Sometimes you’re setting up in a room that’s shaped like a shoebox; very wide but not very deep. This means sound from your speakers is probably slapping off the back wall and creating unintentional reverb that can kill the intelligibility of speech for both listeners and presenters. Not good. So, how do you fix it? You can’t reshape the room, but you can consider changing your speaker placement if that’s an option. Let’s say you’ve got two clusters for mains (4 speakers) and two side fill clusters. That’s 8 speakers total but only 2 zones. Your goal is to be able to decrease the volume enough that sound dissipates before reverberating off the back wall. But here’s the problem: you won’t have even coverage with your current setup. Solution: separate your clusters and spread out your speakers so that you have no dead spaces i.e., more zones. Then decrease each speaker’s output gain until you find the best volume for reaching the listeners in the back without generating too much room reverb. Or better yet, think about the shape of your room beforehand so you have the correct setup in the first place! Summary: Wide, shallow rooms that are prone to excessive room reverb require multiple zones of point-source audio. Side note: you also want to keep your speakers away from walls as much as possible to avoid the boundary effect.

Let’s look at: - Which Listening Environments Need Delays? - What Problem Do Delay Speakers Solve? - What Problem Do Delay Speakers Introduce? - How Do We Solve These Problems? Which Listening Environments Need Delays? Delay Speakers are only necessary in very deep rooms, where in order for the people at the back of the room to hear just as well as the people at the front of the room, you (the sound tech) have to turn your Main Speakers up past their ideal volume until they’re unreasonably loud for the people in the front. What Problem Do Delay Speakers Solve? The main goal of sound reinforcement is to create a listening environment where sound is distributed evenly and with the same acoustic qualities, which ensures that every listener gets the same listening experience regardless of their listening position in the room. So obviously, if the room is very deep and we only have one line of speakers in one depth position (our Mains), we’re going to run into the problem I just outlined where either it’s too loud for the people in the front or too quiet for the people in the back. This is the problem that delay speakers solve. By placing speakers in different depth positions in the room, we can have speakers that are placed closer to our listeners in the back, which allows us to maintain an ideal volume for all of our speakers and all of our listeners. What Problem Do Delay Speakers Introduce? However, delay speakers also introduce a problem that we didn’t have before. The problem is asynchronicity. In order to explain this problem, I’ve got an analogy for you. Let’s say we’ve got 3 people in a racetrack and they’re going to have a race. Person 1 is 100 feet from the finish line, Person 2 is 50 feet from the finish, and Person 3 is 25 feet from the finish. Assuming they all start at the same time and they all run at the same speed, will they cross the finish line at the same time? Of course not. Person 2 gets to start closer than Person 1 and Person 3 gets to start closer than both of them! Based on the current rules, Person 3 will finish 1st, Person 2 will finish 2nd, and Person 1 will finish last. Another way of saying this is that they will run asynchronously. They won’t move through the same space at the same time.

This is a fantastic tutorial for learning to use Shure Wireless Workbench for setting up your wireless network, scanning, and deploying frequencies.

Hey everyone, I thought I would share a conversation I was having with a coworker today. They asked me why you’d ideally want your faders near unity. Here’s my response: 1. Granular scale of control. Faders are adjusting decibel levels and the decibel is not a linear unit of measurement. Every 3dB increase or decrease of gain represents a doubling or halving of signal strength. Input faders adjust the decibel level of an input signal that among other potential destinations, usually gets sent to your master output or matrix. Your master fader(s) adjust the decibel level of your output signal that gets sent to your speakers or some other destination (e.g., a recording). The fader positions are also not demarcated linearly, which means that there is an equal fader distance between -10dB to 0dB as there is between -60dB to -30dB, but as you may have noticed, the former is a difference of 10dB, while the latter is a difference of 30dB. This means that any changes you make with your fader between -60dB and -30dB will be drastic, while changes made between -10dB and 0dB will be more subtle. Subtle signal adjustments are ideal because sudden drastic changes can be startling for listeners and lead to unintended signal levels for operators. 2. SNR. In both cases, your fader position determines your ‘signal to noise ratio’ (SNR), which is the ratio between your desired signal and unwanted noise. In every signal there will inherently be some amount of noise. The threshold at which this unwanted noise is audible is called the ‘noise floor’. The noise floor is both an absolute and relative value. It’s absolute in the sense that there is some signal level at which noise is audible, which is a fixed value. But it’s relative in the sense that it is lower or higher relative to your signal level, in the same way that the floor of a room seems lower or higher relative to the height of the ceiling, even if it’s just the ceiling that’s changing. So why do you want your fader to be near 0dB (unity)? Because this is the ideal signal to noise ratio. It means that the same absolute amount of noise will be masked relative to the strength of the signal. It means the noise floor is lower relative to the signal level. More signal clarity. Less noise. Important for input faders because they affect the outputs that are receiving input signals, and important for output faders because they affect the speakers and other devices that are receiving the output signals. This is a crucial concept in every audio application but especially recordings.

I’ve been reacquainting myself with Q Lab as I’m going to use it for a gig this month doing some interactive audio/LX for a Halloween Party 🎃 So, I figured I might as well share some tutorials with all my fellow audio wizards! 🧙🏻‍♂️ Q Lab is a great tool for programming audio cues and stingers (specific audio fx or songs you want to play at specific times in the song, specific times in your show, in a specific order, you get the idea, it allows you to edit and trigger audio in a specific way). Similar software tools include Playback Pro, Mitti, Sport Sounds, etc. but Q Lab is arguably the best at this point, or at least the most versatile. Q Lab also allows you to edit and trigger videos, or send MIDI cues to your lights and synchronously trigger audio cues as well, which is what I’m doing at the Halloween Party 👻 It has a bunch of other features and use cases too. But assuming you know nothing about Q Lab, here are some beginners tutorials! https://youtube.com/playlist?list=PLjXnSOcWbDXuAww6sWQxrFo2vzjZ9x0Aw&si=7ylNN7SVqzLEsIHo