio devices (consoles) together one thing that you MUST ALWAYS be aware of is clocking (i.e. word clock). The word clock is basically an audio device’s heartbeat. It digitizes the data being brought in from it’s analog to digital converters all together at the same time.  When we connect or network digital consoles, we need to make sure they’re all clocking the audio bits together. If the consoles aren’t all clocked together, then those consoles will fall out of sync and artifacts or even dropouts in audio between the consoles will appear. When networking multiple consoles it is imperative to make sure that you have your clocking set up and working properly. The best way to do this is to have a clock master console or device.  This is a device that all other digital audio devices synchronize with. Most broadcast houses have a specialized “house clock” that handles both video and audio synchronization all at once so. In most cases, where only two or three consoles are being connected, you can simply designate one console to be the clock master. Designating a clock master is like syncing watches back in the days before smart watches. Most analog watches can keep great time, but if a group of people get together to do something very time sensitive, then they need to make sure all their watches all agree that 9am is 9am and that 9am starts at the same second for everyone at once.  Once you’ve understood that all your consoles must be clocked to a single source, you can now begin to understand digital audio distribution a lot quicker.

Another item to consider is the processing rate, or sample rate, of the audio consoles involved.  Most audio consoles in use today have a sample rate (the rate at which the analog audio is sampled (i.e. digitized) per second) of 48K but 96K is catching on and becoming much more popular and some consoles even have sample rates as high as 192K.  When thinking about audio processing there’s actually two factors to consider, one the sample rate which basically refers to how fast and how much your console processes the audio coming into it. The bit depth, which is often given in conjunction with the sample rate is how much audio your console can process at the given rate.  Think about sample rate as the speed limit on a highway and the bit depth the lanes on a highway. You need both a fast highway and a lot of lanes to move traffic smoothly. In this sense, sample rates and bit depths depend on each other. Most modern audio consoles today have bit depths of 16 or 24 bits with some manufacturers offering 32 bit I/O modules for more detailed sound.  This means most consoles have a great “highway” with plenty of “lanes” (bit depth) and lots of “speed” (sample rate) to move our audio along and process it efficiently and truthfully. While bit depth isn’t often a huge deal when networking consoles, the sample rate is very important as most digital networking formats have a hard time converting sample rates when transporting audio between consoles.  This is because a console running at 96K samples or processes the audio twice as much as a console running at 48K in a given moment of time The de facto standard to most digital networking formats is currently 48K because the broadcast world is still in 48k due to the huge complexity of their systems and interconnected technology. Live sound consoles vary much more based on manufacturer and user input.

Making sure all your audio consoles can communicate properly all comes down to sample rates and clocking.  Even consoles with the same sample rate may not be sampling the audio at exactly the same moment. Without making sure all the shared audio is being sampled together, the audio can be ruined.  It is important to understand the role of a master clock and why one is almost always needed when two digital consoles are connected together. Next week we’ll dive into some of the more common formats in use today as we continue our audio distribution primer. Be sure to drop a line below or on Facebook if you have any questions or drop an email to engineers@studiostagelive.com. If this article has sparked your interest in this topic be sure to follow this link and register to receive an email when new content is published. See you all on the flipside!

Posted on

Gear Talk: Audio Distribution Part 1

These days more and more churches are growing their audio setups.  Some, want to do broadcasting to an online audience, others want to grow their worship team by adding a monitor consoles, while others may want to send all or parts of their mix to another space, still others may want to do all of the above.  This leads us into the wide world of audio distribution. In the upcoming series we will take a look at all the different forms of audio distribution from digital to analog and what their abilities and drawbacks are. For this first week, we’ll primarily stick to analog as it will build the foundation for the digital formats to follow.  As we have touched on in the How to Mix for Broadcast series, audio splits can be very useful and come in both analog and digital flavors.  This new series on audio distribution will cover both the analog and digital variants. We will also cover the various pros and cons of each type and of the special needs of some of the digital distribution methods.

