Help me WmAx; you're my only hope!

[Buckle-meister]

Hello WmAx,

I need to pester you about a few things if you don't mind?

First of all, this post is with regard to our last discussion on Behringers and the PEQ/GEQ of my Z9. I have run the Rives test disc 2, which is calibrated to my Radio Shack analogue SPL meter, to obtain my rooms response to the CD's tones from 20Hz to 20kHz in 1/3 octave increments:

1. With all bass management cancelled and
2. With YPAO's PEQ engaged

Note that I only used 63Hz to 20kHz, i.e. ignored the tones below 63Hz on the Rives CD because my sub is crossed over at 80Hz. I also calibrated the response to the 1kHz tone set to 80dB.

The attached Word document shows what I believe to be a dreadfull response for the Front Left speaker. I suspect that the results are simply as a result of my having a square room, a fair proportion of glass on one face, and a double width doorway opening on the opposite face. My couch is also an L-shape, with the return of the L obliterating approximately half of the sound coming from the speaker in question.

Is this the worst response you have ever seen?!

The three plots corresspond to:

a. No PEQ or GEQ in blue
b. YPAO's PEQ in pink
c. The best correction I could make by overriding the PEQ and adjusting the GEQ in yellow

Why do you think that the response trails off by so much SPL from 8kHz plus?, and is this where you recommend the Behringer?

Regards

p.s. To anybody else reading this, please feel free to chip in with your comments; they're all very welcome.

 

[WmAx]

The attached Word document shows what I believe to be a dreadfull response for the Front Left speaker.

So is the response farily similar for the right speaker?

Is this the worst response you have ever seen?!

For one thing, it is hard to know how close this is the actual response, since some of the 1/3 octave sampling may end up coinciding with comb filtered sections of the response, thus giving an inaccurate representation of what is actually occuring. For example, notice the huge discrepancy[10dB] on average, between the 100-800Hz band compared to the 800-7000Hz band. If this was accurate measurement, then your upper midrange and lower treble would be extremely high in amplitude/volume, compared to the rest of the bands. The large and smooth swings[for example 300Hz, 400Hz, 500Hz 625Hz] lead me to believe that you are coninciding with very narrow comb filtered frequency dips at the low amplitude measurement points. You need higher resolution measurements. Besides, no way you can know what is happening in the bass range with 1/3 octave measurements.

Why do you think that the response trails off by so much SPL from 8kHz plus?, and is this where you recommend the Behringer?

Perhaps measurement error[as mentioned earlier in this reply]. Or is that couch that you referred to earlier within the path of the tweeter? As for correcting the tailing off treble[if it's an accurate measurement], you may try to correct this with equalization, but you might even find it to sound worse when corrected to be flat in response. You'll only know by trying. The internal equalizer of the reciever should be able to at least be able to adjust this in manual mode, right?

-Chris

 

[Buckle-meister]

So is the response farily similar for the right speaker?

I knew you'd say that! I'll run the tones for the Front Right and post the results tonight.

...it is hard to know how close this is the actual response, since some of the 1/3 octave sampling may end up coinciding with comb filtered sections of the response...

The large and smooth swings [for example 300Hz, 400Hz, 500Hz 625Hz] lead me to believe that you are coinciding with very narrow comb filtered frequency dips at the low amplitude measurement points.

What is comb filtering?

You need higher resolution measurements. Besides, no way you can know what is happening in the bass range with 1/3 octave measurements.

What can I do?, I don't have a set of tones at closer intervals. Where can I get them from? Also, whilst it is true that I do not know for definite the response between my 1/3 octave sampling points, would I be reasonably correct to assume that the response will follow a smooth elastic curve through the points that I do have? It is for this reason that I plotted with a curved line rather than a series of straights.

...is that couch that you referred to earlier within the path of the tweeter?

Imagine a vertical plane centred about the centre of the speakers drivers (i.e. 'chopping' the speakers into two vertical parts) projected from the speaker to my listening position. The couch more or less blocks the sound to the left of this plane as I look at the Left Front. The tweeter is probably above the top of the couch's arm rest, but not by much.

As for correcting the tailing off treble[if it's an accurate measurement], you may try to correct this with equalization, but you might even find it to sound worse when corrected to be flat in response.

It would certainly be more accurate if corrected, but why would it sound worse?

