Monday, July 27, 2020

EQ Tips for Keys, Pianos and Synthesizers

When it comes to EQ’ing keys, there is a big difference between what you will do for electronic keys/synthesizers and acoustic pianos. So, let’s go through both.

Keys & Synthesizers
Electronic keys don’t often need a lot of EQ, but you can fine-tune the sound by adjusting a few fundamental frequency ranges.
Tip: Clean up the muddiness.
Keys and synths can get a bit muddy in the 400-600Hz range. Use a peaking filter with a Q value around 4 in this area to clean up the sound, especially when it is layered with other instruments.
Tip: Help the keys ‘cut through’ the mix.
If you need the sound to cut through the mix, try boosting slightly in the 1-2 kHz range. Or, you may even need to cut this area to make room for other instruments.
The 3-4 kHz range is where the primary presence and clarity can be found. Boost this area a little if necessary. Or, you can cut this area to make the tone a bit darker.

Acoustic Pianos
Regarding pianos, there are many different types and sizes that will have a range of tonal properties, so these tips will depend on size, playing style and miking techniques.
Tip: Cut the boominess.
Pianos can tend to be really boomy in the 100-200Hz range. The best way to fix this problem is with a low shelf filter at about 200Hz and cut 3-6dB.
This can also help reduce feedback or other low end resonance from drums or nearby instruments on stage.
Tip: Brighten the tone of the piano.
You can brighten the tone and help the piano cut through the mix by using a peaking filter in the 3 kHz range with a Q value around 4 and applying a slight boost.
Keep in mind, too much emphasis in this range can exaggerate distracting elements like damper and string noise.

The Ultimate Guide for Choosing Audio Cables



Does your sound system buzz, hiss or make some other kind of unwanted noise? The problem is most likely rooted in bad cables.
Regardless of how good your mixer, amps or speakers are, you will never get a good sound if you don’t have the right cables.
You don’t need the most expensive cables to achieve the quality of sound you desire. But, you do need to see cables as something worth investing in.
Below, you will find the cables I recommend. I have used these in my church for years and I am very happy with the results.

Balanced vs. Unbalanced Cables

Unbalanced cables have two wires: one that carries the signal and a common ground. They are also shielded to help protect from unwanted hum and other noise. Instrument cables are typically unbalanced as well as many patch cables. They are used to connect high-impedance signals (guitar, keyboard, guitar pedals, etc.) to an amp, mic preamp or direct box. Unbalanced cables should not be longer than 25 feet.
Balanced cables have a third wire that also carries the signal, but in reverse. This causes any unwanted noise to be eliminated, making them ideal for longer cable runs. Balanced cables generally have either a XLR or TRS connector.
While in a few cases balanced and unbalanced cables are interchangeable, it is the output source you are connecting that will determine which type is needed. For example, microphones usually require a balanced cable as do direct boxes.

Direct Box: Converting Balanced to Unbalanced

Recommended Direct Box: Radial Pro DI Passive Direct Box
In short, a direct box converts an unbalanced signal (i.e. from a keyboard) to a balanced signal so it can then be sent over a long cable run back to your mixer. Considering your keyboard player is probably more than 25 feet from you mixer, a direct box is your solution to getting the signal back go the mixer without unwanted hum or noise.

Common Connectors

 TS (aka 1/4”)
TS connectors come in 1/4” and 1/8”. They are used for unbalanced operation with instrument, patch and speaker cables. The easiest way to tell them apart is they only have one ring (usually black) on the shaft of the connector.

 TRS
TRS connectors come in 1/4” and 1/8”. They are used for balanced or stereo operation with microphone or patch cables. They have two rings on the shaft of the connector, making it easy to differentiate from a TS connector.

 XLR
XLR connectors are used for balanced operation with microphone cables. They are most widely used with microphones, but you will also find them on balanced patch cables and DMX lighting cables.

 Speakon
Speakon connectors are used to connect amps to speakers with speaker cable. They are becoming increasingly popular because they lock into place and therefore cannot become accidentally disconnected.

 Banana Plugs
Banana plugs are used to connect amps to speakers with speaker cable. You will most often find them on the end that connects to the amp, which enables you to easily set up bridge mono operation.

Instrument Cables
Recommended Instrument Cable: GLS Audio Instrument Cable
Instrument cables connect a guitar, bass, keyboard or other electronic instrument to an amp, preamp or direct box. They carry low-voltage signals and generally have 1/4” connectors. They are considered unbalanced cables, which means they are susceptible to noise and should be kept as short as possible (certainly under 25 feet).
An instrument cable should never be used as speaker cable. If used as a speaker cable, the quality of sound will suffer and the cable may get hot enough to melt the jacket.

Microphone Cables
Recommended Microphone Cable: GLS Audio Microphone Cable
Microphone cables are generally understood correctly, mainly because of their unique XLR connector. They are shielded and balanced which effectively keeps unwanted noise at a minimum (especially for long cable runs). They are used to connect microphones, direct boxes and other low-impedance signals to the mixer.

Patch Cables
Recommended Patch Cable: GLS Audio Patch Cables
Patch cables are used to connect various audio components together. They are generally short in length and may be balanced or unbalanced, depending on their application. For example, when connecting a series of guitar pedals, you will use unbalanced patch cable considering the signal coming from the guitar is also unbalanced.
Patch cables come in all shapes and sizes. You can get almost any kind of connector, including XLR, 1/4”, TRS and RCA. When choosing a patch cable, you must first determine if the audio source is balanced or unbalanced. Then, it is just a matter of getting the right length and connectors. Keep in mind; the only balanced connectors are TRS and XLR. The others indicate an unbalanced cable.

Speaker Cables
Recommended Speaker Cable: GLS Audio 12AWG Speaker Cables
A speaker cable is an unbalanced cable that has a much heavier gauge (more wire) than most other audio cables. The reason they need heavier wire is because they carry a much higher voltage than microphone or instrument cables.
Many confuse a speaker cable with an instrument cable, especially when they have a 1/4” connector on both sides. However, they are not the same. An instrument cable is not capable of handling the high voltage that is pushed from the amp to the speaker. It may seem like it is working, especially at low volumes, but your amp is having to work extremely hard to push the signal through such a small conductor. Eventually, the instrument cable will heat up, melt the jacket and cause a short circuit – potentially ruining both the speaker and the amp.

Proper Speaker Cable Gauge
When buying speaker cables, the gauge of the cable becomes an important factor to consider. Keep in mind, the lower the number the more ‘thick’ the cable is. So, a 12-gauge cable is thicker (has more wires) than a 14-gauge cable.
A cable that is too ‘light’ (gauge is too high) will result in amplifier power being wasted. It will also cause a loss of low-frequency performance. On the other hand, the only downside to having a ‘heaver’ cable than you need is excessive cost.
Following is a chart to help you determine what speaker gauge you need. I have found that 12-gauge is most often the appropriate choice for church sound systems.

How to get started with a click and backing tracks for your worship band by Jake Gosselin

Worship Leading

I led worship for years without using a click track or backing tracks.

I used to think things like:

“It limits musical freedom.”

“Our band members are not skilled enough to play with a click.”

“It sounds too ‘produced’”

“I do not have time in my weekly preparation for a click and tracks.”

Those were some of the thoughts and objections that ran through my head as I considered whether or not it was worth implementing a click and backing tracks with my worship band.

Then I finally arrived at a place where I felt like I was stagnating as a worship leader and I wanted to challenge myself and my band. That’s when I started doing some research to learn what type of equipment and knowledge I would need to begin using a click and backing tracks in worship. 

I knew I was going to need in-ear monitoring, and I heard of software like Ableton Live, but it all seemed intimidating. How was I ever going to convince my band, which consisted mostly of people 20 years older than me, to play with backing tracks, let alone start using in-ear monitors?

Whelp, both of those things happened. It was not easy and did not occur overnight, but in a couple of months I had my whole band on in-ear monitors, playing with a click and backing tracks, and I even began automating ProPresenter and lighting. My only regret is that I did not learn to do these things sooner!

A lot of worship leaders struggle with “staying fresh.” Playing mostly the same songs every week in the same venue for the same people can get creatively stale. I also have a firm conviction that the local church should be a wellspring of excellent and inspiring creativity that makes a powerful impact on the unchurched. 

