I am not sure. I set the meter to 80dB and then set the amplifier volume so that the needle was reading -5dB, or 75dB. I did notice that if I switch to 70dB, the needle only goes to about +3dB, or 73dB. If I increase volume so it goes to +5dB, then when I switch back to 80dB it reads -3dB, or 77dB.
To get an accurate figure from the RS meter it is better to set the level so it is on the upper half of the scale, the electrical output scales fairly accurately when you change the scale but the meter reading is so-so. So I would set the range to 70, adjust your levels so it reads 75dB there, then change the scale to 80 and tell REW that the level is 75dB in the cal dialog. Easier to do this using one of your main speakers rather than the sub. The actual cal levels are not critical though.

Is the AUX panel intended to be used for checking the entire system response, i.e. all speakers plus subwoofer, rather than just individual channels?
Can be for anything you like, just a spare measurement channel.

That's a strange one, do you mean that closing the app generates a noise, or that your volume levels are very high afterwards? When shutting down, REW restores (or tries to restore) the volume settings it found when it started up.

When I have everything connected, I can go to ETF or TrueRTA and run test, measurements and exit them with no problem. But if I go into REW, run test and measurements, and then exit the program. I immediately get a super loud humming sound from my sub and spealkers. When I say loud i mean loud in that you fear for your ears and speakers. Not sure whats causing it. But it happento me both times I ran REW and exited it.


I came across some SMS-1 response plots in a forum post, the Q figure is interpreted differently to that in the TMREQ setting. I've added an SMS-1 setting to this week's release, so try that. The behaviour you describe seems odd though, if you save measurement data as an mdat file, zip that and email it to me I'll see if it looks like a valid measurement.

Thats good to hear. I will give the new release a try and if I still get the same poor results in the measurements, I will save the data and mail them to you. I have to telll you I thought my room acoustic treatment helped the sound so much. But looking at the sweep 10hz to 20khz really makes me sad =p Sigh back to the drawing board. =p

Thanks for the great software.

Oliver

When I have everything connected, I can go to ETF or TrueRTA and run test, measurements and exit them with no problem. But if I go into REW, run test and measurements, and then exit the program. I immediately get a super loud humming sound from my sub and spealkers. When I say loud i mean loud in that you fear for your ears and speakers. Not sure whats causing it. But it happento me both times I ran REW and exited it.
Maybe there is some feedback loop being set up. To track it down, disconnect the soundcard output signal before exiting REW, and bring up the Windows playback and recording volume setting windows to see how the settings in them change when the Wizard exits. Perhaps line in is being selected in the playback mixer?

If I've got an ECM8000 hooked up to the RoomEQ Wizard, do I need C-Weighting on or off during measurement?

I've got the same issue with the DEQ2496 SPL meter and C-Weighting as I don't know if it should be on or off either...

Turn it off.

The ECM does not use weighting, therefore there is no need to turn on the inverse compensation. I don't know about the DEQ2496, but I suspect it is the same.

Brian

Try leaving the output (and even the input) device selectors blank (i.e. saying "choose device", then the OS should pass the app the default interfaces. Might work better.
That is the way I had it (no other choice really since it can't see the various "cards" attached )
What signal level gets used during the soundcard cal test? Would be interesting to see the screen capture of the Captured Data trace.
You can see it at (arg - message board software will not let me post
a URL until I've made 5 posts - lets try it this way.) Nope, no go.
If you are interested it looks like I have to make some more posts first before I can post the link, or I could PM the link to you, but, see below. . .

As all was OK with the measurement level and SPL cal can just skip the soundcard cal and try making some measurements.

I tried that, got the same error.
"Impulse peak is not where it should be, the measurement may have been corrupted
Check the Impulse Response and the Captured Data plots"

But, I found soundcard calibration works just fine if I use the built
in analog line in and outs on my mac! I suspect that making measurements this way will also work (and should be accurate enough for my purposes since your software corrects for sound card flaws). I guess apple's java doesn't work well with external sound devices (usb, firewire or optical). I just need to go out and buy some adapters to connect the line-in and -out to my stereo and spl meter. My main interest is to make low frequency waterfall plots of my room, and since your software corrects for soundcard performance and I'm really only interested in relative changes, I think this is going to work fine.

Thanks so much for making this software freely available, and using a "machine independent" language. I have other measuring software, but this is the only one I found that will create waterfall plots. Now I've got to go back and read the rest of your online help. . .

Here is the first graph. It seems like i have a large dip around 70hz, maybe there is some cancellation going on?
http://photobucket.com/albums/a79/Amigo207/?action=view&current=VelodyneDPS10.jpg

Edit: The boost i spoke about before wasn't accurate because i was measuring the SPL of my soundcard instead of the subwoofer itself apparently. After selecting to measure the subwoofer I got the above graph.

If you're using Behringer ECM8000 mic, turn off c-weighting and load this mic calibration file. Convert to .cal instead of .txt

That is the way I had it (no other choice really since it can't see the various "cards" attached )
Might need to change the default soundcard, assuming the Mac has a setting for that somewhere - I'd guess default is the built-in so that's what it was being handed by the OS. Glad it seems to be getting close to delivering results!

