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  • Writer's pictureMatthew Losey

Go with the Audio Signal Flow: Part 1 | How-to #4

Updated: May 2, 2023

There are many different flavors of audio. There are hundreds of genres and millions, if not billions of sounds available to the modern musician. The addition of computers and other music reproduction technologies has made it more accessible than ever. Regardless, music in a modern environment takes about the same path regardless of its application or budget. Sure, digital processing adds a step or two over analog processing. But whatever methods we use to reproduce music, we audio engineers are in charge of using it to the best of our ability. To do that well, we also need to be educated on how to utilize these tools.

I have divided the sections of the signal path so we can talk about each one individually and how it affects everything down the audio line. If we want our audio to be as excellent as possible, we have to know that our choices in audio gear matter. If we know our options have weaknesses, we can play them to our strengths. We used to be taught that we need to work on our weaknesses and have as few as possible. The older I get, the more I realize that idea is an immature way of looking at our world. We can’t get rid of every weakness. However, we can make our strengths so strong that our weaknesses don’t seem to matter anymore.

Everything in audio is about balance, including the money you have to spend on that audio gear. Balance that money in the right areas and you will have an excellent product.

The five parts of the audio signal flow are signal origin, AD conversion, processing, DA conversion, and finally amplification. In this post, we will be talking about the signal origin and the first AD conversion process.

Signal Origin

This will probably be the most debated portion of the audio signal chain as long as microphones exist. There are thousands of available microphones on the market today and as long as we are focusing on live sound, most of them are dynamic microphones. If most, if not all, of the microphones in live sound, are dynamic type microphones, what are the differences?

Well, just like there are cars for everyone, there are microphones for everyone too. Some cars are fast, some are slow, some are meant to pull, some are meant to push, some can crush and some are meant to be pretty. Microphones are designed for different applications too. Microphones can be built for quiet audio, some are meant for excessively dynamic ranges and some are meant for high-volume applications. Some microphones are built like tanks, meant to be thrown around onstage and others are very fragile meant only for recording environments.

Ok, let’s get down to more specifics on how to use microphones in live environments. Signal origin is important. We know that. I could sing into a $5,000 microphone, and Beyonce could sing into a $100 microphone, and she will still sound better. With that being said, there is a point of diminishing return for money invested in microphones. Almost any wired microphone that costs more than $150-200 is going to be adequate enough for live sound. That is especially true for trusted brands such as Shure, Sennheiser, Blue, and Audio-Technica.

I’m sure I missed a few of the reputable brands, but steer clear of trying to pinch pennies looking for ways out of spending money on good microphones. They just sound bad and you end up doing way more work trying to get them to sound like the microphones you should’ve bought in the first place. Reputable microphones do cost more monetarily but they also cost you less time. Money isn’t the only currency to consider in audio engineering.

One last thing on microphones. Please don’t use omnidirectional microphones when trying to capture audio and amplify in the same room. Along with getting feedback, you’ll just get frustrated.

The other half of live signal sources are those that come from DIs. DI stands for Direct injection boxes. You have bass guitars, acoustic guitars, keyboards, computers, or drum machines all plugged into DI boxes. DI boxes all attempt to do the same thing. They take unbalanced or balanced signals and prepare them for balanced signals by attenuating the gain, grounding the wiring, and utilizing a typical XLR plug to transfer the signal to the main Front-of-House board.

Just like microphones, this can also be done cheaply but you never want to do that. Cheap DI boxes use substandard components in the interest of saving you money and do nothing to improve your sound or save you time. They may even cause damage to your microphones. More expensive DI boxes, like the ones made by Radial, have circuitry that protects equipment from users accidentally adding voltage down the line by turning on phantom power unnecessarily. While most modern microphones have circuitry to protect themselves, computers, keyboards, guitars, and basses do not. Those things will fry and be costly to repair.

The point I want to get across to churches is that it’s not money that gets you a good sound. It’s firstly good talent and source material produced by the musician. After that comes good equipment to transmit that sound. It just so happens that good equipment tends to cost good money.

When push comes to shove, if I had $5,000 to spend on a sound system, I would rather spend it on microphones, mic cables, and DI boxes. Then I would spend the last $500 on an old analog board to mix it all with little processing power. That would sound better than a $3,000 digital board with only $2,000 spent on microphones, DI boxes, etc.