When I learned how to use a click and backing tracks in worship, it opened so many doors of creativity and drastically increased the excellence of our worship experience.

But getting started was tough. That’s why I want to share my journey with you and provide you with a clear roadmap of how to begin using a click and backing tracks in worship. This is what I wish I could go back in time a couple of years and teach myself to save hours of research and headache.

Here are the three steps of the process I am going to cover.

1. The gear

2. The software

3. Implementation

I also have a complete online course on how to implement a click and tracks in worship. You’ll find the course inside Worship Leader School. Click here to learn more and join today.

1. The Gear


First, let’s talk about the gear you are going to need. Here is a simple diagram of everything that needs to happen to use a click and backing tracks in worship. First, you are going to need a device that is going to play your click and backing tracks. If you are a beginner, the simplest device to use is an Apple iPhone or iPad. 

Sorry Android users, the app I’m going to show you later does not support Android. If you do not have access to an Apple device, keep watching this video, and I’ll show you towards the end how to get up and running with Ableton Live on a Mac or Windows computer for free.

As you can see in this diagram, we need the signal to get from your smartphone to split into a stereo feed with the click on the left and the backing tracks on the right. This is important because at our front of house mixer we need to have separate control of the click track and backing tracks. 

The click track and backing tracks are both sent to your band’s monitors, but we want only the backing tracks to come through the house speakers. Your congregation does not want to hear a click and cues. 

The cheapest piece of gear to split the signal coming from your phone, tablet, or laptop, is this 3.5mm TRS to dual quarter inch cable you’ll find on Amazon for five bucks. You’ll plug the 3.5mm TRS end into your device and then plug the red and black quarter ends inch into two separate channels on your mixer. The red will be the right channel containing the backing tracks, and the black will be the left channel containing the click. 

If you would rather have your playback device on stage, then you will need a stereo DI box which you can plug the dual quarter inch ends into, and then out of the box you will have two XLR cables that run to your stage snake and then to your soundboard.

Now that the signal from the smartphone is at the soundboard, you need to send the click and backing tracks to your musician's monitors. This is why it is important to have in-ear monitors. You do not want the click track to come through floor wedges. 

If you do not have in-ear monitors, I recommend checking out this article in which I cover what monitors to get based on your budget. At the very least, you need to have your drummer and worship leader using in-ears. That’s what I did when our church could not afford to buy enough in-ear monitors at once. This worked great, and I highly recommend going this route if budget is an issue. 

Now your church audio engineer will be able to mix in the backing tracks to the main mix in the house as well as send your band the levels they need for the click track.

That is all you need for gear to run backing tracks. Assuming you have a smartphone or tablet and a stereo DI Box you can get up and running with a click and backing tracks with a minimal investment.

2. The software - Playback or Prime


I recommend two apps for running a click and tracks in worship: Playback by Multitracks or Prime by Loop Community. Either app is great choice and it’s up to your personal preference for which user interface and ecosystem you prefer. Check out this article where I provide an in-depth analysis of the two.

Here’s a brief walkthrough of getting up and running with the Playback app. Prime works the same way. I’d recommend downloading both to figure out which is best for you.

On your smartphone go to the Apple App Store and download Playback by Multitracks.com. Open up the app and sign up for a Multitracks.com account. This site is a great resource for downloading high quality, and original backing track stems from bands like Hillsong, Elevation, and Chris Tomlin, just to name a few.

I also recommend purchasing tracks from Loop Community. They don’t have all of the original master tracks for popular bands like Hillsong, but they do have budget-friendly options and community sourced tracks from worship leaders who produce custom arrangements.

Once you are in the Playback (or Prime) app, take some time to get acquainted with it and its features. Click the gear icon in the upper right and then select “manual” for a walk through. The capability of this app is astounding, and I cannot think of a better way to start using a click and backing tracks. By default, you will be using the free intro version, but you can upgrade for some more advanced features.

In the Playback app, download the free Play of the Week so you can start using the app and testing it with your sound system. Make sure auto-pan is enabled in settings so your click track and backing tracks are separated into left and right channels.

In the end, I want to emphasize how important it is to try both Playback and Prime for yourself. The biggest benefit to the Playback ecosystem is the track rental program. But Prime has some significant benefits such as the app being free and they even provide free cloud storage for your tracks. Many worship leaders utilize both Playback and Prime in their workflow. You may find yourself doing the same.

3. Implementation

Now that you have the gear and software for running a click and backing tracks, it’s time for implementation with your worship band. Here is what I recommend doing to transition a worship band that has never used a click and backing tracks to using the setup we have just discussed on a weekly basis.

First, communicate your intentions and the reasoning for using a click and backing tracks. You could explain how it’s going to give the band a fuller sound and it is going to increase the quality of their musicianship. 

Next, spend one-on-one time practicing with the click track with your drummer. He or she will be the most important musician to make sure they can play with the click. Once you know your drummer is going to be comfortable, allot extra rehearsal time with the full band to introduce the click and backing tracks. Let the band know ahead of time that you are going to first try using the setup only in rehearsal and then implement it on the weekend once you know everyone is comfortable.

A lot of volunteer musicians get nervous when they hear you want to implement technology like this because they worry they are not skilled enough to keep up. 

Playing with a click and backing tracks is one of the best remedies for poor musicianship and a sloppy band because it provides an extra sense of guidance while at the same time forces everyone to play on tempo, making the band tighter. 

The key for implementation is to take it slow and not overwhelm your band.

4. Advanced Techniques

Once your band is used to playing with the Playback app, here is what you can do to continue to challenge yourself as a worship leader and bring production quality to the next level. 

First, learn how to build your multitrack sessions in Ableton Live. This software is a digital audio workstation that will open up a lot more potential for improving your worship production and creativity. You can still download backing tracks from Multitracks.com or Loop Community, but assembling them in Ableton is a bit advanced. I have a video on how to do this.

Once you are familiar with how to use Ableton Live to run your click and backing tracks, I recommend learning how to control ProPresenter from Ableton. Imagine if you never had late lyric or background cues in worship ever again? Ableton Live automation makes that possible. You can also automate lighting software. Check out my full Ableton Live setup here.

I covered a lot in this article and it may seem a little overwhelming. If you’d like more help and guidance implementing a click and tracks in worship, check out Worship Leader School. It’s a membership site where you’ll find all the essential training, advice, and support you’ll need to plan and lead worship. Inside the school is a complete step-by-step course on how to setup a click and tracks for your worship band. Click here to learn more and to apply to join Worship Leader School.

That's how to get started with a click and backing tracks for your worship band. I hope this saves you some time and gives you the confidence to implement these tools at your church. I believe any worship leader at any skill level and with nearly any budget can accomplish this. If you have not started using a click and backing tracks, what's keeping you from doing so? Share your love and opinions in the comments below and let me know if you have any questions.
 

Tagged: ableton, beginner


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How to Make Lead Vocals Sound Amazing



I have been in many church worship services where you can hardly hear the lead vocal. To tell you the truth, I have even led some of these services.
There are many causes to this problem, so let’s go through the chain of events so you can get your lead vocal sounding amazing.

First, examine the source.
This may seem obvious, but the first thing you need to check is if the lead vocal is actually projecting. In worship, you often come across people who have little to no training or experience, so they have yet to discover how to put the ‘umph’ behind their singing voice.
Generally, they just need to learn proper breathing technique or how to open up their throat and let the sound move freely. I found this great post on breathing: Learning to Breathe.
Next, pick the right microphone.
Not all microphones are created equal. Yet, we often give every vocal the same one. I like to have three different microphones on hand Wharfedale DM series, Sennheiser mics, and the good old. Keep in mind; you should level the EQ on the soundboard before running this test so you have level testing ground.

Don’t forget about the importance of microphone placement.
You should address this before testing microphones as microphones react differently depending on how close or far away it is from the mouth. The microphone should be no more than 1″ from the mouth. Also, never put your hand over the mesh part of the microphone.