Before we dive into the wide world of digital audio formats, it’s important to understand the analog variants many digital audio formats are modeled after.  The world of analog audio distribution is just as varied as digital distribution. Just like digital audio, there are a bunch of different formats that carry analog audio in different ways.  Analog audio distribution is done with splitters. While there are many kinds of splitters, the simplest is just a Y-cable. In fact, all big splitters are basically just multi-channel Y-cables.  Some have extra features and some splitters split more than just two ways, but all analog splitters, at their heart are just simple Y cables. Analog splits come in many formats and channel counts.  A simple 1:2 all the way up to 64:3 are common. Some of the more common formats of split are transformer isolated and groundliftable. Transformer isolated splits are the preferred method for large format splitters.  They work by splitting the audio 2-3 ways. One output is a direct Y cable split. The other output or two outputs are wound around a transformer in a 1 to 1 ratio to remove any DC voltage. This makes the transformer isolated outputs much less susceptible to any interference but has the effect of also isolating the outputs from phantom power.  This is mostly a protective measure to ensure that multiple consoles aren’t sending phantom power at once which could damage some older phantom powered microphones. The remedy for this when using a transformer isolated split is to make sure the console connected to the main output is able to phantom power any and all channels needed on the split.  Professionals have used splitters to allow any console using the splitter to their own gain settings instead of having to share and digitally trim (there were issues with doing this in the early days).

Another area of the analog realm that carries over into the digital arena is patch panels.  Analog patch panels are a really useful tool in live sound. You won’t typically find patch panels on tour setups but they’re still very ubiquitous in venues, even venues with digital consoles.  Patch panels are also still heavily in use in the broadcast world because of the flexibility they provide. Most audio patch panels in use today use the TT connector. It’s kind of a cross between a 1/4in cable and a mini cable and is a leftover of the old analog phone days.  TT actually stands for TinyTelephone and was primarily used by AT&T phone switchboards. However it is a fully balanced connection and is nice because its so small you can fit a lot more connectors into a block of TT patch panels than if you used XLR or even 1/4in connectors.  Like splitters, not all patch panels are created equal and different patch panels have different abilities. Patchbays can come in three variants. Normal, half normal, and non-normal. All of these have different characteristics that make them useful in certain situations. Normal patch bays are setup to work when nothing is plugged into the top or bottom.  They are wired so that whatever is plugged into the top port, flows to the bottom port. Plugging something into the top will break the connection and route the audio away from where it was originally flowing. Half normal patch bays work similarly except that when you plug something into the top port, it splits it and will continue to flow to the bottom unless you plug something else into the bottom port.  It is essentially a Y splitter. Non normal patch bays NEED you to connect the top and bottom ports to work. These are typically used in spaces that have much more physical I/O than console channels. The user can decide what goes where for each use. All of these options have their uses in certain cases and are a great option depending on your needs.

One more area of the analog world that I would like to touch on is snake connectors.  There are a lot of different analog snake connectors floating around these days but I’d like to touch on some of the more popular ones in use today.  One connector that sees a lot of use in studios more so than live environments is the DB25 connector. This looks loosley like the letter D and has 25 pins.  This allows for 8 channels of audio to pass through it. The DB25 connector is often seen on the back of patch panels, on a lot of recording gear, on the back of some Digital I/O modules where space is tight.  Be careful as there are a couple different pin-outs. The main two are TASCAM and YAMAHA pin-outs. It’s still 8 channels of audio but the two versions have differences on what pin is ground. Another connector is the CPC connector.  This is a circular connector that houses at least 8 but potentially 56 or more channels. CPC connectors are ubiquitous all over the place on tours, in theaters, and Houses of Worship (HOW) and are very useful. CPC snakes are a little scary because there is no one correct pin-out for them so you have to be a little careful when fixing one or changing one.  CPC snakes are cool because the CPC format can be made to fit just about any need and when paired with a patch panel, it can be extremely flexible. The last of the analog snake connector types is DT12. DT12 is a standard that carries 12 channels of audio in a round connector similar to the CPC format. What’s nice about DT12 is it conforms to one standard pin-out and has a VERY rugged connector.  DT12 is primarily used by OB (Outside Broadcast) trucks and other broadcast applications. It’s great for its size and weather proofing. While there are a ton of other snake connectors, these are the main players on the market today.

