problems with acoustic measurements

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wintermute said:
Thats looking heaps better 🙂... with regards to your 2nd graph there, it looks to me like your tweeter isn't working!!! are you sure it is?? It looks like you are getting a 6db/Octave roll off from around 3.5K.....
I too am worried.
Have you run rmaa on your sound card??
Good idea... I'll do that. I have the software downloaded, but I've never used it. However, if soongsc is correct, I'll face this problem only with the mic input of the soundcard, not with the line-ins. And I don't think one can test a soundcard in loopback mode using a loopback cable for its mic input. Can one?

One thing I am sure of is that the line inputs of my soundcard don't roll off. I've done conversions of albums from analog to digital using this sound card, and I get lovely highs. I might not have noticed a rolloff at 12KHz or 15KHz, but I am certain I'd have noticed a roll-off as early as this one.

If you can get a cheap SPL meter, why not try playing a sine wave at say 2Khz 4Khz 8Khz 10Khz record each reading and then compare to your graph....
Why didn't I think of this before??? I have an RS SPL meter. This is the easiest thing to do.
You probably need to do some debugging of the 48Khz!!! check and see if there are any updated drivers for your card, could make a big difference 🙂
Will try this. Is there any other thing I could do for the debugging, other than new drivers? This is really painful.

But then, as Angshu would probably say, who cares about SPL curves beyond 12KHz? For an xo designer it's academic anyway... 🙂
 
Re: Re: Re: my earlier readings...

soongsc said:
I think if you see around over 16K in the UV meter then it would be good(I recall 32K is full scale).
Actually, I initially started my experiments with any VU levels of 17-25K (because the Unofficial SW Manual says so), then did them at 8-10K, and still continued to receive feedback on this thread that I need to lower the levels even more. So make up your mind. 😀

If you just want to build it from a kit, the MadAboutSound kit is a good implementation of the Wallin II preamp.
Thanks! Will try. I need something very simple, which will probably have to be battery-powered to reduce the messing around with xfmr-diodes-caps-regulators.

Almost all sound cards go at least 48K sample rate, why do you say you only get 24K?
Let me recall what I've said on this topic on my earlier posts:
  • My sound card is the Creative Soundblaster Digital Music USB
  • My sound card operates till 48KHz. I tick-mark a check-box in SW which makes it check for 48KHz, and SW tells me I can go up to 48KHz.
  • When I try to do a pulse response with SW and use a sampling rate above 24kHz, I get no pulse in the graph. At 24kHz, I get clean spikes (which you've seen). This is not affected by sample size: I get clean spikes at sample sizes of 64k to 256k
  • When I try to do gated SPL measurements at 48kHz, it gives me some graph, sort of similar in appearance to what I get at 24kHz. (You can see both graphs in my earlier posts).
So, I don't know what else to say, other than the fact that my setup doesn't seem to work really well at 48KHz.

Almost any design using OPamps you can use NE5532.
I wanted an actual design, something which would also supply phantom power to the mic. I know NE5532 would be fine for most opamp-based designs... the reason I mentioned the chip is because I didn't want to go discrete. I'm looking at the Eric Wallin design.
 
tcpip said:
Why didn't I think of this before??? I have an RS SPL meter. This is the easiest thing to do.
I guess I can use the RS SPL meter as a combined mic + mic-preamp, right? In that case, I can feed the signal into my line-input of my sound card, and I won't suffer the high-end roll-off that my sound card's mic input seems to enforce. Great. I can postpone making my mic preamp, in that case.
 
can't think of anything else for debugging, but it could possibly be that the mic input doesn't support sampling rates above 24Khz???? would make sense since voice over 12Khz is a non-issue.

soongsc did point to vikashes kit and also the url for the original design, it can be made on ic experimenter board if you don't want to outlay for the kit, I've done it on experimenter board and on Vikash's pcb 🙂 it works well, then only thing I would like is some more gain, but that is something I'll have to experiment with 🙂. Vikash I think has the pc boards on special due to a problem with the initial layout (easy to fix), a board will make your life a whole lot easier 🙂

There is also this one --> http://sound.westhost.com/project13.htm but you will need to build a power supply for it so it is not so suitable if you want to run off batteries.

Tony.
 
tcpip said:

I guess I can use the RS SPL meter as a combined mic + mic-preamp, right? In that case, I can feed the signal into my line-input of my sound card, and I won't suffer the high-end roll-off that my sound card's mic input seems to enforce. Great. I can postpone making my mic preamp, in that case.


If it has a line out on it, I don't see why not 🙂

Tony.
 
wintermute said:
If it has a line out on it, I don't see why not 🙂
See here (PDF, pages 3-5 of the document). It does. It also has one more feature which you taught me about: it has a ready-made threaded socket for mounting on a photo tripod.

Now I need to look somewhere for a calibration file for this mic. (I have read that the RS meter's mic SPL response is not very flat, but the piece-to-piece consistency in its SPL curve is very tight, thus allowing a calibration file for one of these meters to be used for correcting the non-linearities of any other sample of the same model.)
 
