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Tune your system the way live pro sound people do for cheap

barredowl

Before and after Science
I am taking a live sound engineering class now and learning a lot about how PAs used for live reproduction are tuned to sound optimally well in a venue. The basic process is to play pink noise and then listen with a real time analyzer and then cut frequency bands until you have a curve displayed across the real time analyzer as close to an equal loudness curve as possible.

http://en.wikipedia.org/wiki/Equal-loudness_contour

Sounds too expensive for a home audio listener, right? Here is the thing, you can download the free Audacity sound editor that has a pink noise generator, play the pink noise into your amp and speakers and then measure the result with the free RTA (real time analyzer) available for the ipod touch and iphone.

http://itunes.apple.com/us/app/rta-lite/id337536333?mt=8

Hook up a 50 dollar or less equalizer you can find at a pawn shop into your system and you are ready to go to tune your system to an an equal loudness curve across the display of the iphones RTA app.

Judging from what I heard in class the difference between a system dialed into an equal loudness curve and a non tuned system is dramatic. IMO it's worth a try if you have an itouch or iphone already or can borrow one, it's a cheap experiment. Let me know here if anyone tries it.
 
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I want to get into pro sound reinforcement, it looks like a good gig, I have friends who do it professionally and they seem to enjoy their work. So far I think I am doing well in the class, thanks for asking.
 
1/3 octave would be best of course but even an octave equalizer is better than lumpy room response and no equal loudness to compensate for the ears increased sensitivity to mid range IMO.
 
What are you using for a microphone to pick up the pink noise? You DO realize, don't you, that microphones have their own pickup patterns, and many of them don't even come close to matching the pickup pattern of the average human ear?
 
The RTA app is calibrated to the microphone on the iphone. I compared it to the expensive dedicated RTA in class and it does a surprisingly good job, maybe not perfect, but again better than people are going to do by ear unless you've done extensive training of hearing at different frequencies. Sadly room response which is normally not dealt with at all in home audio is perhaps the biggest factor in the sound of any reasonable system that is reaching say 40 to 18,000 +- 5 db which most Akers systems will.
 
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Cool tip - thanks! :thmbsp:

Another possibility is to pick up one of the old Soundcraftsmen EQ/Analyzer units, which have the noise generator and everything else you need built-in, except the microphone, which came with them as an accessory. Get one with the mic, or find a compatible mic, follow the instructions (which you can find online, if needed) and you're set.

Some audiophile purists don't want to add the extra component in the signal path, but in many cases of less-than-perfect real world listening environments, using one of these is probably a better choice than not using one. Pros don't do these things just for the heck of it, nor just for the fun of it.
 
I'd guess the mic in the iphone is cheap carotid dynamic but that won't matter much if the app is properly calibrated. Would it be good enough for a recording studio? No of course not, better than an untuned system, I'd bet money yes.
 
What are you using for a microphone to pick up the pink noise? You DO realize, don't you, that microphones have their own pickup patterns, and many of them don't even come close to matching the pickup pattern of the average human ear?
Yep. With a good amp, good speakers and a poor mic, you're just calibrating the mic.
 
Actually you couldn't be more wrong aabottom since the mic is already pre-calibrated to the app and lets say is accurate +-2db in any particular frequency band you are still going to want to end up rolling off frequencies around 3.15k more than any possible error in the app itself due to the Fletcher Munson derived data that shows the ear is way more sensitive around that frequency. Plus frequency response in any room often has large 5+ db or greater anomalies due to things like bass nodes, reflections, position of ear relative to the tweeter and it's dispersion, etc. I am going to guess that you frankly don't have experience with this equipment, have you ever used an RTA? The difference between a system tuned to a room is dramatic and easy to hear for all you subjectivists out there. The system tuned to an equal loudness curve with rolled off mids has clearer vocals, and more upper bass (say up to 250 hz) and lower mid weight (up to 2000 hz) and more high freuncy energy for cymbals that not only allows you to hear all frequencies equally but that sounds much more satisfying.

To hear all frequencies equally loudly you do *not* want flat frequency response due to the varying frequency response of the ear and to compensate for room anomalies. A glance at the equal loudness curve shows just how large those frequency response anomalies in the ear alone are.
 

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I'm weak and often use a graphic EQ and/or dbx 3BX in my system. Both pairs of speakers are set up in a soundstage type configuration now, so once i find a compatible microphone, i'll give it a go.
 
You can get a calibration mic at a music store for around $100, that and a simple RTA (real time analizer) unit is well worth the bux. DOD and DBX both work well.
 
:scratch2:.... Interesting points. Wouldn't it make sense to start with an objectively "equally loud" curve, and then modify it by ear, to take into account the psycho-acoustic perceptions? The result should be what "sounds best" to one's own ears... right?
 
If the app for iPhone takes into consideration the response of the mike as built into the iPhone then it should be great. The mic in the iPhone should be an electret condenser.
 
I already do this with a Sansui SE-9 Compu-Equalizer. It has a graphic equalizer and spectrum analyzer with pink noise generator and comes with a mic. I got a good Behringer condenser mic. The same one Zilch uses to take his measurements. The cool thing about the SE-9 is that you can program several presets and when selected, all those sliders move up and down mechanically on it's own.

EQ_Deck.jpg
 
:scratch2:.... Interesting points. Wouldn't it make sense to start with an objectively "equally loud" curve, and then modify it by ear, to take into account the psycho-acoustic perceptions? The result should be what "sounds best" to one's own ears... right?

If one is ultimately going to tune it by ear anyway why even bother with the RTA part? :dunno:
 
I am a pro sound engineer, i work large arena and stadium systems and have done for many years.
Love your enthusiasm for the craft, but I feel I have to chime in here.............
You cannot simply fix the sound of a PA system in a room with an RTA and a 1/3 octave iso frequency equalizer, there are more sophisticated ways of doing this now, what you are describing belongs back in the 80's/90's, it would be over 5 or 6 years since Ive used a graphic for that matter................probably 20 years since a 10 band graphic..........