One last analog snake format that has become VERY popular in the past couple of years and can be very useful in spaces where there is a lot of cat5e but not a lot of traditional copper, is the analog audio over ethernet snake.  Analog over ethernet snakes utilize the 8 twisted pair cables inside a shielded cat5 cable to run 4 channels of audio with a shared ground over the shield. These can be extremely useful in situations where there is more cat5 than traditional copper.  A lot of companies make many versions of these. Some make boxes you can attach on either end of the cable, others make pigtail versions, a couple companies even make wall boxes. Most of them allow you to daisy chain snakes so you can multiply your four channels down the line.  Whirlwind, RatSound, Radial, and ProCo all make these.  Most of these manufacturers make DMX versions as well.  One thing to note is that a shielded cable is required for phantom power.

Come back next week when we dive into digital networking and all the different formats in use today.  We’ll also touch on best practices for digital networking and some ways to make sure you don’t get into trouble when connecting multiple consoles together. Feel free to ask any questions that came to mind below, on facebook, or by sending an email to engineers@studiostagelive.com. If you want to be notified when a new post is available, register at this link to receive an email when something has been published. Happy mixing!

Posted on

How To: Mixing for Broadcast Part 3

For the last two weeks we’ve talked about crafting a broadcast mix for your church and the various ways to set it up, whether it’s with your current FOH console or whether it’s with a separate console for broadcast. We’ve discussed analog vs. digital splits and now we come to the fun stuff. Actually mixing content. Now that we’ve got our mix options set up, whether it’s a feed from FOH or a split to a separate console, we need to get a good mix crafted. Before we dig into the good stuff, it’s worth noting that you should always keep track of the speech mics in the field and whether or not they should be on. If a speaker leaves his/her mic on while in the audience you may not hear anything different in the house but in the broadcast it will make a huge impact. This is another great reason to utilize automation to help you easily conquer the little stuff.

So, broadcast mixing typically needs to be a little more precise than FOH. Because we don’t have a large room to hide mistakes in, we need to be very conscious of various instruments and singers and how they’re blending with everything else. Typically I tend to mix the band a little heavier than vocals when doing broadcast because even with reverb and a band, it is often very easy to hear mistakes made by vocalists when singing. They’ve got one of the hardest instruments to control and keep in tune and even the best singers in the world aren’t perfect so I actually keep them in front, but not overtly so. I’m also constantly on the lookout for band members who may be struggling. You’ll also want to try different reverbs out and see what works best. Typically reverbs that work in an auditorium don’t work as well in broadcast. Try to find something that fits your mix and doesn’t tail too long. Too much reverb can render a vocalist unintelligible and it’s much easier to overdo it in broadcast than FOH. Mixing for broadcast is very different from mixing a room. As discussed earlier, it’s mainly different because you’re not dealing with interactions from the room and PA nearly as much as FOH has to. It’s also not quite like producing an album. It’s a blend of studio work and live sound. Broadcast mixing has it’s own challenges.

One main area broadcast differs from FOH is in EQ. EQ for broadcast is much easier in many cases because you aren’t fighting the room. This is where plugins can make a huge impact. If you’re using Waves you can use plugins like the F6 to dynamically carve or add frequencies back into channels, you can use the C6 or C4 to dynamically compress or expand bands of frequencies. We previously spoke of the fact that most listeners will be listening on phones and tablets and this is another area that comes in to play. We need to make sure we EQ in such a way that things sound pleasant on all devices. Because of that limitation we have to make sure we don’t have too much low end on channels that don’t need it and we should keep from having too much in the low-mids because that will really muddy up a mix on smaller speakers. We also have to remember that we aren’t filling up a big room, just a small pair of speakers or maybe headphones, so everything needs to be EQ’d as tight as possible. That being said, a lot of corrective EQ you might instinctively reach for at FOH, is not needed in broadcast. Mostly just watch your lows and low-mids. Inserting a plugin with an RTA on it like the F6 or HEQ can really help to monitor this as well.