Assume for the moment that I had the Behringer. I believe that it has been mentioned that incorporating a single Behringer into a multichannel setup is a bit of a problem. However, it is quite likely that (for the moment at least) I would be content to have it act only on two-channel material, i.e. my CD's. I am presuming that the Behringer is not any kind of amplifier, and that it takes a Line Level input, plays about with the signal, then outputs a modified Line Level signal? If this is true, then would the following work:

I could use either the i-link, coaxial or optical digital output from my universal player to my Z9 for DVD's, but route two (Left/Right) RCA interconnects (or the coaxial/optical digital output if unused for DVD's) from my player through the Behringer and then into my Z9's analogue/coaxial/optical CD input jacks as appropriate. The Z9 would either be in 'Pure Direct' (for analogue) or 'Direct PCM' (for coaxial/optical) mode, so the Z9 would not be using any kind of bass management, i.e. the Behringer is doing it?

Regards

 

[malvado78]

Bucklemeister,

Below is a link to a freeware tone generator. You can out it on to your computer then save any freq. wave form to a .wav file and create an audio cd from those files.

I believe this will work.

If there is anyone out there that has any reason why this would not le tme know I have been curious as to why this type of thing is not mentioned measuring room response at all. Is there any issues with calibrating a system to a compsite source so the respose will be different for the 5.1 response?

NCH Tone Generator

 

[Buckle-meister]

Below is a link to a freeware tone generator. You can o
put it on to your computer then save any freq. wave form to a .wav file and create an audio cd from those files.

I believe this will work.

Thanks for this Malvado78. On the assumption that it will be ok to use, I will still have to calibrate the tones to the Radio Shack SPL meter's c-weighted curve, but this shouldn't be too difficult (I think). First though, I need to find out how to correct the tones.

Anyone?

If there is anyone out there that has any reason why this would not [work]...

As for the above, I believe WmAx will know the answer (don't tell him, but he's really quite intelligent!)

Oh, hello WmAx!

Please find attached the room's response to my Front Right speaker (the final two plots). The colours match those of before, as does the 80dB reference level at 1000Hz.

It is certainly better than that of the Front Left, especially at the lower frequencies, although it is by no means perfect. Strangely though, in the upper frequencies especially, the response seems to be better than YPAO's adjustments. Any ideas as to why this might be?

I also think that the Behringer is becoming more interesting to me. If I plot only the response of the Front Right at the 1/3 octave sampling points, I can get an very good flat response, but in practice the 'in-between' results are making a mess of things, and of course I presently have no way of adjusting those.

Step in the Behringer think you?

Regards

p.s. I have looked at the manual for the Behringer, and cannot see any mention of the number of bands that can be adjusted, nor the corressponding levels. Do you know what they are?

 

[WmAx]

What is comb filtering?

Comb filtering is an effect where sharp comb-like nulls occur in the frequency response. In this case, it is a result of the direct vs. reflected/delayed signals combining. When the phase relation is significantly different between multiple sources of sound at the same frequency, nulling/cancellation will occur. If you happen to measure at a frequency where such a null/cancellation occur, it can really throw off the accuracy of such a coarse resolution measurement as you are doing.

What can I do?, I don't have a set of tones at closer intervals. Where can I get them from? Also, whilst it is true that I do not know for definite the response between my 1/3 octave sampling points, would I be reasonably correct to assume that the response will follow a smooth elastic curve through the points that I do have? It is for this reason that I plotted with a curved line rather than a series of straights.

It's not safe to assume anything with the coarse 1/3 octave measurements. If you rip the RS corrected test tones[just the bass to lower midrange ones] and compress them to mp3 for me, then send me these, I'll check the relative amplitude levels for reference, in order to confirm it's a standard C weighting compensation. Using the known C-weighted compensation curve, you can use a sine wave generation program such as the previous person recommended.

Imagine a vertical plane centred about the centre of the speakers drivers (i.e. 'chopping' the speakers into two vertical parts) projected from the speaker to my listening position. The couch more or less blocks the sound to the left of this plane as I look at the Left Front. The tweeter is probably above the top of the couch's arm rest, but not by much.

Well, then the answer to my previous question is 'yes'. You should try to make some position adjustments to prevent the couch from being in the line of site of the drivers.

It would certainly be more accurate if corrected, but why would it sound worse?

Why do some people like the color blue? That's why I recommend that you try to correct it with the reciever equalizer and see if it's an improvement. I, for example, despise a flat response into the treble, when measured at the listening position, for most music, but that's just me, although this view does see to be supported by at least some prior perceptual research [1].

