Archive for Theory
Nuts and bolts of PA tuning part 2
Yesterday we covered the thought process in tuning the main speakers in our PA. Today we’re going to discuss delay fills.
After starting with the main speakers, I work outward from there until we get to the speakers in the back of the room. Often the first step after mains is downfills. I like to work my way out from the mains because as I add more speakers to what I’m listening to, they all start to work together and I have to follow that energy rather than fight it. Every change you make to one set effects everything else so you have to think holistically and experiment.
The key to making fill speakers work is two things – delay time and EQ. If the main speakers are our starting point and benchmark that everything else must align to, fill speakers such as downfill or delays will normally arrive at either slightly different or perhaps greatly different times to your ear depending on where you are standing in the room. The trick is to use delay so that speakers that are separated by 5 feet or 75 feet sound like they are all arriving at your ear at the same time and thus working together. I use my software program to help calculate the necessary delay times, often times with a bit of trial and error in real world to nudge things forward or backward just a bit so it feels right.
After the delay time is right, I like to first turn off the speaker I’m getting ready to work with and just measure how everything else we’ve already optimized sounds in the location of the fill speaker. This tells us what is missing sonically in this location from what we’ve already done and that is what we will focus on with EQ’ing the fill speaker. I will high-pass the speaker so the low frequencies that are already hitting the area from the main speakers are not competing with additional stuff trying to come from the fill. Now balance the sonic EQ of the fill so the response curve sounds as close to standing in front of the main speaker as possible. Continue this process with each set of fills and eventually all parts of the system are operating at the same time.
As I said, this is a very high level discussion because the reality is that the process of tuning a PA is as much art as it is science. For me, depending on the complexity of the system and the acoustics of the room it is installed in, there can also be a fair amount of experimentation & trial/error. One of our largest rooms at newlifechurch.tv has a distributed audio system design – 4 main arrays in the front of the room with downfills below them and then a ring of delay speakers 2/3′s of the way back. It took me three attempts at tuning this room before I arrived at a product that I’m reasonably happy with as a long-term starting point. Each time I did it, I learned more about how all of the speakers interacted with each other, the acoustics of the room, and the result was less and less needed EQ each time.
In my experience with trying to get a great board mix for broadcast from the same console mixing FOH, a well-tuned PA is absolutely required to have any chance of success.
Nuts and bolts of PA tuning
When I first visited newlifechurch.tv last winter, I knew immediately that one of my first tasks when we got here would be to start from scratch with PA tunings and system optimization across the board. The tell-tale sign this would be necessary came from looking at channel EQ on the consoles. Channel after channel showed LOTS of EQ and I heard complaints that the board mix that fed the internet campus never sounded as good as it did in the room.
I have a basic process that I follow when tuning a PA. Everyone probably approaches something like this differently so your mileage may vary, but here’s a sampling of my thoughts.
First and foundational for me is to not take anything for granted pre-existing in the system from past engineers, the installation company, or “helpful” volunteers. Crossover points in bi or tri-amped speakers, the manufacturer recommended EQ points on a box, amplifier processing either bypassed or enabled, every speaker functioning, balanced levels between boxes, and on and on. I like to begin by ensuring that every box is actually functioning, the array is balanced so that the level remains consistent as you walk the room, and that all DSP in the system is either flattened, bypassed, or disabled.
There is nothing worse then spending hours working on a set of speakers to then find that a circuit was engaged somewhere, thus coloring what you’re doing, and you didn’t know about it. In one of the newlifechurch.tv rooms, just this step alone brought huge improvements to the system because I found that one of the three boxes in each of 4 arrays was operating at 50% of the volume of the other two boxes due to amplifier trim. The result was unbalanced coverage front to back that had likely existed for a long long time. Don’t take anything for granted!
Next, I’ll always begin a tuning process with the main speakers. In some rigs, like one of our main rooms, this is all I need to deal with because there are no delays or fills to add into the mix. The main speakers will always carry the biggest load of the work in a system and put the most energy into the room. Because of this, I like to start here and then fit the other boxes around the mains. I’ll talk about the software I use in another post. For now we’re keeping it to a 10,000 ft level and just talking process.
The ear doesn’t hear things “flat”, especially as the volume level increases to concert levels. As such, I’m not looking to create a flat PA. Some guys named Fletcher & Munson did lots of research years ago on this hearing phenomenon, resulting in the Fletcher – Munson curves. (Google it if this is completely new to you – fascinating stuff.) In working with the main speakers, I’m looking to smooth them out sonically and ideally emulate a Fletcher-Munson curve (smooth cut in the PA that starts around 1k and has its deepest point at 4 or 5k before returning to normal by 10k).
I go back and forth between measuring a 2 second sine sweep in order to graph frequency response and listening to a playlist of room tuning songs from my iPod that I’ve been using for this purpose for years and KNOW how they should sound. Often times you can over-tune a set of speakers by going crazy with every little dip and peak on a frequency response curve, but the real test is how it actually sounds with music. The magic is in a healthy balance between the two – science and art. In the end, the ears always win.
Tomorrow we’ll cover delay speakers…
System Tuning 101
One of the first tasks that I’ve undertaken as I dive headfirst into audio at New Life Church is the tuning of our PA systems in our main rooms. However, before addressing the journey of each room from where it began to where it has ended up, I thought it would be good to outline as concisely as I can a philosophy for how I think a PA should be configured and what equalizers in the chain should be used for what purpose. Eventually, over the course of multiple posts, I hope to come around full circle to not only show what I’m trying to accomplish in the room, but how that translates to an audio for video capture process as well.