Now, you must properly set EQ.
Have you ever moved around EQ knobs without really knowing what you are doing? I know I have. And it usually just makes the situation worse.
Luckily, it is not as hard as you might think. For tips on how to EQ vocals, check out this post:

The Most Important EQ Techniques for Church Sound
Don’t forget the compressor!
Vocals have a wide dynamic range. To get a lead vocal loud enough to cut through the mix, you will often have problems with it getting too loud during bigger parts of the song. So, you can either memorize when these louder parts are coming, or invest in your best friend: the compressor.
Basically, a compressor narrows the dynamic range so you don’t have to worry about it poking out too much, but you can still get those quieter moments to cut through. Here are some baseline settings for vocal compression to get you started:
Set the attack and release to ‘auto’ (or, set attack to 30ms and release to 300ms)
Set the ratio to 3:1
Use the soft-knee setting if available
Set your threshold while the vocal is singing so that so that the gain reduction meter rarely reads more than 6dB
To learn more about compression, see How to Set a Compressor for Vocals.

Get the Vocal EQ Cheat Sheet
You are well on your way to making the lead vocal sound amazing. It will definitely take some work, as every sound system and vocal is different, but don’t give up.

Speaker Impedance Changes Amplifier Power

SOUND FACTORY Speaker tips. 

Speaker impedance changes amplifier power output. In fact, your amplifier power could be nearly half or double its capacity – depending on the impedance of your speakers. But how much should this concern you?
Impedance is measured in ohms. The Omega symbol (Ω) is used for shorthand.

Amplifier Output Power

Let’s say we have an amplifier. The specifications might say the output power is 100 watts RMS at 8 ohms.
Notice the power output (100 watts) is at a specified load (8 ohms). This is telling us that with an 8 ohm speaker, the maximum output power will be 100 watts.
An Ideal Amplifier
If our sample amplifier were an ideal amplifier, then we can also calculate¹ that:
With a 4 ohm speaker, the maximum output power will be 200 watts.
With a 16 ohm speaker, the maximum output power will be 50 watts.
The above shows that for an ideal amp, halving the impedance doubles the power output. Doubling the impedance halves the power output.
Halving speaker impedance doubles amplifier power.
Doubling speaker impedance divides output power by half.
An ideal amplifier is an amplifier which is theoretically perfect. Of course, such an amplifier does not exist, but they are useful when explaining how speaker impedance changes amplifier power.
In summary, in an ideal amplifier, the current from the amplifier will depend on the speaker impedance (ohms). The lower the speaker’s impedance (in ohms) the greater the current that can be drawn from the amp, which means the greater the power.

Real World Amplifiers

The above calculations work well for an ideal amplifier, and help show how speaker impedance changes amplifier power output.
In reality, amplifiers cannot maintain the theoretical output levels as calculated above. This is because the power supply on most amplifiers cannot maintain the maximum power when driving the lower impedance speakers.
In a real amplifier, the above principles still hold but the theoretical values will not be achieved. The power output will be increased with lower impedance speakers, but the maximum power output will not be doubled when the impedance is halved.
As an example of a real world amplifier, let’s look at the specifications of a popular PA Amplifier purchased at Amazon through this site, the Crown XLS1000.
 
This shows that for this amplifier, with both (dual) channels used at the same time, the maximum power output of the amplifier changes as the speaker impedance changes:
With an 8 ohm speaker, the maximum output power will be 215 watts.
With a 4 ohm speaker, the maximum output power will be 350 watts.
With a 2 ohm speaker, the maximum output power will be 550 watts.
This example shows that in a real world amplifier, the principle of “speaker impedance changes amplifier power output” is true, just not as much as in an ideal amplifier.
Please note: this amplifier is designed to work with a speaker’s impedance as low as 2 ohms. Most HiFi amps are only designed to work with a speaker impedance of (or above) 4 ohms.
So What?
So what should you do with this marvelous knowledge? If 4 ohm speaker gives you nearly double the power of 8 ohm speakers, should you only use 4 ohm speakers?
Answer: Yes, and No.
4 ohm speakers are used widely in the car audio industry, as they want to squeeze every bit of power capable from a fixed voltage (~12-14 volts from a car battery). They also mostly design and build their amplifiers to cope with 4 ohms and often 2 ohm loads.
However, it may not be wise to run your Hifi amp flat out at 4 ohms. The reason being, it may mean you are running your amp at or beyond its design limits. The cheaper the amp, the closer you are likely to be at the limits of the power the power supply can cope with. Better to use 6 ohm or 8 ohm speakers, and let your amp comfortably drive them without reaching full capacity. This is similar to a car: better not to constantly drive with the motor at full revs. Interestingly, most Hifi speakers are 6Ω or 8Ω.
A common method of changing speaker impedance is by adding another speaker, either in series or in parallel with the existing speaker. While this will change the output power of the amp, the speakers will share that power. For more details see How Multiple Speakers Share Power.
Most modern amplifiers will, if they are overloaded,  either turn themselves off or reduce the output to protect themselves. However, it is wise not to rely on this self-preservation circuitry, best to design your system conservatively.
Keep in mind that all this is describing the maximum power output of an amplifier. If you don’t run your amp anywhere near full volume, then all this is fairly much irrelevant.
Also keep in mind that doubling the amplifier power only increases the volume by around 23%.  To double the volume you need around ten times the power. For an explanation of this, see the article on Double Amplifier Power does not Double the Volume. 
If you need maximum level from your speakers pay attention to the sensitivity in the specifications Using a speaker with a sensitivity of 90dB (1W/1m) compared to another speaker rated at 87dB (1W/1m) is the same as doubling the amplifier power driving the speaker. For more details on this see the article Understanding Speaker Sensitivity.
While speaker impedance changes amplifier power output, it is not a major consideration for most users. It only becomes relevant when running your amplifier at full power, and then it is best not to run it too close to its design limits.
Never use a speaker (or speakers) below the minimum impedance the amplifier is designed for. If you hear any distortion, it is an indication that major trouble is just around the corner – turn the volume down, eliminate the distortion and consider a redesign of the system.

Top 3 Things All Church Sound Techs Should Know


So many of us got our start in audio by simply being a “warm body” that was willing to help. And what started as “just helping out” turned into a trial-by-fire method of learning how to run live sound and deal with all of the challenges associated with it.
Being thrust into a sometimes-chaotic moment and entrusted with delivering great sound at your church can be equal parts exhilarating and terrifying. I’ve been there! The lessons you learn in those moments can really grow your technical skills and confidence behind the mixing console.
But what are some things we can do to prepare for great sound before we have to endure the fire of a live event or worship service?
A lot of the things we learn through trial and error can be powerful lessons. But we can also form some bad habits if we don’t understand the fundamentals of audio and why something works the way it does.

3 Fundamental Concepts that Lead to Quality Sound

1. Input Signal Quality
Garbage in Garbage out! Make sure your input signal is clean, stable, and high quality. It will make the rest of your job mixing a whole lot easier!  There are two main ways to ensure a good, clean signal:
Use the right cables for your equipment on stage.
Use good mic placement to capture the best sound possible from the source (vocals / drums).

2. Setting Your Gain/Trim
This one is probably the most critical step to ensuring great sound is sent through your console and on through the other sound system components. If you have a bad gain structure, you’ll be battling the quality of your mix all day!
Gain does two things:
1. It allows you to adjust the initial signal level passing through the input preamp of the mixing console, which is then sent to all other functions of that channel and mix.
2. It provides you the opportunity to set the headroom of your audio signal so that it won’t clip, peak, or distort the console’s input electronics.
Be sure to check out this post about how to set your gain properly.

3. Understand EQ & Audio Dynamics
Getting the first two things right will go a long way in helping you achieve a better sounding mix, but you’ll be missing out on the real “sweet spot” for your sound if you don’t also address some simple EQ and signal dynamics control.
This doesn’t have to get super complicated to start with. Think of EQ as a volume control for specific frequency ranges. Knowing what ranges to turn up and down can really help liven up an instrument, give body to a vocal, or allow a mix to sound cohesive and natural.
You need to spend some time practicing with the EQ on your console and train your ears for what to listen for. There is a frequency chart that I use to help me with this.
Bonus: Download Frequency Chart
Here are a few more tips for how to use the EQ on your digital console or analog console.
Controlling some of the audio dynamics in your mix can also help clean up your sound and tighten up your mix. This is done primarily with Compression.
Compression allows you to control the dynamic range of an audio signal. This means that you can limit how loud the instrument or vocal channel can get, and you can set a compression ratio that will determine how much of the signal is compressed or “squashed”.
Here are some great tips for how to get started with compression.
Just remember, practice with these settings BEFORE applying them to your main mix. Use your rehearsal and sound check to experiment with the different features on your console.

