Editor’s Note: Here’s an interesting thread from the PSW Live Audio Board (LAB) forums. It’s lightly edited for grammar and formatting. Enjoy.
Posted by Ivan
What is it measuring? For the most part it is measuring what it is “hearing” during the spaces in which there is no signal.
So the sound that is present after it stops (reflections, reverberation etc) and how those interfere with the next signal give the quality of the signal.
In order to get a high STIpa number, you need a dead space (one that does not add to the original signal).
So what about the loudspeaker system itself. If you have more than one source that you are hearing (as in a loudspeaker array)
The signal should “stop” after the closest arrival, but if there is another arrival (because of another loudspeaker a little distance away), the sound quality is degraded, or not as pure as if you had heard just the single signal arrival.
This will happen with any array in which you hear more than a single speaker. The more arrivals you hear-the worse the “silent gap” is filled in.
And if you “think” you are not hearing the extra speakers in an array, simply turn off the one facing you, and see if you hear sound from the others. I assure you, you will, and a lot more sound than you would think.
The same thing happens DURING the signals. Many people talk about harmonic distortion yet hey have NO IDEA what it actually is, how to look at it etc.
Basically it is additional “free sound freq.” The distortion is the upper harmonics that are added to the sound. The less of these freq that are added, the cleaner the sound.
For those who have never done this, here is a little experiment that you can do with a simple cheap phone RTA app.
Grab a speaker and put a sine wave in (Let’s say 500Hz) at a low to modest level.
Look at the RTA. You should see the single peak.
Now turn up the level. and look at the higher freq. You will see more more higher freq start to appear. These are the “harmonic distortions” that are being added to the pure sine wave.
Notice also that as you turn up the level, the original 500Hz does not go up as quickly as the harmonics do.
So at higher and higher level, the distortion gets higher and higher. Since the original signal is just a sine wave-all you should see is the single peak, but you don’t, you get the “extra free freq.”
So what your ear is actually hearing is NOT the original signal, but free stuff the loudspeaker is adding TO the original signal. I say “down with the free sound” in order to get a better, clearer, more true sound.
Sorry for the ramble, but it just “hit me” yesterday and started me to think—–sometimes that is not a good thing.
I have another theory on why some systems tend to make your ears ring more than others-while at the same SPL (it has to do with the first part of this post), but that is for another time.
Reply by Scott
I feel like I walked into a half-finished conversation…
One thing to watch out for when measuring with a phone-based RTA is that you don’t know when the microphone/input starts clipping. That’ll add a lot of harmonics for sure.
Reply by Ivan
One way to reduce this possibility is to move further away from the source to reduce the level, but then the direct to ambient level will increase.
The reason I mentioned a simple phone app was because of the price. And hopefully some people would start looking at some measurements, when they would not if they had to spend good money on a real measurement system that could take a higher input level.
When you start to measure (getting started is the hardest part), you start to question all sorts of things. And hopefully with those questions, you start to seek answers. And then you have increased your knowledge and understanding-causing you to ask MORE questions-and continue to learn.
I figured this would be a easy quick cheap introduction.
Reply by Stephen
Ummm, what would happen if he hooked the analog scope up to the digitized signal? e.g. what is going into the DAC.
While theoretically samples are infinitely small points, the reality of electrical signals is that they aren’t. As his explanation of band limiting shows. All practical electrical signals are band limited to some degree. So in practical terms you have more of a stair step than he admits. The aliasing filters do the band limiting to average the samples into his output waveform.
The issue with time is not that something has to line up with the start of a sample, it’s that the samples need to be in sync. Any variation in the timing of the samples means that the wrong data could be sent at the right place in time or the right data is played back at the wrong place in time.
Reply by David
So how do you start? I can’t afford a Smaart rig and class but I’d like to begin learning about measurement. Will a cheap measurement mic, USB interface and REW get me into something I can use?
Reply by Keith
Lay it on us Ivan. I’m all (ringing ) ears.
Reply by Ivan
It depends on what you want to do/measure and to what accuracy/quality.
There are several measurement mics that are under $100 (some under $50) that will be “good enough” for most applications. Generally where they “start to be off” is above 10Khz and below 20Hz or so.
I have not used REW, but it is mentioned a lot, so I am not aware of its capabilities. Most of the USB interfaces should be “good enough” for most general applications.
The HARDEST thing is NOT learning how to use the particular program or hardware, but rather learning how to measure, how to determine what the measurements mean, what are indications of a bad measurement etc.
The big thing is don’t simply accept that just because it shows up on the screen that it is accurate/correct. Learning to look at the phase trace and learning what it is telling you is important.
Reply by Keith
Amen brother! The trick is to learn what to ignore.