1. We dont use RTA's exclusively to tune systems. Most of us have been using a program called Smaart Live for about 8 years now, it has three functions. One is the RTA, which I only use the spectrograph mode, during live performances, NOT for tuning. The second is to take time snapshots, for time aligning. The third is what we call the transfer function where we look at the actual response of the speaker system by eliminating the room, it also looks at the phase of the system over the bandwidth of the system.

2. The mics I use are made by B&K and cost nearly $2000US per mic. I have two.
You are simply wasting your time with anything less than this.....
My rack of digital processors made by Dolby Laboratories cost in excess of $60,000US, complete with encrypted wireless network to a Motion Labs PC tablet which runs the software for the processors.
So I can wander the venue tweaking the PA system as I go.
In the top of my rack there is a 31 band BSS960 stereo graphic for those old school engineers we get occasionally that like a "grab eq".

3. We do not make the system flat, after many years of tuning, time aligning, phase aligning live audio systems, I have a feel for what flat is, and it certainly is not a straight line across a response trace in an RTA.
Our hearing is not flat, our ear canals resonate at around 3khz, and we also take into account the Fletcher Munson Curve.

4. Depending on the size of the array, the system will behave differently because of the coupling effect of the cabinets, what we do is get the system as neutral as possible, neglecting the room.
This means balancing the 4 bands SUB/LOW/MID/HI, to obtain a nominal starting point.
All the measurements are done using pink noise........
Because the subs are mostly on the floor and the Mid/high cabinets are in the air, we need to make sure the top end of the subs are in phase with the bottom end of the mid/high cabinets. I use the transfer function to do this.
We will have four hangs on PA, LEFT SIDE/ LEFT MAIN/ RIGHT MAIN/ RIGHT SIDE.
The main hangs have to be in time with the side hangs, so I position my B&K at the midway point between the side coverage zone and the main coverage zone.
I take a time shot of the main hang, then a shot of the side hang, I then apply delay to whichever hang is closer so it matches the other hang.
Bingo, seamless transition between zones.
I also do this for the front fill cabinets along the front of the stage.
Sub phase alignment, midway down the arena I place the B&K mic directly in front of the PA system. I always stack the subs directly under the main hangs so that there is no horizontal discrepancy of distances between the subs on the ground and the flown mid highs.
In the transfer function I take a shot of the mid high system, and apply the delay to the reference signal so we get the true phase representation of the system, I am interested in the low end of the flown PA.
Then i mute the flown system, and turn the subs on (previously the subs are muted so as to not screw up the readings), I overlay the phase trace of the mid high (concentrating on the very bottom end) over my sub phase trace, I can see if there is a discrepancy with the phase at this point, normally i need to apply time to the subs to bring them in line with the flown hang.
Im looking to line it up at the crossover point, which in our case is 60hz, there is always phase shift at the crossover points because of the nature of the active filtering curve.
It makes sense if you think about the distance to the flown hang the bottom of which is 27feet in the air, and then the distance to the subs.
The subs are always closer.
In the window below the phase trace we see a response trace, to check I have the phase right, I will run the SUB and the LOW together and then invert the phase of the SUBS, if I have it right I should see a large "hole" appear at the crossover point (which is what I aligned the subs to).
When I put the phase back, the hole will disappear, when I mute the subs the response at the crossover point will decrease, bring the subs back in and we will see the coupling of the two bands at the crossover point.
One thing to be careful of is to not apply too much time, because you can end up being one whole wavelength out, if you are having to apply that much time, generally you will need to use less time and invert the phase of the subs............

Once I have done this, yep with even more pink noise I will tune the system.
This is done using the parametric equalizer in the Dolby Lake processors.
There are standard filters and "mesa" filters (dolby's own linear phase filters) which can have the shape of the filter manipulated at each end of the centre frequency. These filters have no phase shift. The standard filters do.
So I will equalize the speaker system (neglecting the room). The transfer function can distinguish what is coming from the speakers and what is coming from the room because, it receives the pink noise before it goes into the system, and with the time delay applied to the reference noise to appropriate the distance the mic is and the latency from the processors, the program can eliminate what is a reflection from the room or room resonances........or the stage hands driving forklifts around!! Or guitar techs making noise with guitars!!
Once the system is tuned, then I will look at how the PA performs in the room environment, I sometimes use the spectrograph to do this but not all the time.
I have the information on what the system is actually doing so this part is very easy, once you are experienced
But most of all I USE MY EARS for the final appraisal of the sound......yes MY EARS......even on the biggest of systems, which in our case is upwards of 300,000watts..............60,000 stadium sized systems.

With a perfectly timed aligned PA in an arena, the final touch of room compensation can be very minimal, because the system is starting from a normalized state, yes there are room frequencies which need to be attended to, but all too often PA systems are over EQ'd because they are often setup incorrectly, and far to often system techs are not qualified enough to set the system up properly with a good understanding of their craft...........

A graphic and an rta with an average mic is not going to be as anywhere near useful as a highly trained ear.
If Im doing a small gig without my high tech gear, I use my ears!!
A mic picking up a curve in a room tells you nothing about what the system is actually doing, a resonance from the speaker system or a room resonance, all it tells you is "this frequency is louder than this one".
With the sophistication of the Live Sound systems we use these days, it is just simply not enough.....................

Theres a brief overview of system tuning which I do all the time, there is more, but the post I think is long enough.

It takes me about 25 minutes with an assistant running around the various zones with the analyzer mic......

Man thats a lot of typing, I hope somebody finds it interesting :D
 
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