Understanding Audio Levels


Geoff the Grey Geek http://geoffthegreygeek.com/audio-levels/
 
A basic understanding of the general audio levels mentioned in this lesson will help you avoid the common mistakes often made when connecting audio devices together. We are going to talk about three different general levels of audio signals. The names of the three general audio levels are speaker level, line level and microphone level. For simplicity, the different audio levels are described in volts. For an understanding of decibel levels used in audio, see Appendix 2 on decibels. . 
Speaker Level 
A speaker needs a few volts of electrical audio signal to make enough movement in the speaker to create a sound wave that we can hear. Small speakers need only a few volts, but large speakers need 50-100 volts to make a loud sound. 

Line Level 
A speaker is connected to an amplifier. Think of your HiFi amplifier at home. What plugs into your amplifier; DVD player, CD player, radio/tuner, video camera?  All these devices plug into the “line in” or “Aux in” of your amplifier. "Line IN", "Aux IN" and "Line OUT" all have an electrical audio signal at line level. 
You are probably aware of the standard red and white leads used in HiFi equipment, these all use line level. Other plugs are also used for line level. Line level is about half a volt to one (½ - 1) volt. It is the job of the amplifier to amplify the half to one volt of line level, up to the 10 volts or more of speaker level. 
Note: A common error is to connect plugs and sockets together just because they fit. Don't assume audio level based just on the type of plug being used. The same type of plug can be used for different purposes (and different audio levels). 

Microphone Level 
Ok, so we have line level (about ½ - 1 volt) which goes into an amplifier to make it up to speaker level (about 10 volts or above). What audio level do you think Mic level is? How much voltage do you think comes out of a microphone, as a result of you speaking into it? Answer: Stuff all! 
The output voltage of a microphone is very low. It is measured in milli-volts, that is 1/1000th of a volt. A mic can give as little as 1 mV, or up to 100mV, depending on how loud you speak into it. That is not very much. So what do you think is going to happen if you plug a mic directly into the line in of an amplifier? Answer: A very low level of muffled sound if anything. 

Mic Pre-amps 
The amplifier wants line level, ½ – 1 volt to produce enough signals to make the speaker work properly. But the mic is only producing milli-volts. So what is needed is a small microphone amplifier that amplifies the audio level from mic level to line level. This should go between the microphone and the amplifier. Because it is for the microphone and it is before the main amp, it is called a mic pre-amp. A mic pre-amp amplifies the milli-volts from a microphone up to line level. 
Mic pre-amps are normally built into devices designed for connecting to a microphone. Equipment like an audio mixer, a digital recorder, a video camera or a computer - all these may have mic level inputs as well as line level input, or just a mic level input. . 
The picture on the right shows for each input on this mixer there is a line level input (labeled Line 3 and Line 4), as well as a microphone pre-amp (labeled MIC PRE). 
Obviously a microphone plugs into the mic input, as the mic inputs are connected to the in-built mic pre-amps. 
A line level device would obviously plug into the line in socket. 
But what if your mixer (or computer/recorder) only has a microphone input, and you need to connect a line level source to it? This would result in the line level (½ - 1 volt) being connected to the input of the mic pre-amp. The trouble is, the mic preamp is expecting only a few milli-volts. The resulting sound will be much distorted as the mic pre-amp is completely overloaded. 

Attenuators 
So how can we do this? How do we connect a line level to a mic level input? We have to reduce the line level down to mic level. The technical word for this is to attenuate the signal. As an amplifier amplifies, or boosts the signal; an attenuator attenuates, or reduces the signal. 
You can buy attenuators at a music shop, they are called DI boxes. DI stands for Direct Input, Direct Injection, meaning you can directly inject a line level into the mic input without any problems. 

Audio Level Summary 
There are three main audio signal levels: mic level (millivolts), line level (around 1 volt) and speaker level (around 10 volts or more). The rule is, only plug speakers into the speaker socket of an amplifier; only line level into the line in of any equipment; and only mic level in the mic input of your mixer, camera or laptop. The most common cause of audio distortion comes from not understanding the different levels, and how to connect them all together. 

Practical Example 1 
Scenario: A keyboard (electric piano) located on the stage needs to connect to a mixer located at the back of the hall, with a microphone multi-core cable connecting between the two. 
Issue: The output of the keyboard is at line level, and the microphone input at the mixer requires mic level. (There is also the issue of different plugs and balanced/unbalanced inputs but these are the topics of other lessons). 
Solution: Use a basic DI box available from most music or electronic stores. A DI box acts as an attenuator which reduces the line level of the keyboard to mic level for direct connection to the mixer (via the multi-core cable). The DI box also overcomes the issues of matching plugs and going from unbalanced to balanced - so this is a perfect solution. This solution also works for connecting electric guitars, electronic drums and DVD players. 

Practical Example 2 
Scenario: The output (line level) of an audio mixer needs to connect to a digital camera or digital recorder which only has a microphone input. 
Issue: The output of the mixer is at line level, and the microphone input of the camera/recorder requires mic level. 
Solution: A basic DI box could be used, but this would require an input lead, and output lead and the DI box - a lot to carry in your camera bag. A neater solution is to have a lead with a 40dB attenuator built into it. This will reduce the line level from the mixer by a factor of 100, which will bring the line level down to a reasonable mic level to connect directly to the microphone socket of the camera/recorder.

HOW TO SET GAIN LEVELS IN LIVE SOUND – THREE METHODS


SoundFactory Gain tips.

Audio mixer channel gain can be set in three ways.  Someone reading that last sentence just went ballistic.  Is there only one right method for setting the channel gain?  Let’s explore.
Follow the signal
Typically, the audio that comes into a channel from the stage is coming in via a mono balanced connection.  The signal comes in at microphone level (a few millivolts), gain is applied, and the result is boosted to line level (a couple of volts) via the channel preamp.  Though we talk of these levels generically, there is some fluctuation of the signal strength such as 5 to 50 mV for a microphone level signal.  For example, if a vocalist sings softly, they will send a weaker signal compared to if they were singing loudly.
The gain (a.k.a. trim) control allows the FOH tech to allow for more or less of the signal to come into the console.  For example, a hot signal from an instrument would need less of that signal to come into the console.  It would be like turning a faucet valve so less water comes out although the water pressure behind the valve stays the same.
Some signals come into the console so strong they can still be heard with the gain at zero.  When this is the case, the Pad option should be used.  It cuts 20 dB from the signal and places it into a manageable range.

Mixer needs
The signal sent to the console needs to be a strong clear signal.  This is why vocalists need to put their mic to their mouth.  It’s also why a guitarist needs to turn up the volume on their guitar.  Otherwise noise can be heard.
Noise can be picked up within the signal path either via interferences on instrument cables or any time the signal goes through a connection.  Noise also comes from microphones as even air particles hitting the diaphragm will produce a sound. Turn on the audio system and open a mic channel.  Hear the small noise hum.
The mixer needs a strong CLEAR signal for optimal mixing.  This happens with a high signal-to-noise (S/N) ratio which means the true source (guitar, vocal, etc.) is so strong in the audio signal that natural noise is overpowered.
One more point on the S/N.  Equipment can be set up so the signal from the stage first goes into a rack component or other off-board processor and that equipment has gain control.  If there’s an initial low S/N, by the time the signal reaches the mixer, that noise can sound substantial in relationship to the desired sound.  This is another reason you should get a high S/N as soon as possible.
Ideally, you want strong signals with a high signal-to-noise ratio where the signals are all in close relativity to each other.  This enables precise volume mixing on the faders.

Gain-Before-Feedback
One aspect of gain setting is gain-before-feedback.  The microphone properties, the speaker properties, and the room properties all contribute to the level in which the audio system can produce sound without producing feedback caused when the reverberation (empty room echo) field crosses over into the microphone field.
This is why floor monitor placement, microphone type, and microphone placement are so critical.  The best sound is the strongest one produced at the source and contains the greatest amount of the source as possible.  A keyboard routed directly into the system is 99.99% pure keyboard.  A vocal microphone, not so much, but the microphone’s proximity to the singer’s mouth along with the strength of the singer’s voice plays into that percentage.