In my mind there are three places in a system where equalizers normally reside. CHANNEL EQ –> SYSTEM EQ –> SYSTEM PROCESSOR. We’re going to talk about them out of this order in order to get my point across so here we go…
First is the channel EQ that exists on the mixing console. I view the responsibility of this part of the chain to provide correction or artistic shaping of the interaction between a microphone and an instrument or vocal. If a mic is too woofy on a vocal or a guitar amp is too harsh (and you’ve already tried to improve the situation at the source by working with the artist), the channel EQ is the place to make the necessary adjustments. Carrying this out, once you’ve made all of the corrections to the various inputs, you should then be able to listen to the results of this mix through headphones or record the left/right mix and play it back on another speaker system and have a good representation of what you intended the mix to sound like. It shouldn’t be too bright or too dull because you’ve fixed all of those individual combinations on the channel strips.
The second equalizer/processing location in a large sound system is the system processor which addresses the PA speakers themselves – the crossover points of a bi or tri-amped system, signal delay necessary to get all of the speakers arriving in a common time domain, and equalization that addresses the natural interactions between all of the speakers in an overall system. Normally this degree of processing occurs inside a system processor – a Dolby Lake, BSS London, proprietary amplifier, or in my case, Shure P4800′s. It is located in the chain directly before the signal coming from the sound mixer hits the amplifiers that drive the various speakers.
The final equalizer/processing location is the overall system equalizer. This point in the chain falls between the output of the mixer and the input to the system processor/amplifier/speakers. Sometimes this processing point might be combined into the system processor, in the old days it used to be simply a trusty 31 band stereo graphic equalizer. The purpose of this equalization step is to optimize and correct the speaker system as a whole to the acoustics of the room where it is located. If the room is boomy or harsh, live or dead, etc. the system EQ allows the engineer to craft the overall sound of the speakers so that what comes out of the mixer sounds transparent (meaning what came out of the mixer is what comes out of the speakers) in the room.
I outlined the three parts in the order I did because I kind of look at them in that order of mental processing as well. The channel EQ corrects for individual inputs and builds the mix itself, the system processor corrects for all of the individual speakers and amplifiers and builds them into a cohesive system, and then the system EQ is the glue that brings those first two pieces together into what is heard in the room. If any of those three places in the chain is not properly optimized, the overall product will suffer.
The primary goal I’m trying to achieve as I think about these three parts of the sound system is transparency. I want what sound came out of the console to come out of the speakers with the same sonic characteristics (transparency). The simplest way to test this is to play a known recorded track (I have a playlist of songs that I use to test how a PA sounds) through the speakers and see if what comes out sounds like how I know that track should sound because I’ve listened to it thousands of times through hundreds of different PA’s – the balance of low end to top end, the harshness or lack thereoff, the warmth of the system, etc. Why does this matter? Because if the PA is not transparent, if what comes in is NOT what goes out on the other end, I’m going to have to make corrections somewhere else in the chain (normally at the same places on every channel’s EQ) to make the mix work and this will likely compromise that board mix. It also makes it far more difficult for a more novice engineer to achieve a great mix unless they are more comfortable with channel EQs or for me to get something great happening quickly because I’m going to have to apply more corrective EQ which will take more time to dial in.
In my experience, the biggest weakness I find in installed PA systems is in the optimization of the system EQ. More often than not (and exactly what I’ve dealt with at NLC.tv), the system EQ does not make the PA system transparent. Playing an iPod track through the console without any channel EQ will sound far different coming out of the speakers than it would listening through good headphones on the console or a different speaker system. The core sonic characteristics of the song do not translate through the room. As a result, the engineer needs to apply channel EQ to correct for these room problems in order to start mixing with a blank canvas. In our iPod example this can be easily done because we’re only talking about a single stereo input. However, explode this out to a 6 or 7 piece band, 5-8 vocals, and 30 or more channels and the task can be far more daunting. The byproduct is that the L/R board mix, which often feeds video recording or internet feeds, suffers greatly. All of the board mix changes that are made to make inputs work in the poorly tuned room are more drastic than what was really needed so the L/R mix isolated by itself is nothing to be excited about.
Next time we’ll start getting specific about a tuning philosophy to correct these problems.
Youtube: Audio Myths Workshop
This is a video version of a workshop from the October 2009 AES show in New York City called Audio Myths workshop by Ethan Winer. In this video you will hear what phase shift sounds like, compare high- and low-end converters, learn about proper test methods, understand why hearing is not as reliable as test gear, and much more. So set aside an hour when you won’t be disturbed, and enjoy.
The original high quality example Wave files mentioned can be downloaded from Ethan’s web site: http://www.ethanwiner.com/aes
Bruce Swedien
I recently ran across the work of another audio engineering legend that is worth checking out. The names of the people he has worked with are too many to list, but when one mentions musicians like Count Basie, Duke Ellington, Oscar Peterson, Sarah Vaughan, Eddie Harris, Quincy Jones, Jennifer Lopez, and even Michael Jackson, a great deal is immediately understood.
Mr Swedien wrote a book in 2004 called Make Mine Music that gives away detailed information from his lifetime in the studio-from a musical, technical, and very personal perspective. This book has something for everyone who is interested in music, especially those curious about the stories behind the scenes of some of the best music to ever come out of the recording studio. I came across the book in 2005 but had forgotten I even owned it. Upon rediscovery, there are too many pearls of audio wisdom in this book to list.
Being such a fan of Bruce, it was with great interest that I found this snippet a few weeks ago on ProSoundWeb from the book that is solid content for anyone who practices this artform of mixing modern music. Here Bruce writes about developing your own “audio personality” for how you evaluate what you hear and translate it into an actual mix “product”.
Do yourself a favor and check this out. If you’re like me and the article resounds with you, pick up the book. I think you’ll be glad you did!