Assume for the moment that I had the Behringer. I believe that it has been mentioned that incorporating a single Behringer into a multichannel setup is a bit of a problem. However, it is quite likely that (for the moment at least) I would be content to have it act only on two-channel material, i.e. my CD's. I am presuming that the Behringer is not any kind of amplifier, and that it takes a Line Level input, plays about with the signal, then outputs a modified Line Level signal? If this is true, then would the following work:

Yes, the Behringer is a line level device. You can feed it unbalanced inputs[using a simple RCA->XLR adapter] or digital coaxial input[with an adapter]. The output lines, however, may need to have some potentiometers attached on the output[a couple of resistors used in an L-Pad configuration will also work] cables in order to adjust the output drive voltage, which is at higher voltage than typical consumer equipment is designed to use. All you need be able to do is use a soldering iron. I can instruct you step by step, what to do, via PM or e-mail if you so require.The Behringer would handle bass management. But can you live with only having one source being affected?

-Chris

[1]

5 study conclusions, of the average in-room response of speakers considered to be 'flat' in response, and that measure a flat power response in close proximity.

http://www.etymotic.com/ephp/er4-ts.aspx

 

[WmAx]

It is certainly better than that of the Front Left, especially at the lower frequencies, although it is by no means perfect. Strangely though, in the upper frequencies especially, the response seems to be better than YPAO's adjustments. Any ideas as to why this might be?

I can not know what/how the YPAO bases it's compensation(s) upon.

p.s. I have looked at the manual for the Behringer, and cannot see any mention of the number of bands that can be adjusted, nor the corressponding levels. Do you know what they are?

You can achieve about 18 bands of parameteric equalization per channel in the way[a stereo equalizer] you want to use it.

-Chris

 

[Buckle-meister]

If you happen to measure at a frequency where such a null/cancellation occur, [comb filtering] can really throw off the accuracy of such a coarse resolution measurement.

Why is it throwing off the accuracy? Even if the direct sound is being altered by reflected sound, the SPL meter is only 'hearing' what I would myself. No?

Ah. Are you meaning the accuracy of the results between my 1/3 sampling points? i.e. the extrapolated results?

If you rip the RS corrected test tones [just the bass to lower midrange ones] and compress them to mp3 for me, then send me these, I'll check the relative amplitude levels for reference, in order to confirm it's a standard C weighting compensation. Using the known C-weighted compensation curve, you can use a sine wave generation program such as the previous person recommended.

1. What frequency would be regarded as lower midrange?

2. I will check what format they are in on the CD, but I would imagine that they are .wav files. If so, I can send you those, but I don't know how to compress to mp3 format I'm afraid. I would also need you to PM me an email address to send them to.

I, for example, despise a flat response into the treble, when measured at the listening position, for most music, but that's just me, although this view does see to be supported by at least some prior perceptual research.

1. I know you dislike subjective terms, but if I understand things correctly, going by your plots, you dislike a flat response because you would have to raise the amplitude of the upper frequencies, thereby giving rise to a 'bright' sound?

2. I am relieved to see that your plot's average in-room response also reduces the upper frequencies; I had wondered if there was something up with my tweeters! But why would the 'average' room do this?

The [Behringer] output lines, however, may need to have some potentiometers attached on the output[a couple of resistors used in an L-Pad configuration will also work] cables in order to adjust the output drive voltage, which is at higher voltage than typical consumer equipment is designed to use. All you need be able to do is use a soldering iron. I can instruct you step by step, what to do, via PM or e-mail if you so require.

It has been a long long time since I used a soldering iron, but yes, I could do it if required. I do own one.

...can you live with only having one source being affected?

I think so. I love watching DVD's, but I don't want to kill those I own by watching them too often, and I don't buy so many new DVD's that I constantly watch new films. Besides, by both watching the picture and listening to a compressed soundtrack (DD/dts), it is easy to be distracted by the visuals over the sound or vice versa.

No, my system is used by far for music, and whilst I don't yet own any DVD-A's or SACD's, I would only get them for 'better' two-channel recordings of my existing CD's (covered in one of our old discussions), so given the choice, I would always maximise my two-channel experiance first.

You can achieve about 18 bands of parameteric equalization per channel in the way [a stereo equalizer] you want to use it.

Are you saying that you can select which frequencies to assign as the 18 (or so) bands, or do you mean that they are fixed, i.e. in a similar manner as I have at present, but at approximately half the intervals?