How to Set Gain
There are three primary ways to set the gain and much has to do with the audio environment because of how the faders come into play.
In a studio environment, it’s all about having the clearest cleanest sound possible.  In the live environment, the highest level of fidelity can’t be appreciated because of the nature of the environment itself.
In studio work, gain is set and then hours are spent on EQ and effects manipulation with a little fader manipulation tossed in.  In the live environment, the gain is set and then fader adjustments are frequently made.  This will be an important consideration in two of the methods of setting gain.

Role of the fader
When a channel fader is set at unity (0 dB on the mixer), the console is neither boosting nor cutting the signal.  Using studio gain theory, the fader location (fader is post-gain) isn’t critical as long as it allows for good granular volume control.  That was an overly-simplified statement but it gets the point across.  It’s also where opinions start to fly.
A quick note on clipping, the process of clipping off the signal when it gets too high.  Occasional clipping is ok – that’s why the console has the clipping process.  However, excessive clipping is a sign of an incorrect gain setting.

Method #1
The first method of gain setting follows the studio mentality and says the fader should start at the infinity position (at the bottom) and the gain is increased until the input meter reads almost to the red while allowing for signal increases without distorting, such as when a vocalist sing a lot louder for a passage or a chorus or a single line.  Then, the fader is raised to the point where the volume is right in the room.
Some people use this method but aim for the 0 dB level on the channel’s metering.
Be aware, every two channels at the same output level will create a louder combined sound by around 3 dB.  Therefore, while the main fader would initially be at unity, after setting all gains, it’s possible to have a hotter output than you’d expect.  If I correctly recall, on my typical weekend setup, I see about an extra 6 dB overall.
 
The result of this method is the strongest signal in each channel and with faders all over the place.  They may or may not end up at unity.  This can depend somewhat with what’s happening on stage and how the system is configured regarding the amplifiers.

Method #2
The second method of gain setting, the one I use, follows a live environment mentality which says the volume balance (channel volumes in relation to each other) for one song are often different in the next song and therefore, fader control is very important and it helps to have a BASELINE balance.
To use this method, set the fader to unity and increase the gain until the volume level sounds right for the room and for a general mix.  The result is faders all set at unity with a general volume balance between channels.  For example, lead vocal always on top.  Therefore, when moving from one song mix to the next, subtle volume changes can easily be made with the most granular control (remember fader controls are logarithmic).  Is it the strongest signal?  Maybe, maybe not.  Will it be a noticeable strength difference and produce a sub-par mix?  I’m saying no, with one exception.
 
There are times when we have little-to-no control over the source.  For example, someone hands you a CD they burned from a file they made after they did who-knows-what to it.  In such cases, set the gain for the strongest signal and then set the fader.  You don’t have to ride a fader for a backing track or for the audio from a DVD.  Well, you shouldn’t.
The key to making method #2 work is using proper microphone selection and usage for the cleanest strongest signal from the stage.

Method #3
The third method of gain setting goes a completely different route and uses the location of the faders to indicate where the channel should sit in the mix.  For example, the lead vocal channel fader is higher in relation to all other channels.  A lead guitar fader would be lower than the lead vocal but higher than the keyboard.  By looking at the faders, you are seeing where everything sits in the mix.  You’re seeing the volume balance.  Only then do you increase the gain.
 
Of these three methods, I’ve seen professional live FOH engineers use all three.  They use a method they were taught or what they found works best, such as with the third method.
It comes down to getting the best signal from the source and that’s the area where I see churches doing it wrong because of incorrect microphone selection or usage.

HOW TO EQ A ROOM AND OPTIMIZE A SOUND SYSTEM

Any good technology system with multiple components goes through a basic creation process, whether that is your home stereo system, an IT network or a professional AV system. First, you design the system—you select the proper components, determine how they should be connected and identify their proper setup so they can accomplish the goal the system is intended to solve. Then, you install the system—you physically mount and connect all of the pieces of the system together.
 Finally, you configure the system—this involves adjusting all the settings and performing any tasks that take the system from “physically ready” to “performance-ready.” For an audio system, this involves (among other things) room optimization, which includes adjusting EQs, delays and other settings to account for unique aspects of a room (reflections, materials that absorb different parts of the audio spectrum, etc.) so the system will sound the best it can.
Although there are other aspects that come into play with room optimization (delay, etc.), we’ll focus on the primary one, which is equalization. Now, this is not the equalization of a single instrument or vocal mic that may be adjusted depending on the particular event. Instead, we are talking about more basic equalization settings that are set once when the room is configured and then often don’t need to be adjusted again unless something in the room is changed that affects the sound of the room or sound system itself.
 When we think about EQ, there are three basic types. First, there is the channel EQ on the soundboard, which is the EQ you may think of when I say equalization and the EQ the sound engineer will adjust for an event. The second EQ you will encounter is the speaker EQ. This ensures the transducer sounds its best within the speaker cabinet. These days this is often set within the digital signal processor (DSP) built into the amplifier and the correct setting is typically provided by the speaker manufacturer. Crown amps, for example, have multiple built-in DSPs and contain preset speaker EQ settings for various JBL speakers. Finally, there is the room EQ. This is where you can correct issues caused by the way the sound interacts with the physical room itself. This is the EQ that we are most concerned about in this article.
 There are a number of ways to go about adjusting room EQ, and opinions about which method is the “best” are just as numerous. That said, I’ll list a few of the popular methods that are available and the process. The first method is perhaps the least scientific, but still remains popular and can be effective for engineers with well-trained ears. This is what I’ll call the “mix to your ears” method. In this process, you play a familiar recording through the sound system (clean with no channel EQ adjustments or effects) and then adjust the room EQ until the recording sounds “right” to you. There is, of course, some debate on what constitutes a “good” test recording (I tend to prefer “Donald Lawrence''s Through the fire" for this), but the important thing is that the engineer be extremely familiar with how the recording “should” sound and has a lot of experience. Of course, you can also monitor through a good set of flat-response reference headphones for comparison.
The second process is what is known as “ringing out” a sound system. This method attempts to identify feedback issues and echoes that effect sound balance, and involves connecting a clean microphone and then cancelling out feedback on any infringing frequencies. You start by taking each vocal or instrument microphone and plugging it into a clean channel on the board. Turn the gain to negative infinity (off), bring the master output fader to unity (0dB), and then set the fader for the microphone channel to +5db.  Slowly turn up the sound until there is just a bit of feedback when you say “check.”
For a stereo installation, you will want to use the pan to EQ each side separately. While saying your “mic check” phrases (I like the “Speak the Speech” monologue in Hamlet III.ii), take a channel (or a node with a narrow Q, if parametric) on the EQ and slowly turn up the gain for the channel/node. Slowly test each channel on the graphic EQ (or adjust the node frequency). When the feedback increases and you hear either a “ringing”/feedback or a hollow/booming sound, you have found an offending frequency. Turn down the gain so the “peak” becomes a “scoop” and the sound should not feedback anymore at that frequency. Repeat for other channels/nodes until there is no more feedback.
Of course, many argue that this is still not that scientific (though proponents point to its effectiveness). However, there are other methods as well. For example, some dbx DriveRack signal processors have AutoEQ™, an algorithm that provides a fast, accurate and automatic EQ of a room. The processor plays sine waves in particular frequency sweeps. An attached Real-Time Analyzer (or RTA) reference microphone “listens” to the room as the tones play, and the processor automatically adjusts the EQ in a matter of seconds.
 Finally, you can also adjust the system using room optimization software or a hardware spectrum analyzer. In this approach, the tools use anywhere from two to eight or more microphones to provide detailed spatial analysis of the entire space. You can then adjust the EQ for groups of speakers (a single array, for example). You then copy and adjust across the space until you have as consistent a sound as possible throughout the entire room. This process is obviously the most detailed and can require training to do properly (SynAudCon is a great place to start), but it also provides the sound consistency that top engineers say is vital to a great experience.
The particular approach you choose will greatly depend on your particular situation. Which approach do you prefer (and for what type of application)? Let us know in the comments.

TERMS AND DEFINITIONS

Absorption - The process of changing sound energy into heat energy with special materials. This will then lower the amount of sound reflected back.

Active - A type of PA speaker that is self powered; meaning that the speaker has a built in amplifier. Opposite to passive speakers.

Acoustical Environment - The features of a given room, influenced by the absorption, refraction and it's dimensions.

