ARTA Question - Gate Tapering

[DBMandrake]

Originally Posted by dumptruck

Oh, the marker was not carefully set above - I was just showing the approximate time of the impulse for you. It won't actually let me set it anywhere near the beginning of the impulse (because of the min. sample requirement).

Ok, understood.

Okay, so I guess I'm stuck doing that unless I can manage to get set up for a longer time to first reflection then, huh? Like I said, the greater the time before the impulse I set the marker, the more I get the differences shown in the first post above. I don't understand what it is doing to the math behind the scenes when setting the marker back farther, which makes me nervous...

What might be happening when you use a very short window time, (64 samples) use a heavily tapered window (hanning) and set the start marker a long way in front of the beginning of the impulse, is that either the entire impulse or at least most of it is falling into the tapered part of the window instead of the flat part, thus being attenuated.

Basically your sample period is far too short for usable measurements, and using a hanning window will make it much worse since it has a lot more tapering than uniform.

Doubling your sample rate would help, as would obtaining a greater reflection free time. I know what a PITA it is to put a speaker up on a stand in the middle of the room to increase the reflection free period just to take a quick measurement, but sometimes its the only way.

I find with my speakers in their normal resting position near the corner of the room that there just isn't a long enough reflection free period to get any kind of usable measurement below 3-4Khz, so I can't even measure the crossover region. On the other hand if I put one up on a small stand in the middle of the room I can get a long enough window to measure down to about 500Hz quite well.

I have wondered about that. That was just a quick and dirty measurement with a tiny fft so I could post a picture, but I am getting the little dip every time. This is just a woofer measurement.

That's just the woofer on its own with its low pass filter in place, or direct ?

[DBMandrake]

Originally Posted by dumptruck

Apparently there are 3 pulses above the noise floor in the measurement... see below.

Was that measurement taken with "centre peak of impulse response" ticked in the measurement window ? I'm guessing so because of the fact that it shows nearly 1000ms on the scale at the bottom.

The "centre peak of impulse response" option is a special mode which only works with log swept sine which is designed for doing harmonic distortion sweeps. (After the measurement you choose Analysis->frequency response and distortions)

The way the log swept sine is processed into an impulse response results in harmonic distortion products appearing before the main impulse in the impulse time line.

This is an artefact of the FFT process but it is a clever one which is used to be able to extract harmonic distortion data from a single swept sine wave separately from the fundamental. I think this is what you're seeing in the impulse when you expand the gain, and if you do a normal sweep without "centre peak of impulse response" you won't see that before the main impulse because it is cut off.

Have a look in the ARTA manual in section 6.1.8 covering distortion measurement.

[dumptruck]

Thanks for the response. My experience since the last post agrees with what you're saying there. I will just have to work on getting set up for a better sample period and rate. It also seems that my first reflection may actually be enclosure-related, so there may be some hope for my current setup yet. I haven't been looking into it more yet because....

Originally Posted by DBMandrake

That's just the woofer on its own with its low pass filter in place, or direct ?

Direct. Since you mentioned it, I have become determined to track down the source of that with no luck thus far. See my other thread (link above) for much more details.

[DBMandrake]

Originally Posted by dumptruck

Direct. Since you mentioned it, I have become determined to track down the source of that with no luck thus far. See my other thread (link above) for much more details.

See if that funny dip in the impulse response is still there when you measure the woofer through its normal low pass filter. It may just be due to a high Q resonance at high frequencies in the cone breakup region which will not show up once it is low pass filtered. If that's the case its just a driver characteristic and not a problem with your measurement setup.

[DBMandrake]

Originally Posted by EssB

Hi DBMandrake,
You can permanently fix that Audigy loopthrough - read this :http://www.trueaudio.com/downloads/audigy-setup.pdf

Or 2nd Q down from here:
TrueRTA Real Time Analyzer FAQ Page

Worked fine on my original Audigy.

Thanks for that, unfortunately although it also worked on my original Audigy and Soundblaster Live before that, it does not work with the Audigy 2 ZS.

I'm not sure if its a hardware difference or a driver difference, but the only option with the 2 ZS is to go into the volume control panel, record settings, advanced button under analog mix, and choose "Record without monitoring".

This mutes the pass-through only when an app is recording, so it's ok during the actual measurement but it can cause a brief glitch just before the beginning of an impulse, and potentially cause mic/speaker feedback when not taking a measurement. Annoying...

[dumptruck]

Originally Posted by DBMandrake

See if that funny dip in the impulse response is still there when you measure the woofer through its normal low pass filter. It may just be due to a high Q resonance at high frequencies in the cone breakup region which will not show up once it is low pass filtered. If that's the case its just a driver characteristic and not a problem with your measurement setup.

Well, see the other thread - I tried 3 drivers. They were all unfiltered woofers though, so I will throw an inductor on and see what happens.

[DBMandrake]

Originally Posted by dumptruck

Well, see the other thread - I tried 3 drivers. They were all unfiltered woofers though, so I will throw an inductor on and see what happens.

One other thing I forgot to suggest is have you tried an analogue loopback test ?

Try connecting line in and line out together with a 3.5mm to 3.5mm jack lead and take a measurement. Look at the frequency response and impulse response to see the raw response of the card. It's sometimes normal to see a little bit of high frequency ringing in the impulse during a loopback if your card has a steep low pass anti-aliasing filter, but if the impulse is particularly bad looking or the frequency response has ripple, excessive sag above 10Khz or other oddities during a loopback test you might need to play around with your sound card settings a bit.

[dumptruck]

Please see the other thread for an answer/question about that in response to someone saying the same thing. Thanks!
Mysterious Impulse Measurement Problem