I just emailed Doug Plumb and ask him to chime in.

--Ethan

"Does anyone know how to set RPlusD to not make use of Filters that boost (when doing the automatic EQ fitting)?"


This is necessary to properly set cut filters. The dips in the response is always artifacts - these artifacts need to be removed to properly set the cut filters - ie dips in the response artificially shift peaks - the peaks need to be shifted back through dip correction before they are correctly EQed.

Hi all.

This is probably a pretty basic question but I have not seen it described anywhere. Normally, I am pretty good with setting up electrical connections but I do not seem to "get" how to hook up my system to use RDPlus with the following components.

Notebook computer
Firewire card
Presonus Firestudio
Power amp + speakers
Microphone (SM 58 and Rat Shack digital sound level meter)

I have a sneaking suspicion that the problem is in the setting up of the Firestudio mixer to deal with loopback but I am not sure what should plug into where on the Firestudio to create the loopback either. (I am a little nervous about looping something in the wrong way on my new toy).

I know that a better mic is needed also. That is coming soon.

It looks like some of you have used this or similar preamps and have gotten RDPlus to work. Any tips/diagrams would be appreciated. I have begun to pull together some room treatments and hope to use RDPlus to document how well each works - and maybe understand the why as well.

Thanks much.
--Hamp

Hamp,

I don't have a FireStudio, so I can't tell you about its mixer other than to say you should set it up to mimic a standard sound card. That is, you don't want the inputs echoed to the outputs, as is sometimes done for "zero latency monitoring." Also, the drawing below is from the ETF program's Help file, but with the addition of a Y splitter I added for clarification.

--Ethan

Ethan,

Thanks. With your diagram and a reread of the documentation I think that I have learned how to wire up the Firestudio (FS) for the loopback. However, I am not sure if in the end it is possible to use a FS as I cannot find a way to have RPlusD "listen" to any of the FS inputs. It seems that RPlusD (and this is a guess) is looking for an input signal from a "Windows line_in" source, i.e., one that shows up on the Recording Control dialog called "Line In." This would explain the statement in the docs about notebook computer sounds cards "that they generally do not work." They generally do not have the correct input connection.

In this case, it looks like I will have to invest in a simpler USB soundcard. Any recommendations? From the discussions above, it seems that pretty much anyone will do.

I have emailed Doug to inquire directly of him as well.

Thanks.

--Hamp

You need to tell Windows to use the FireStudio's line inputs as the default sound card for recording. This is in Control Panel, Sounds and Audio Devices, Audio tab.

--Ethan

Hi all,

I'd like to use a laptop with a usb mic (such as MXL mic mate).

Loopthrough measurements show no problem (+0, -0.5dB 20Hz to 20Khz) but the problem is that RplusD allows only same stereo soundcard input for L and R. If you could choose different inputs connectors for R and L, such as :
R = mic input (usb)
L = line input (loopback from L output)
this should work, I suppose...
So the question to Doug is if this possibility is on the development list or not. Or is there another way ?
I also tried using Virtual Audio Cable but with no success yet.
I also tried with usb input alone (no loop) but then my measurements are false, no impulse seen and speaker distance estimation sometimes 600 meters)

Any idea ?
Thanks a lot and my congratulations to Doug for his great soft.

I don't think this will work because the purpose of the loop-back is to assess the sound card. But in this case you'd go into one device and have the loop-back on the other. My best suggestion is to get an inexpensive USB sound card such as a SoundBlaster USB.

thanks Ethan,
I use an Edirol UA5 and it works fine with RplusD.
But this was more to simplify everything (and avoid travelling to friends with lot of boxes !)
But I wonder how it is done with the IBF usb preamp (without loopback as I read in the R+D manual)

Cheers
Jean-Luc

The loop back is internal with our USB unit.

If you are using other hardware, you can choose whether the the mic is used in the left or right input. The other channel not used for the mic becomes the loop-back channel. This selection is on the Options (yellow button) box in the main Audio Data Gathering form.

If you have an existing USB mic, it may not be possible to make the required connections.