Acoustical Material- A fabric like material designed to absorb sound in reflective environments. (ex. gymnasium walls or church sanctuaries…etc.)

Acoustics - The science of sound and its properties/effects on a given environment.

Anechoic - Sound without any echo.

Amplifier - Audio amps (amplifiers) are the units that convert the finished, mixed signals (from the soundboard) into electrically, powerful signals for powering the loudspeakers.

Artificial reverberation - Sound passed through an acoustic or electric process in a common effect known as reverb. This method is used to make a sound/signal more realistic. Effect units may have default settings to simulate the sound in desired places or surfaces like; rooms, plates, concert halls, bathrooms, stadiums…etc.

Attack - The start/beginning of a sound. On a compressor unit, the attack refers to the speed in which the compressor begins to reduce the signal as it passes through the threshold. You will generally see an 'Attack' knob on compressor and gate units.

Attenuate - The reduction of an electrical or acoustical level.

Audible frequency range - Perceptible range of frequencies heard by the human ear. (20Hz to 20,000Hz)

Auditory system - The sensory system in the ear that gives the sense of hearing.

Background noise - Unnecessary noise from sources not pertaining to the object of significance. Structureborne, airborne and instrument noise are forms of unwanted noise. Background noise can be a cause of sound bleeding from other microphones. To help resolve this, the use of gates will be very helpful.

Bandpass filter - A filter that reduces signals under and above the desired passband.

Bandwidth - The entire frequency range of a system.

Bass - The lowest range of perceptible frequencies starting around 20 Hz.

Boomy - A listening term, that refers to an overload of lower end frequencies.

Bidirectional - Known as a type of pick-up pattern in microphones that will capture sound from the front and back of the diaphragm.

Bright - A listening term, that refers to an excess of higher end frequencies.

Channel balance - In a stereo system, the channel balance refers to the symmetry of sound levels between the left and right speakers. This action, on a sound board, is usually called 'panning'.

Clipping - A type of distortion that can occur when an amplifier is driven into overload. Clipping can happen throughout all the stages of your church PA system; starting from an input on a soundboard, to the final output of the amplifiers/speakers.

Cloud - An acoustical panel suspended from the ceiling to reduce the amount of reflections.

Compression- The process of reducing the dynamic range of a signal. Generally used for vocals, instruments and drums. How to set up compressors.

Crossover frequency - The dividing signal in a speaker of different frequencies

Decibel (dB) - A number represented for the loudness of sound to the human ear.

Digital Delay - Effect which controls the input of a signal and then repeats after a period of time. Delay units have controls for decaying of the echo and reverb.

DI Box - Direct Input Boxes are used to change different impedance levels from instruments to the soundboard.

Dynamic headroom - The ability of a sound device to reach musical peaks.

Echo - A sound that has returned from its original source and was delayed multiple times.

Equalization - The process of altering frequency responses of a device to achieve a desired response.

Equalizer - In a system, the equalizer is designed to change the frequency response of a signal.

Feedback - Unwanted interaction between the speakers and microphones of a PA system.

Frequency - The calculation of fast variations of a periodic signal, expressed in cycles per second (Hz).

Frequency Response - Changes in the sensitivity of a circuit or room with regards to frequencies.

Front of House - The area that the audience (or congregation) has access to. Mainly excluding the stage and behind stage. Normally where everyone sits or stands at a service or show.

Fundamental - The lowest frequency of a note.

Fusion Zone - When all sounds (natural, eclectically generated or reflected) are fused together and heard by an observer’s ears; this will result in the apparent increase in sound level. Also called, the Hass Effect.

Gain - An increase in level. On most soundboards, this is a function knob that enables the input level into a channel.

Hard room - An environment which the surfaces have a very low value of sound absorption and are reflective. Opposite from a soft room.

Headphones - A tool (device) for the ability to hear any specific instruments/channels.

Headroom - The capability of an amp to go past its rated power for short durations without distortion.

Hertz (Hz) - The unit of frequency that means the same as cycles per second. (Abbreviated as Hz)

Impedance - The resistance to the flow of electric energy measured in ohms.

KHz - Kilohertz - 1,000Hz.

Live end dead end - A treatment plan for acoustics that at one end of the environment it is reflective and the other end is very absorbent.

Loudspeaker - An electroacoustical transducer that alters electrical energy into acoustic (audible) energy.

Masking - Adding a presence of sound to one source by another.

Microphone - A device that converts sound waves (acoustical energy) into electrical signals.

Midrange - In a speaker, the midrange is created with a tweeter for high frequencies and a sub-woofer for the low end frequencies.

Monitor (Wedge) - A speaker or in-ear monitor system, used on a stage to send sound to the musicians.

Muting - A dramatic reduction in the volume level. On a soundboard this function is called the ‘mute’ button.

Noise - Interference of either an electrical or acoustical signal. Gates are ideal for greatly reducing acoustical noise on stage.

Octave - In interval multiples of two between the frequencies bands 20Hz-40Hz. Each octave you add on the lower end requires that your speakers move four times as much air.

Omnidirectional - Referred to as a type of pick-up pattern in microphones. Omni, meaning ‘all around’; captures sound from all directions (360°).

Patch Cable - Typically an unbalanced quarter inch (1/4) phone jack used to connect instruments and other devices. Not suggested to be used for powering monitors or speakers.

PA system - Public address system; its purpose to amplify any given source used for communication in public areas.

Passive - A type of PA speaker that is unpowered, so that the speaker needs an external amplifier source, like a powered mixer or a power amplifier, in order to operate. Opposite to active speakers.

Phantom Power (+48V) - A function on most soundboards used to send 48 votts of electrical currents through audio cables. Mostly all condenser microphones require phantom power to work.

Phon - Unit of the audible loudness level of a tone.

Pick-up Pattern - For every microphone there is a property know as directionality. Directionality is described as the microphone’s sensitivity to sound from numerous directions.

Pitch - The perspective frequency of tones.

Psychoacoustics - The science of sound and its interaction on the auditory system.

PowerCon - Used for connecting the amplifiers and speakers in a PA system. The end of the cable has a twist lock feature, to ensure the connectivity of the cable. Has a very similar design to the speakon cable.

PZM Microphone - A special type of condenser microphone that has a plate (flat surface) that vibrates to all sound locations near it. Also called a Pressure Zone or Boundary microphone.

Refraction - Sound that is redirected by a process of bending sound waves through material with different sound velocities.

Returns - A process used with effect units. A signal must be sent to an effect unit and then 'returned' back to the soundboard.

Reverberation - The persistence of sound when enclosed in a space, which has reflective properties, after the source of the sound, has stopped.

Reverberation time - The falling off of a sound in a closed environment because of reflections.

Slap back - A distinct reflection from a nearby surface.

Snakes - Audio snakes are used for carry many signals in a single cable, great for long distances.

Soft room - An environment with highly absorbent surfaces. Opposite from a hard room.

Sone - The unit of measurement used for the subjective loudness on the auditory system.

Sound insulation - The ability of a given environment’s, physical composition, to stop sound from leaving the wanted origin. Different types of insulation have numerous effects on the sound. This sound energy is not necessarily absorbed; however the sound maybe reflected back or the impedance may become mismatched.

Sound isolation - The degree of acoustical separation between two environments. Like a control room to the live recording room. Some headphones are also sound isolating.

(SPL) Sound pressure level - Expressed in decibels as the loudness or volume of a level.

Sound spectrograph - A device used to measure the level, frequency and time of a signal.

Sound waves - Frequency determines the length of waves and the amplitude (loudness) determines the height of the waves. 

Speakon cable - Used to connect amplifiers to speakers, generally used in pro audio systems. This connector has a twist lock feature, to ensure the connectivity of the cable. (Similar to the PowerCon connector)

Spectral balance - The balance across the whole frequency spectrum of the system.

Splaying - A physical attribute to an environment where the walls are purposefully constructed off square. This is done to imperfect the flow of returning sound waves. Another method used for reducing echoes.

Stereo - Artificial simulation of natural human hearing by creating a 3-D image through a system of supplying two different sources with slightly singular mixes and sounds. In the case pertaining to a PA system, this would be called the Left and Right channels. For true stereo to be achieved; the soundboard, processing amplifiers and two separate speakers must be available.