Once we have that initial conversion from vibrating sound waves through the air to electrical impulses, how do we get them to the board? The short answer is through cabling. The two primary cables involved are a TRS cable and an XLR cable. Both utilize a 2-wire plus ground system (positive, negative, and a ground). XLR is still considered king though. The nature of the format of the actual plug means there will be no clicking or popping when you plug or unplug the wire. Because of how a ¼” TRS cable is received into the female receptacle, the channel needs to be muted. Otherwise, each receiving contact will pass over the tip and ring of the connector, which makes for an unpleasant sound, especially if it is transmitting voltage. You get a lot less of that with XLR cables.

Either way, once you have the balanced cable, it can travel over incredibly large distances with virtually no noise introduced for “science” reasons. All you need to know is that it works brilliantly to get the audio signal to the board or digital snake with minimal losses. The digital snake is the final stop for analog audio before it gets converted and processed by the Front-of-House board.

Analog to Digital Conversion

This audio conversion is called an AD conversion and is the topic of a lot of debate as well. AD stands for Analog to Digital Conversion. The only thing I’ll say about the subjective qualities of AD Conversion is that when it comes to sound quality and clarity in the conversion, you get what you pay for. Other than that, we will stick to the basics of the technical settings and not get into the minutia of manufacturer preferences.

One of the first settings you have to set on your digital board is the sampling rate. In the good ol’ days of analog, sine waves are not sampled and were inimpeded curves. Since the introduction of digital audio, there are necessary steps to be taken to convert the sine waves into bit forms (I and O). The first became an issue on tape machines and the sampling rate was changed by how fast the tape would spin as it captured audio. They landed on 44.1khz, 48khz, and 96khz all the way up to 192khz. This means that if I chose 48khz, the sine wave is sampled 48,000 times each second. The sum of all those samples, with the addition of noise (called dither), added in between the samples, gives our ears the audible representation of a pure sine wave in digital form.

So which is best? You might think the highest possible sample rate is best. You’d be correct. But high sample rates come at high prices as well. So what’s the highest feasible sample rate? In my experience 48khz is the lowest sample rate that professional digital boards will come in and I’ve had zero issues with noticing any differences between the 48khz and 96khz sample rates. An added benefit of running at 48khz is the ability to have twice as many inputs and outputs.

The last aspect of the sample rate is bit depth. In live sound, you usually don’t even get bit depth options but if you decide to record your performance, 24-bit is best if 32-bit float isn’t available. I’ve been recording for years without issue at 24-bit but if bigger is better, go for it. The audible differences between 16-bit and 24-bit are very noticeable. The differences between 24-bit and 32-bit are not as noticeable, especially in a full mix. Practically, the primary difference is in the depth of the information captured. I know I just used the defined word in the definition but you get a more effective dynamic range for your audio. 16-bit has a theoretical dynamic range of 96db, 24-bit has 144db, and 32-bit has 1,528db. Having a bit depth of 32-bit is probably excessive here but how much effective dynamic range you actually have is dependent on the noise floor of your converter in relation to bit depth. In as few words as possible, the lower your noise floor, the more of your dynamic range you get to use.

If you don't know what noise floor is, have you ever tried to gain up a channel enough to hear the instrument but you are getting a lot of noise with it? That’s the noise floor. When your noise floor is too high, it’s an unusable signal. The goal with analog and lot bit depth systems is to gain your channel low enough where the audio doesn’t clip but high enough to maximize the separation of the audio signal from the noise floor.


This is only about half of a very brief, surface level of the understanding of what is going on behind the scenes of your digital soundboard. Hopefully, after you read these couple of blog posts, you will be able to understand the first few pages of the manual and make an educated purchase decision. If this was too long and you didn’t read it here are my points…

  • Spend more money on reputable microphones

  • Don’t buy cheap DI boxes

  • Stop using instrument cables when you should be using balanced audio cables

  • Read the manual of your soundboard before you buy it

  • Sample rate isn’t everything and bit depth isn’t usually an option for live sound

  • Consider your channel count and make sure it fits your system


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