Panning is another important tool in shaping a good mix for broadcast. When we’re listening on headphones or a small device, similarly to listening to a PA in a room, our brains need help to pick out all the various inputs in a mix. Panning can be a huge help in this area. With drums, I like to pan them from the perspective of the drummer so that the sound of each individual drum matches the sound of that drum in my overheads. This helps keep the sonic image clean but also sounds really cool on a mix when panned from left to right. It also helps with visuals if you have camera on the drummer and he plays from one side of the kit to the other. I’ll also pan instruments based on stage location and what they’re doing and I change it up from time to time as well. Singers are also usually slightly panned with any leaders being dead center. Be careful not to pan singers too drastically as it can cause certain singers to stand out too much. I also typically keep either the keyboard or acoustic guitar center depending on which instrument is the anchor for a particular song.

One other important part of the broadcast mix is audience mics. If you can have them at all, I would recommend it. It’s best to have them hanging in line with your PA and pointed at your main audience area but even choir mics hanging halfway back is better than nothing. Being able to get claps, cheers, laughs and singing from the congregation connects the broadcast audience with the audience in the room. The best place for an audience mic is either the lip of the stage or hanging over the lip of the stage pointed at the audience. Most broadcast guys go with the a shotgun mic like the Sennheiser 416 and will place anywhere from 2-4 depending on the room. I’ve seen as many as 8 handling a large crowd. If you don’t have the budget for new mics utilizing a condenser or even dynamic mic you already own will be better than nothing (just anything cheap and omnidirectional like this pair on Amazon). Try to get it as close to in-line with the PA as possible so you don’t have weird delay issues. Nothing sounds better than a well placed crowd mic picking up the congregation singing as the worship team falls back. Crowd mics can also be used to feed in ears for musicians so they can serve more than one purpose.

Unlike mixing a live PA where dynamics are incredibly important, in the broadcast world we must try to do a better job of keeping the loudness more uniform. If people have to turn the volume way up for the sermon or can’t even hear the sermon, then we’ve only done half our job. If you’re using your FOH console to mix you can accomplish this by using a post fade aux and turning all your musical channels down more and your speech channels up more, or doing the same with subgroups. For live events on broadcast on TV, your overall loudness is supposed to be within +-2db of -24LKFS (this is a measurement standard use to standardize perceived volumes across all devices) for the length of the program. While we aren’t necessarily required to do this for streaming online, I like to try to follow it as a good rule of thumb giving myself a range of +-3db LKFS. Getting consistent loudness on a separate broadcast console is a little easier and we have some better ways to do it as well. One of those ways is by using the WLM Meter plugin in waves to monitor (and if needed limit) the loudness of your mix. There are also several outboard options like this one that just allow you to monitor. The basic principle is to check three things: 1) level match speech vs music, 2) be sure to check for headroom across your entire signal chain, 3)If nothing else, take a recording of your mix and analyze it to see how you line up with the LKFS standard. No one is going to be angry that your mix is too quiet or too loud but getting things setup correctly can really contribute to the overall quality of the art you create.

At the end of the day, the best way to craft a mix is to just work at it. Like playing an instrument it takes time and practice. The more you do it the better you’ll become. Hopefully this series was a good primer and a first step towards better crafted mixes. If you have any questions or thoughts, please drop a comment below or on Facebook. As always, if you’ve enjoyed reading this blog, feel free to sign up at this link to subscribe and be notified when new content is available. We hope to see you next week!

P.S. – If you didn’t already notice have a new author, Justin Fugett! To help you get to know who is behind the scenes here I’ve launched a new page on the site with bios for each author. Be sure to check it out at this link!