Re: Re: Re: Re: my earlier readings...

tcpip said:

Actually, I initially started my experiments with any VU levels of 17-25K (because the Unofficial SW Manual says so), then did them at 8-10K, and still continued to receive feedback on this thread that I need to lower the levels even more. So make up your mind. 😀

These values are just starting points. In reality, you want to use the highest signal without staturation and get good results. The larger the signal, the better the signal resolution when digitized.

tcpip said:

Thanks! Will try. I need something very simple, which will probably have to be battery-powered to reduce the messing around with xfmr-diodes-caps-regulators.


Let me recall what I've said on this topic on my earlier posts:
  • My sound card is the Creative Soundblaster Digital Music USB
  • My sound card operates till 48KHz. I tick-mark a check-box in SW which makes it check for 48KHz, and SW tells me I can go up to 48KHz.
  • When I try to do a pulse response with SW and use a sampling rate above 24kHz, I get no pulse in the graph. At 24kHz, I get clean spikes (which you've seen). This is not affected by sample size: I get clean spikes at sample sizes of 64k to 256k
  • When I try to do gated SPL measurements at 48kHz, it gives me some graph, sort of similar in appearance to what I get at 24kHz. (You can see both graphs in my earlier posts).
So, I don't know what else to say, other than the fact that my setup doesn't seem to work really well at 48KHz.


If you use 48K and can't see the impulse, you need to change the scale of the chart to minus time to see if it's there, or increase the plus side time scale to see if its there. USB devices have different latencies for different settings, so just lok forward or aft untill you find it. It's just playing hide and seek with you.😀

tcpip said:


I wanted an actual design, something which would also supply phantom power to the mic. I know NE5532 would be fine for most opamp-based designs... the reason I mentioned the chip is because I didn't want to go discrete. I'm looking at the Eric Wallin design.

Just w word about SPL meters. Most of them are A or B weighed, thus cutting the High and Low frequencies. I beleived the C weighed is full pass, otherwise you will need to midify the SPL meter.
 
Re: Re: Re: Re: Re: my earlier readings...

soongsc said:
These values are just starting points. In reality, you want to use the highest signal without staturation and get good results.
How do I understand at what point the mic is saturating? I presume you've seen the earlier comments made by wintermute and others about "furry" SPL graphs?

If you use 48K and can't see the impulse, you need to change the scale of the chart to minus time to see if it's there, or increase the plus side time scale to see if its there.
Good point. My device is working currently with 20ms of latency setting in the Debug pane of SW. I'll increase and reduce this latency and see if I get anything. I've increased the X-axis of the graph right till the end to see if any peak is there anywhere on the extreme right. It's not there. Some more experimentation will be needed, I can see. Tonight will be so exciting... oooh! 😀

Just w word about SPL meters. Most of them are A or B weighed, thus cutting the High and Low frequencies. I beleived the C weighed is full pass, otherwise you will need to midify the SPL meter.
This one actually defaults to C weighted. So that part should be okay. If you see the manual (I've given a link to it in my earlier post), you'll see all these options and settings.
 
Hi TCP/IP what soongsc was saying was to change the X axis in the pulse response graph make it something like -100ms to 100 ms and look for the peak 🙂 you can then adjust the latency based on information rather than trial and error 🙂

on the saturation, I'd say turn it up until it starts getting fury then take it back a little, this is assuming your ears aren't bleeding 😉 I wear hearing protection even with 8K levels I hate to think how loud (with my preamp) it would be to get 15K levels!!!

Tony.
 
wintermute said:
Hi TCP/IP what soongsc was saying was to change the X axis in the pulse response graph make it something like -100ms to 100 ms and look for the peak 🙂 you can then adjust the latency based on information rather than trial and error 🙂
Will do. In fact, I'm pretty sure something is there somewhere because the VU meter tells me the recorded signal strength is, say, 5k, but on the graph, using auto-minimax on the X1 axis, I see only +/-6. (Yes, not 6K, but 6). So I'm sure the peaks are there... I have to look more aggressively.

this is assuming your ears aren't bleeding 😉 I wear hearing protection even with 8K levels I hate to think how loud (with my preamp) it would be to get 15K levels!!!
All I can think of is that your mic preamp has much lower gain than mine. My mic goes beyond 20K levels long before the sound reaches anything near hard-to-tolerate levels for my ears. And no, I'm not (yet) deaf. 🙂
 
possibly one reason is that I run the line in on 0db gain on the sound card, if I bumped that up I'm sure I'd get higher levels, but I figure I'd rather not do any more amplification than necessary 🙂 this is the setting that gives the best results in RMAA so I stick with it 🙂

Tony.
 