Regards

 

[sploo]

Hi Buckle-meister,

I'd strongly recommend taking a more accurate measurement than using 1/3 octave readings. As WmAx points out, you'll be averaging out many room effects, so what you're seeing isn't really what you're hearing.

If you have the RadioShack SPL meter, you can use its RCA output with a PC and Doug Plumb's ETF software (http://www.acoustisoft.com/).

This will take a complete 20Hz-20kHz room measurement in 5 seconds! It will also allow you to see the effect of your room's modes, and even decay times for each frequency.

Have a read through my bass traps thread (http://forums.audioholics.com/forums/showthread.php?t=12624) and you'll see numerous before/after graphs, all taken with ETF.

Comb filtering can have a major effect on what you're hearing. I'm my room (relatively thin, with hard walls and a wooden floor) the result was truly foul (both in terms of the graphed output, and the resulting sound).

Some fiberglass/rockwool panels have made a huge difference.

Sploo.

PS I'd strongly recommend reading this too (it mentions the use of ETF, and gives some other real-world results): http://www.ethanwiner.com/density/density.html

 

[Ethan Winer]

Sploo,

> I'd strongly recommend reading this too <

Thanks, and there's also a video on my company's site that explains comb filtering in detail with audio examples. The presentation may be a bit dry, but the information is solid and useful.

--Ethan

 

[WmAx]

Why is it throwing off the accuracy? Even if the direct sound is being altered by reflected sound, the SPL meter is only 'hearing' what I would myself. No?

Refer to sploo's thread[that he has linked for you]. You can see examples of comb filtering. Notice the narrow effect of the comb filtering. Imagine some of your measurements coinciding with the deep and narrow nulls. You do not hear in such a narrow bandwidth[you approximately detect things tonally in a minimum of 1/3 band octaves]. But this narrow effect can really screw up your measurements, and it may not be represenative of the tonal response that you hear, as a result.

1. What frequency would be regarded as lower midrange?

About 250Hz.

2. I will check what format they are in on the CD, but I would imagine that they are .wav files. If so, I can send you those, but I don't know how to compress to mp3 format I'm afraid. I would also need you to PM me an email address to send them to.

I don't want the [probably huge in size] wave files. You could actually check the files yourself using a wave editor. Just open the files in a wave editor and not the relative maximum dB value of each one. If you need a freeware wave editor, let me know, and I'll provide an appropriate hyperlink.

1. I know you dislike subjective terms, but if I understand things correctly, going by your plots, you dislike a flat response because you would have to raise the amplitude of the upper frequencies, thereby giving rise to a 'bright' sound?

It is of my opinion that it sounds 'bright', yes, if 'bright' is defined as excessive amplitude of high frequencies.

2. I am relieved to see that your plot's average in-room response also reduces the upper frequencies; I had wondered if there was something up with my tweeters! But why would the 'average' room do this?

Air has a higher co-efficient for higher frequencies. Higher frequencies will be absorbed/damped by air at a greater rate per distance as compared to lower frequencies. Here is my theory why I dislike a flat response at listening position: most music is miced/recorded at a distance that is far closer than a realistic distance when listening as an audience member would represent. Therefor, the recorded music has far greater percentage of high frequency content as compared to a real performance, based on the listening perspective. Since I have experience listening to much unamplified music/sound, it is very annoying to me to hear exaggerated high frequencies, at least on music with considerable non-electronic content. I do, however, generally prefer a more 'flat' response when it comes to electronic and modern pop music.

Are you saying that you can select which frequencies to assign as the 18 (or so) bands, or do you mean that they are fixed, i.e. in a similar manner as I have at present, but at approximately half the intervals?

The Behringer is not fixed; you can manually specify the Q, frequency and amplitude of each filter.

If you are interested in doing computer measurements, I suggest you refer to the sploo thread, as it offers some useful information on computer measurement of your room. However, to do this correctly does require aquisition of a sufficient microphone, microphone preamplifier and software package. You can use a SPL meter for your purposes, but it will just take alot longer to get good resolution measurements[imagine having to manually plot in at least 1/12th octave increments for example, which still will not be as good as can be done with computer measurement].

-Chris

 

[sploo]

The presentation may be a bit dry, but the information is solid and useful.

Nonsense, I thoroughly enjoyed the video of you wandering round a room filled with balloons .