(Thanks for the compliment on the software)

Hi all,
after my above posts, I found a way (a bit complicated) to use RplusD with my laptop and a usb microphone.
The loopback is done in software with Vitual Audio Cable. See hereunder.
But before using such a solution, and because RplusD cannot check the loopback performances, it is necessary to verify the frequency response of your mic+usb adapter and of you line output.

hello,
one of the main problem in low frequencies is that some room modes have quite high Q and so long decay times. In R+D, you can use waterfall plot to see those modes. But sometimes, if Q is high but amplitude low, it is not so clear on the waterfall display. Another way to show those "ringing" modes would be a groupdelay display. What does Doug think about such a feature ?

Thanks everybody
Jean-Luc

Hello all

I have conducted 4 series of measurements using RplusD to illustrate the changes in my room response while added some home made acoustic treatments.

All measurements seem fine except the RT/60 ones. I get values of '0' at several frequencies in each measurement series. Surely this can't be correct?

Why is this happening and what (if anything) can I d about it?

I don't have a direct answer for your question, but hopefully Doug Plumb will see this and reply. Or maybe you should email him through his site and send him a link to your post here.

--Ethan

I seem to recall options from the old ETF that allowed naming the measurement trial. I don't think I see that with R+D. Am I missing something, or are we constrained to "1", "2", etc.?

Hi all,

another question for Doug :
in fractional octave display, when you choose ie a gate time of 10ms and a BW of 0.33oct, you should have a display start near 300Hz but it starts only at 400Hz.
Same for 9ms, BW=0.15, display should start at 740Hz but starts at 1000Hz.
This is a factor of about 1.35, why ?
Thanks for any info.

jilo,

The 1/3 octave resolution is only preserved down to about 400 Hz. If the display was permitted to go down to 300 Hz, the graph resolution would be reduced and not be 1/3 octave, perhaps 1/2 octave.

You often see this with other "1/3 octave" displays. There is a loss of resolution as you get down to the lower part of the spectrum in the display.

In RPlusD I decided not to display this data rather than mis-represent it as 1/3 octave resolution.

eaf,

You are right, you cannot label individual curves in RPlusD. It assigns numbers to the curves. The reason for this is that a single RPlusD measurement was supposed to consist of several statistical curves to be averaged.

I cannot add this feature into RPlusD.

krusty,

The zeroed parts of the RT60 measurement are that way because there was too much noise at those frequencies to get good readings. Change your time limits and re-calculate.

Hi all
I'm new to the forum and am struggling to find info about measuring sound reduction with R+D.
I would like to know what the reduction between my live room and my neighbours house. I understand the software has a basic SPL meter (which must be calibrated - with a hand held meter?) but I would like to know the level of reduction at different frequencies printed on a graph.
Is this a possibility and if so where do I start?
Many thanks for your help with this
Mark

Yes, you could use R+D for this. You need to "calibrate" the SPL level before doing the first measurement. Otherwise R+D will "normalize" both readings and hide the difference in absolute levels. You'll also need a long microphone wire so you can measure on both the inside and outside of the dividing window or wall.

--Ethan

Thanks for the information, I will give it a try :o)

Mark,

Take a measurement with the mic on either side of the door. Compare the two measurements in any of the frequency response graphs.

The difference in levels between the two graphs gives your sound insulation effectiveness.

I would look at the full octave or 1/3 octave to filter out a lot of the artifacts.

No calibration of anything needed. You can use a crappy mic. Just make sure you have good s/n over range of interest.

Its the easiest thing ..sometime easy things warp our minds.

Mark,

BTW:

Use manual mic measurement instead of auto. Input mic/spkr dist to the nearest 0.5 meter or the nearest foot. This does not have to be accurate.

Gate for 200 ms or so.

What locations should I use for my test mic for 32 room measurements? I just took measurements from the small area in and around where my head normally is. Is that right? Or should I have been taking measurements from all over (back of room, etc.)?

Thanks.

Ken

EQ room wizard is free and RPlusD is $150 so which is the best product.