Re: Re: Re: Re: Re: Re: my earlier readings...

tcpip said:
How do I understand at what point the mic is saturating? I presume you've seen the earlier comments made by wintermute and others about "furry" SPL graphs?
Yes, teh furry graphs are a sign.
tcpip said:

This one actually defaults to C weighted. So that part should be okay. If you see the manual (I've given a link to it in my earlier post), you'll see all these options and settings.
Actually I think Eric Wallin did a mod on the Radio Shack SPL meter, you might want to see if it's similar to the one you posted.
 
wintermute said:
Hi TCP/IP what soongsc was saying was to change the X axis in the pulse response graph make it something like -100ms to 100 ms and look for the peak 🙂 you can then adjust the latency based on information rather than trial and error 🙂

on the saturation, I'd say turn it up until it starts getting fury then take it back a little, this is assuming your ears aren't bleeding 😉 I wear hearing protection even with 8K levels I hate to think how loud (with my preamp) it would be to get 15K levels!!!

Tony.

Yep.

Also the volume to turn up would be the mic input volume in the Windows volume control. It is quite unlikely that this will saturate if this control is just the software portion. If it controls the hardware section of the sound card, then there might be a point where saturation occurs.
 
Re: Re: Re: Re: Re: Re: Re: my earlier readings...

I spent one more night trying out experiments, and have been able to conclude the following:
  • There's no rolloff at the upper freq in the mic or the mic preamp. I did a loop record of sine waves at 8KHz, 10KHz, 4KHz and 1KHz. With the same volume setting of my amp, I got the same amplitude in the recorded signal on the computer, and I got clean sine waves, no clipping, no sharp sawtooth edges. So I can't blame the hardware or the audio device drivers any more.
  • I am still failing to get any pulse at 44.1KHz. (I'm not sure I want to try at 48KHz, but I'm sure the card hardware supports 44.1KHz correctly.) I get sine waves cleanly at that frequency, but I don't get square waves, and when I do "Measure->pulse response", I don't see any pulse. The VU meter shows me high amplitude signals are being received and recorded, but I just can't see it in the graph even after I increase the X-axis from -100msec to +200msec.
  • I've learned about the mic input volume control in Windows. I've used sine wave loop recordings to set the gains now, and I am pretty sure there is no "furry" sine wave or clipping of any kind.
At 24KHz, I continue to get that rolled-off gated SPL curve which I'd shown you last night.

Therefore, at least one thing I'm now sure of: I don't need to look for an alternate mic preamp: my mic input is already doing clean 10KHz recordings.

soongsc said:
Actually I think Eric Wallin did a mod on the Radio Shack SPL meter, you might want to see if it's similar to the one you posted.
Right now, I've not yet reached the stage where I'll need to look into the finer points of the RS SPL meter.
 
Started a new thread with this stuff

I've tested the patience of the two of you enough, and I wanted to see if there are others in this forum who may want to chip in. I don't think anyone other than the two of you are reading this thread, and I can't believe my problems are all that unique that I need to struggle so much.

Let's see what other responses we get. 😕
 
is one more response ok?? 😉

I just thought of something. do you have the use pre-emphasis box ticked??? this will cause hf rollof if you do (especially if you haven't done an amp calibration and don't have the ref channel. If it is ticked turn it off!!!

with the pulse, did you try setting the x axis to start at -200ms?? It must be in there somewhere if the vu meters are showing a reading 😉

If no one chimes in, it might be time to start a new thread with a different title 😉

Tony.
 
Re: Started a new thread with this stuff

tcpip said:
I've tested the patience of the two of you enough, and I wanted to see if there are others in this forum who may want to chip in. I don't think anyone other than the two of you are reading this thread, and I can't believe my problems are all that unique that I need to struggle so much.

Let's see what other responses we get. 😕

What kind of mic do you have? How long is the capble from the mic to you sound card mic input? Most mics start decaying around 5K. At one time I did get some that did that. Very frustrating when you think the mic meets spec and it doesn't. The MadAboutSound mic caps are quite up to spec. Also, if the mic cap cables are too long, you will also see some effect due to capacitance.
 
Re: Re: Started a new thread with this stuff

wintermute said:
is one more response ok?? 😉
Hey, any number are welcome. Always. 🙂

I just thought of something. do you have the use pre-emphasis box ticked???
No, no way. I know this setting, and I've never switched it on. Moreover, see what I've posted about 8KHz and 10KHz.... I'm getting absolutely uniform amplitude at all those frequencies. From 1KHz to 10KHz, the same amp volume gets me the same recorded amplitude.

with the pulse, did you try setting the x axis to start at -200ms?? It must be in there somewhere if the vu meters are showing a reading 😉
No, I didn't try 200ms, but I did try -100ms. I am now desperate enough to take an exported dump of the file and look at the actual text values to see where the high readings go. In software development, we say, "When all else fails, look at the source." 🙂

soongsc said:
What kind of mic do you have?
A Panasonic WM61, purchased from Digikey.
How long is the capble from the mic to you sound card mic input?
About two metres, I think.
...if the mic cap cables are too long, you will also see some effect due to capacitance.
Yes, but then how do I get clean sine waves at 10KHz?
 
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