Seriously though, the videos are very informative and worth taking the time to watch.

Also, if you're really interested/mad/enthusiastic (delete as appropriate), try getting hold of F. Alton Everest's "Master Handbook of Acoustics". It's heavy going, but I'm working my way through it now, and it's most useful.

...I suggest you refer to the sploo thread...

Whilst it's very flattering to have someone refer (favourably) to one of your posts, twice, perhaps it's time to stop people having to refer to me by my user name (I know it's disturbing Ethan ).

Cheers,

Gordon.

 

[Buckle-meister]

If you have the RadioShack SPL meter, you can use its RCA output with a PC and Doug Plumb's ETF software (http://www.acoustisoft.com/).

Thanks very much for this. I have downloaded and installed it, but cannot use it yet; my computer has L/R line-in's, but not line-outs. It does have a headphones jack, but the output would be variable right?

Is there any way that I can get this to work?

Have a read through my bass traps thread...Some fiberglass/rockwool panels have made a huge difference.

Yes, thanks again. However, we appear to be tackling our problems from different ends; you appear to have concentrated on traps/absorbtion materials and note that you'd like to look into a PEQ, whereas because I live in a rented flat, I can't attach stuff to the walls/ceiling etc and would prefer (for the moment at least) to concentrate on what a PEQ can do for my ills.

...there's also a video on my company's site that explains comb filtering in detail with audio examples.

Very interesting. Thanks for this. Examples make understanding so much easier. Cheers.

Regards

 

[Buckle-meister]

I don't want the [probably huge in size] wave files. You could actually check the files yourself using a wave editor. Just open the files in a wave editor and not the relative maximum dB value of each one. If you need a freeware wave editor, let me know, and I'll provide an appropriate hyperlink...

I checked on a program I have, and actually it can SaveAs mp3 files, although the dialogue box that pops-up has me stumped. Instead, I have provided a screenshot of the 20Hz tone.

What exactly am I looking for?

I have experience listening to much unamplified music/sound...

This is probably a stupid question, but how can you hear unamplified sound?

If you are interested in doing computer measurements, I suggest you refer to the sploo thread, as it offers some useful information on computer measurement of your room. However, to do [computer measurements] does require aquisition of a sufficient microphone, microphone preamplifier and software package. You can use a SPL meter for your purposes, but it will just take alot longer to get good resolution measurements [imagine having to manually plot in at least 1/12th octave increments for example, which still will not be as good as can be done with computer measurement].

Any specific recommendations that I could investigate?, and isn't this method by Sploo doing a full sweep rapidly?:

If you have the RadioShack SPL meter, you can use its RCA output with a PC and Doug Plumb's ETF software (http://www.acoustisoft.com/).

This will take a complete 20Hz-20kHz room measurement in 5 seconds! It will also allow you to see the effect of your room's modes, and even decay times for each frequency.

Regards

 

[WmAx]

I checked on a program I have, and actually it can SaveAs mp3 files, although the dialogue box that pops-up has me stumped. Instead, I have provided a screenshot of the 20Hz tone.

What exactly am I looking for?

Unfortunately, I could not open the attached file. However, what you are 'looking' for, is the relative amplitude level[in -dB] of the frequencies compared to each other. What I was interested in, is if the calibrated CD was using a standard C weighting correction, or if the RS SPL meter had a not-quite standard C weighting curve. I would presume it's a standard C weighted curve, but I like to verify these details.

This is probably a stupid question, but how can you hear unamplified sound?

By amplified, I mean something that is electronically amplified. By un-amplified, I mean a live performance that does not use electronic amplification, such as a classical music concert, or music recital, etc..

Any specific recommendations that I could investigate?, and isn't this method by Sploo doing a full sweep rapidly?:

I did not think of the possibility to use the output jack from your SPL meter inputted to the computer. Yes, that is a perfectly valid method for low and mid-range frequencies. However, if you want accurate results for upper midrange and treble frequencies, you will need to use a measurement microphone and microphone preamplifier.

-Chris

 

[Buckle-meister]

Unfortunately, I could not open the attached file. However, what you are 'looking' for, is the relative amplitude level[in -dB] of the frequencies compared to each other. What I was interested in, is if the calibrated CD was using a standard C weighting correction, or if the RS SPL meter had a not-quite standard C weighting curve. I would presume it's a standard C weighted curve, but I like to verify these details.

Very odd. I can open the attachment no bother, on two different computers.