Subwoofer - A speaker cabinet created for low-frequency reproduction. The best ability that a subwoofer should reach is into the bottom octave (20-40Hz).

Tangential mode - A physical room atmospheric mode produced by reflections from four of the six surfaces.

Threshold of feeling - Sound pressure that can cause discomfort and pain. Situated around 120 dB above the threshold of hearing.

Threshold of hearing - The lowest sound level that can be heard by the auditory system. (around 20uPA)

Timbre - The superiority of a sound that can be notable from other sounds of similar pitch and level.

Tip-Ring-Sleeve (TRS) - Audio cable ends, generally used for connecting instruments through to the soundboard. Called TRS for its physical features. Tip is the left channel signal the Ring is the right channel signal and the Sleeve is normally the ground.

Tone - Is the result of an auditory sensation of the given pitch.

Treble - The highest of all frequencies in the audio spectrum.

Unidirectional - Referred to as the type of pick-up pattern in microphones that will only capture sound from one direction.

Watt - A unit of electricity and acoustical power. The energy is expressed in intervals of seconds.

White noise (ANS) - An audible noise with a constant frequency spectrum. This is used to calculate and equalize the response of a system.

CLICKS AND BACKING TRACKS


How to get started with a click and backing tracks for your worship band

Jake Gosselin 

 April 26, 2017 

 Worship Leading

I led worship for years without using a click track or backing tracks.

I used to think things like:

“It limits musical freedom.”

“Our band members are not skilled enough to play with a click.”

“It sounds too ‘produced’”

“I do not have time in my weekly preparation for a click and tracks.”

Those were some of the thoughts and objections that ran through my head as I considered whether or not it was worth implementing a click and backing tracks with my worship band.

Then I finally arrived at a place where I felt like I was stagnating as a worship leader and I wanted to challenge myself and my band. That’s when I started doing some research to learn what type of equipment and knowledge I would need to begin using a click and backing tracks in worship. 

I knew I was going to need in-ear monitoring, and I heard of software like Ableton Live, but it all seemed intimidating. How was I ever going to convince my band, which consisted mostly of people 20 years older than me, to play with backing tracks, let alone start using in-ear monitors?

Whelp, both of those things happened. It was not easy and did not occur overnight, but in a couple of months I had my whole band on in-ear monitors, playing with a click and backing tracks, and I even began automating ProPresenter and lighting. My only regret is that I did not learn to do these things sooner!

A lot of worship leaders struggle with “staying fresh.” Playing mostly the same songs every week in the same venue for the same people can get creatively stale. I also have a firm conviction that the local church should be a wellspring of excellent and inspiring creativity that makes a powerful impact on the unchurched. 

When I learned how to use a click and backing tracks in worship, it opened so many doors of creativity and drastically increased the excellence of our worship experience.

But getting started was tough. That’s why I want to share my journey with you and provide you with a clear roadmap of how to begin using a click and backing tracks in worship. This is what I wish I could go back in time a couple of years and teach myself to save hours of research and headache.

Here are the three steps of the process I am going to cover.

1. The gear

2. The software

3. Implementation

I also have a complete online course on how to implement a click and tracks in worship. You’ll find the course inside Worship Leader School. Click here to learn more and join today.

1. The Gear


First, let’s talk about the gear you are going to need. Here is a simple diagram of everything that needs to happen to use a click and backing tracks in worship. First, you are going to need a device that is going to play your click and backing tracks. If you are a beginner, the simplest device to use is an Apple iPhone or iPad. 

Sorry Android users, the app I’m going to show you later does not support Android. If you do not have access to an Apple device, keep watching this video, and I’ll show you towards the end how to get up and running with Ableton Live on a Mac or Windows computer for free.

As you can see in this diagram, we need the signal to get from your smartphone to split into a stereo feed with the click on the left and the backing tracks on the right. This is important because at our front of house mixer we need to have separate control of the click track and backing tracks. 

The click track and backing tracks are both sent to your band’s monitors, but we want only the backing tracks to come through the house speakers. Your congregation does not want to hear a click and cues. 

The cheapest piece of gear to split the signal coming from your phone, tablet, or laptop, is this 3.5mm TRS to dual quarter inch cable you’ll find on Amazon for five bucks. You’ll plug the 3.5mm TRS end into your device and then plug the red and black quarter ends inch into two separate channels on your mixer. The red will be the right channel containing the backing tracks, and the black will be the left channel containing the click. 

If you would rather have your playback device on stage, then you will need a stereo DI box which you can plug the dual quarter inch ends into, and then out of the box you will have two XLR cables that run to your stage snake and then to your soundboard.

Now that the signal from the smartphone is at the soundboard, you need to send the click and backing tracks to your musician's monitors. This is why it is important to have in-ear monitors. You do not want the click track to come through floor wedges. 

If you do not have in-ear monitors, I recommend checking out this article in which I cover what monitors to get based on your budget. At the very least, you need to have your drummer and worship leader using in-ears. That’s what I did when our church could not afford to buy enough in-ear monitors at once. This worked great, and I highly recommend going this route if budget is an issue. 

Now your church audio engineer will be able to mix in the backing tracks to the main mix in the house as well as send your band the levels they need for the click track.

That is all you need for gear to run backing tracks. Assuming you have a smartphone or tablet and a stereo DI Box you can get up and running with a click and backing tracks with a minimal investment.

2. The software - Playback or Prime


I recommend two apps for running a click and tracks in worship: Playback by Multitracks or Prime by Loop Community. Either app is great choice and it’s up to your personal preference for which user interface and ecosystem you prefer. Check out this article where I provide an in-depth analysis of the two.

Here’s a brief walkthrough of getting up and running with the Playback app. Prime works the same way. I’d recommend downloading both to figure out which is best for you.

On your smartphone go to the Apple App Store and download Playback by Multitracks.com. Open up the app and sign up for a Multitracks.com account. This site is a great resource for downloading high quality, and original backing track stems from bands like Hillsong, Elevation, and Chris Tomlin, just to name a few.

I also recommend purchasing tracks from Loop Community. They don’t have all of the original master tracks for popular bands like Hillsong, but they do have budget-friendly options and community sourced tracks from worship leaders who produce custom arrangements.

Once you are in the Playback (or Prime) app, take some time to get acquainted with it and its features. Click the gear icon in the upper right and then select “manual” for a walk through. The capability of this app is astounding, and I cannot think of a better way to start using a click and backing tracks. By default, you will be using the free intro version, but you can upgrade for some more advanced features.

In the Playback app, download the free Play of the Week so you can start using the app and testing it with your sound system. Make sure auto-pan is enabled in settings so your click track and backing tracks are separated into left and right channels.

In the end, I want to emphasize how important it is to try both Playback and Prime for yourself. The biggest benefit to the Playback ecosystem is the track rental program. But Prime has some significant benefits such as the app being free and they even provide free cloud storage for your tracks. Many worship leaders utilize both Playback and Prime in their workflow. You may find yourself doing the same.

3. Implementation

Now that you have the gear and software for running a click and backing tracks, it’s time for implementation with your worship band. Here is what I recommend doing to transition a worship band that has never used a click and backing tracks to using the setup we have just discussed on a weekly basis.

First, communicate your intentions and the reasoning for using a click and backing tracks. You could explain how it’s going to give the band a fuller sound and it is going to increase the quality of their musicianship. 

Next, spend one-on-one time practicing with the click track with your drummer. He or she will be the most important musician to make sure they can play with the click. Once you know your drummer is going to be comfortable, allot extra rehearsal time with the full band to introduce the click and backing tracks. Let the band know ahead of time that you are going to first try using the setup only in rehearsal and then implement it on the weekend once you know everyone is comfortable.

A lot of volunteer musicians get nervous when they hear you want to implement technology like this because they worry they are not skilled enough to keep up. 

Playing with a click and backing tracks is one of the best remedies for poor musicianship and a sloppy band because it provides an extra sense of guidance while at the same time forces everyone to play on tempo, making the band tighter. 

The key for implementation is to take it slow and not overwhelm your band.

4. Advanced Techniques

Once your band is used to playing with the Playback app, here is what you can do to continue to challenge yourself as a worship leader and bring production quality to the next level. 

First, learn how to build your multitrack sessions in Ableton Live. This software is a digital audio workstation that will open up a lot more potential for improving your worship production and creativity. You can still download backing tracks from Multitracks.com or Loop Community, but assembling them in Ableton is a bit advanced. I have a video on how to do this.