For the 20Hz tone, the attachment shows the sine wave centred along the horizontal axis at -90dB, and it's peaks and nulls reaching -3dB (the vertical axis increases from the horizontal axis each way).

Sorry for being really thick here, but I still have no idea what to do.

Am I comparing the -3dB peak/null value with the peak/null values from other frequencies? And if so, what frequency do I make them relative to?

Regards

 

[sploo]

Thanks very much for this. I have downloaded and installed it, but cannot use it yet; my computer has L/R line-in's, but not line-outs. It does have a headphones jack, but the output would be variable right?

Is there any way that I can get this to work?

This will be fine. I've used a laptop which had line in, and headphone out.

I set my receiver to 80dB (and I'd recommend ear defenders!), SPL meter in the 90dB range, and spent quite a bit of time fiddling about with the PC's playback and recording volumes in order to get good levels. However, once you're done, you're done - measurements from then on are easy.

Yes, thanks again. However, we appear to be tackling our problems from different ends; you appear to have concentrated on traps/absorbtion materials and note that you'd like to look into a PEQ, whereas because I live in a rented flat, I can't attach stuff to the walls/ceiling etc and would prefer (for the moment at least) to concentrate on what a PEQ can do for my ills.

Understood, though let me make the following two points:

1. You're always fighting a losing battle by doing EQ first. As the frequency response of the listening position will change depending on where you are in the room. This of course means that a boost or cut that's good for one location could be really bad for another. Oh, and an EQ won't help you with comb filtering (it's not going to remove reflections).

2. All the panels I'm making (bar the ones on the ceiling) are not fixed. The side wall panels are on stands, and can be moved around as desired. Even the bass traps (which have been cut with 45 degree edges) just 'push' into the corners of the room, so nothing's permanent.

Instead, I have provided a screenshot of the 20Hz tone.

What exactly am I looking for?

I think what Chris is after is a visual display of the complete sound sweep. If the peak sound level across the range is constant then you have flat frequency output.

If the peak sound level changes, then it's likely it's been weighted to match some particular SPL curve. I believe there's at least one CD around that's made specifically to match the RS SPL meter (probably in C weighting mode).

 

[Buckle-meister]

This will be fine. I've used a laptop which had line in, and headphone out.

Phew! Thank goodness for that. I had already started looking into purchasing an external sound card. Many thanks.....but......em......I still can't quite get it going.

I have a 1/4" to two RCA jack cable. The 1/4" leaves my headphones jack and the two RCA's go into the Left/Right Line-in jacks to create the necessary loop.

ETF has identified my 'SoundMAX Digital Audio' soundcard, and it has the necessary 'Rec Master' (with Line-in in it), 'Rec Soundcard', 'Play master', and 'Play Soundcard' frames (refer to attachment), yet when I press 'Level check', I get no response from the 1-second test signal.

Any ideas?

EDIT: I have just pressed 'Level check' with a pair of headphones plugged in, and ETF is definitely outputting the white noise test signal. Now, if I could just figure out why it's not hearing the Line'in's....

EDIT2(!): Have just run the headphones jack to my Z9's AUX jacks to hear the test noise through my mains. Nothing wrong with the cable then. Getting there...

Regards

 

[Ethan Winer]

Buckles,

> if I could just figure out why it's not hearing the Line'in's.... <

I never use ETF's mixer because I could never get it to work reliably with my sound cards. Also, in the Windows mixer there are separate screens to pick what sources are played versus which input is recorded. You need to go the the Input panel and select Line Input specifically.

--Ethan

 

[Buckle-meister]

Buckles,

> if I could just figure out why it's not hearing the Line'in's.... <

I never use ETF's mixer because I could never get it to work reliably with my sound cards. Also, in the Windows mixer there are separate screens to pick what sources are played versus which input is recorded. You need to go the the Input panel and select Line Input specifically.

--Ethan

HaHa! You are just ahead of me! I had just adjusted that and behold!: now it does register the tone.

Only thing is, even if I completely disconnect the cable, it still works. Geez! first I can't get it to work at all, then I can't not get it to work!!!

What on earth is going on?

By the way, assuming I get the connection working:

1. Presumably, since the single RCA jack on the RS SPL meter=mono results, I only need to connect the matching end of a single RCA-to-RCA interconnect to any one of my L/R Line-in computer jacks?

2. Is the RS SPL meter still set to C-weighting, and Slow response?

Regards