Once you are familiar with how to use Ableton Live to run your click and backing tracks, I recommend learning how to control ProPresenter from Ableton. Imagine if you never had late lyric or background cues in worship ever again? Ableton Live automation makes that possible. You can also automate lighting software. Check out my full Ableton Live setup here.

I covered a lot in this article and it may seem a little overwhelming. If you’d like more help and guidance implementing a click and tracks in worship, check out Worship Leader School. It’s a membership site where you’ll find all the essential training, advice, and support you’ll need to plan and lead worship. Inside the school is a complete step-by-step course on how to setup a click and tracks for your worship band. Click here to learn more and to apply to join Worship Leader School.

That's how to get started with a click and backing tracks for your worship band. I hope this saves you some time and gives you the confidence to implement these tools at your church. I believe any worship leader at any skill level and with nearly any budget can accomplish this. If you have not started using a click and backing tracks, what's keeping you from doing so? Share your love and opinions in the comments below and let me know if you have any questions.
 

Tagged: ableton, beginner


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HOW AUDIO COMPRESSORS WORK


A compressor is a very important item to have in your sound system arsenal, especially in a church or worship environment. A compressor is a device that controls the dynamic range of your input (guitar, vocals, piano, bass, etc…). When engineering sound for a church, you will note the dynamic differences that each song and song style may be. For this reasons, compressors come in handy big time. They can be used to add some extra punch to the instrument/mix; as well as some back bone to some looser sounding instruments, like the bass drum or piano. Also, a compressor can add sustain to instruments like guitar, or vocals. This can help enhance the realism of the signal. Setting up compressors can be a little confusing, but after you understand all of the required elements a little more you should have no problem.



How many Compressors should you have?

Ideally, you want to have as many compressors as you can get. Having each instrument and each vocalist compressed will even out your sound a lot. Being in control of every instrument is ultimately what every sound engineer is shooting for. However, if you do not have access to that many, you will want to compress your most important instruments first. If you only have two, a great start is by compressing your lead vocalist (worship leader). Regardless what you do after this, make sure he/she is compressed. This way you have control of the most important aspect of worship. Another instrument to start with is either the drums or the guitar. Both of these have a mixture of highs and lows that need to be controlled. If you have only one compressor, you can do one of two things. You can either compress the leader, or you can compress the whole mix.

The Compressor

A compressor has the same controls as gates do; the Threshold, Ratio, Attack, Release, and Output. Each control affects the next. The Threshold controls the point in which the compressor starts to reduce the audio level. You should always start at zero, and adjust from there. For example, if you set this to -10, the compressor activates sooner, which will compress sooner; giving you more control of your audio. Ratio is what adjusts how much compression takes place as the signal passes through the threshold. There would be no compression with a ratio of 1:1, because 1 DB of input would equal 1 BD of sound. So say we change this to 3:1, this would mean it now takes 3 DB of sound to equal 1 DB of output. A good starting point for your vocal input is 4:1. The more you adjust the ratio, the more the compressor becomes a limiter. Eventually you will turn the dial all the way to Infinity, a spot on the dial that limits all input volume to produce 1 DB of output.

The Attack dial on the compressor determines the speed the compressor compresses the signal as it passes through the Threshold. There will be LED lights that display this in an easy to read layout. The Release dial controls the speed the compressor returns the sound back to its original level. This is where you can add sustain to a vocalist or guitarist. Great for special equalizing! The last dial is Output. This measures the amount of output the compressor gives. With this, you can cover for the loss in volume you may have due to the compression.

Conclusion

This may seem like a lot of information to obtain, but it is all rather simple. There is no exact rule for setting up compressors, each church, auditorium, or sanctuary has different acoustics and instruments. Each sound tech must experiment with their own set up and find what works the best. Compressors are excellent for controlling dynamics of vocalists. They are also perfect for controlling the bottom end, (lower frequencies) of the bass drum, bass guitar, etc… If you can afford them, they are a great asset to have in your sound system! Experience and knowledge can be helpful by someone who can show you how its done.

HOW DO AUXILIARIES WORK?

Throughout a soundboard there are many different functions and knobs that all operators must know.  On most of the common types of soundboards; on every channel, there are auxiliaries (or labeled as ‘aux’). Auxiliaries are used much like the master volume on a sound mixer; and that is they send signals out from a specific channel that was selected. Aux’s are located below the gain control and above the level output sliders. They will be rotary knobs with numbers representing the different auxiliary channels.

How Auxiliaries Work For Monitors

Mixing monitors through a main mix soundboard will only be achieved through the use of auxiliaries. Auxiliaries are great for selecting individual channels to be sent to the monitors. Simply assign a monitor to a musician or to be shared with more then one. Make sure you know how to set up monitors. Check out, Setting Up Monitors At Church to learn more. From here, select an auxiliary channel to be the master of the monitor you would like to use. Every channel will have a few auxiliaries to choose from. Make sure you know which auxiliary channels you have designated for each monitor.

How To Setup Audio Sends

Setting up audio sends requires a few steps and the operator must know how these setups work. Check out, How to Setup Audio Effects, to learn exactly how to send and return effected signals by using auxiliaries on your soundboard. There are typically only 4 to 6 auxiliary outputs available on most mixers, so make sure that you use your auxiliaries wisely and that not too many effects are taking up the potential monitor configuration. 

Recording To An External Device

Some churches want to have the capabilities to record their services and music that is played. In some cases, designating an aux to be sent to a recorder can work. This will enable the person who is recording the audio, to choose the amounts and levels of each instruments and/or people talking. This again will take up more auxiliaries and it is best to do all you can for monitors and making the musicians happy.

How auxiliary work for subwoofer

Auxiliaries for subwoofers are mostly used these days by a lot of engineers to have a clear separation of lows from all other frequencies. Instruments like mics, woodwind instruments, and a lot of instruments that lie above 100Hz on the frequency spectrum or need cutting don't need to go through subs, so their auxiliary knobs can be turned down. 


Conclusion

Knowing auxiliaries and the functions of them, on your churches mixer is important. Try to take care of your monitor mixes first, settle the musicians and make it more comfortable for them on stage. Then, after that said, it is safe to play around with effect sends and recording.

Thursday, July 16, 2020

The Importance Of Using Headphones


Whether you’re mixing sound in a studio or live environment, the use of headphones is important at times. Headphones enable the desired selection of individual channels while mixing; helping to ‘zone’ in on a precise instruments and thus improving that channels sound. By using your soundboards solo or PFL button, one can choose any channel(s) wanted to be heard. You can also hear the stereo sounds closer while using headphones. If you are looking for a stage monitor headphone system check out the In Ear Monitor article.



Noise Canceling Headphones

Noise canceling headphones are preferred compared to regular headphones. This is because, when listening to a specific signal, the bleeding of other sounds is distracting and will change the perception of the sound. Ear-bud style earphones are not recommended for audio mixing, because of their inability to produce all the range of frequencies. Also, ear-bud style headphones can be a pain to get in and out of the ears. Having full style head-phones is the only way to go.

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When To Mix With Or Without Headphones

Headphones, regardless of the isolating power, should never be warn when mixing a whole set of instruments for long periods of time. This is because while someone is mixing sound for a church will, as much as possible; want to hear exactly what is coming through the mains and what everyone else is hearing. Headphones are great for EQ-ing channels prior to the worship service, because the sound technician can dial in and hear specific sounds. If there is an issue with a particular instrument on stage or anything else that may arise; headphones are great for checking the status of signals and help to understand any problems that may arise. It is best to only use headphones when needed while mixing live sound applications.

Cost Of Headphones

Having a good, solid set of headphones to go along with a mixing board, in your church’s worship center, is a good resource to the techs. Isolating headphones are recommended in all applications. However, to get professional noise canceling headphones will work its way up to $250-$500 online. These headphones can be very expensive but are defiantly worth the cost. Here are some good choices to make, although getting a decent one can also work wonders. 

Conclusion

The use of headphones for church audio is a really handy tool for sharpening the over all sound of a mix. It isn’t a good idea, however, to be dependant on headphones all the time. This will slow the progression of learning how to hear sounds and to get the ‘ear’ for mixing. Try to only use headphones only when needed and only during microphone tests and sound checks.