When I have everything connected, I can go to ETF or TrueRTA and run test, measurements and exit them with no problem. But if I go into REW, run test and measurements, and then exit the program. I immediately get a super loud humming sound from my sub and spealkers. When I say loud i mean loud in that you fear for your ears and speakers. Not sure whats causing it. But it happento me both times I ran REW and exited it.




same here for REW. On exit I get a super loud feedback loop type damage your ears and speaker hum/tone. I have to remember to unplug my input/output cables or turn off my amp every time before I exit REW.

If you're using Behringer ECM8000 mic, turn off c-weighting and load this mic calibration file. Convert to .cal instead of .txt

given that this is some sort of "generic" cal file and not specific to a specific ecm8000, how useful or required is it really? Can it do more harm then good?

Oliver, dm,

Could you send me copies of the soundcard debug file generated via the entry in the soundcard menu, please, might give me some clues as to why this feedback loop happens after REW has closed.

V3.29 is now available for download, changes are detailed in the Change History (http://homepage.ntlworld.com/john.mulcahy/roomeq/changehistory.html)

This version adds:

Impulse response import from WAVE files (format must be Signed PCM, with 16, 24 or 32 bits per sample)
Impulse response export at 16, 24 or 32 bits per sample with/without normalisation, two channels' responses can be exported as a stereo WAV
Controls on the Impulse Response Controls page to re-scale the impulse response to achieve a desired peak SPL figure in the frequency response
Display of the full range Energy-Time Curve in place of the impulse response with adjustable smoothing
An SMS-1 equaliser setting
Measurement notes for each channel
An overlay capability for the waterfall plot with variable transparency

http://homepage.ntlworld.com/john.mulcahy/roomeq/wizardhelp/images/waterfalltransparency.jpg
I have also changed the shortcuts to add channel selection from the keyboard via Alt+1,2 etc. This mechanism only works with the Alt keys, so I have had to reassign the graph group selection to Ctrl+1,2, etc and use Ctrl+Shift+3 to apply 1/3rd octave smoothing. To make it easier to see which channels have measurements in them, the Filters tab text is black if a channel has data, grey if not. The app also no longer restores the initial mute states on exit to see if this prevents the feedback loop some users have experienced, suspect this occurs if Line In was not muted in replay volume before startup

NO! I have to hold an extra key for smoothing! ;)

Very nice John. Would it be possible to save the impuse wavs in varied sample rates? MP3's are 44.1KHz, and the impulse waves are 48KHz, so when you convolve them there is a mis-match. I tried converting the wavs to 44.1KHz, but then they were shorter. It seems either way is not ideal, and the impulse response wont correctly sum with MP3's, correct?

I will try it out later. Format needed for the Foobar convolver is (from Cool Edit Properties of the file I have been using):

44100Hz, 32-bit, Stereo
Uncompressed Size: 940.81 KB (963,392 bytes)
File Format:Windows PCM
IEEE Float (0.24 float type 3)
Size on Disk:940.85 KB (963,436 bytes)
Last Written (local): 1/28/2006 19:40:03.190
Length: 0:02.730
00:00:02:21 at 30 fps
120,424 samples

Thanks for the new version
Darren

Would it be possible to save the impuse wavs in varied sample rates? MP3's are 44.1KHz, and the impulse waves are 48KHz, so when you convolve them there is a mis-match. I tried converting the wavs to 44.1KHz, but then they were shorter. It seems either way is not ideal, and the impulse response wont correctly sum with MP3's, correct?
The filter impulse response is generated at the current soundcard sample rate setting, so you can change the soundcard sample rate to 44.1k, export the filter response, then change the rate back to 48k (assuming that is the native rate for the soundcard).

Some soundcards use 44.1k, running them at 48k adds some noise to the impulse responses due to sample rate conversion. Here is a loopback impulse response from a 44.1k card running at 44.1k:

http://homepage.ntlworld.com/john.mulcahy/roomeq/wizardhelp/images/44kcardat44k.jpg

This is the same soundcard run at 48k:

http://homepage.ntlworld.com/john.mulcahy/roomeq/wizardhelp/images/44kcardat48k.jpg

You can also see the effect of running a 44.1k card at 48k in the top end of the loopback frequency response around the 22.05k Nyquist limit of the 44.1k sampling:

http://homepage.ntlworld.com/john.mulcahy/roomeq/wizardhelp/images/FR44kcardat48k.jpg

Best to run the card at its native rate, though 48k is the most common.

My soundcard control panel (and I suspect many others) has no option to change samplerates. Therefore, I can only output 48KHz samplerate wavs (unless there is a universal way to change soundcard sample rates).

I'm still learning on the whole samplerate subject. My soundcard, the Turtle Beach Santa Cruz, natively upsamples to 48KHz before outputing to the speakers if I am not mistaken. Using MP3's as a source, they start at 44.1KHz. Then, played in winamp, they are still 44.1KHz. Now the MP3 passes through the Winamp DSP convolver plugin, still at 44.1KHz, but is convolved with a 48KHz impulse wav. Next, the 'new' MP3 exits Winamp at (44.1KHz?), and enters the soundcard. Finally, the MP3 is converted to 48KHz in the soundcard, then output to the speakers.

Is my description of the process correct? If it is, isn't that 'oddball' 48KHz impulse wav skewed in time (or some other anomaly) compared to the 44.1KHz MP3? Or do I just not grasp samplerate interaction at all (which is entirely possible)?

And doesn't changing a 48KHz wav to 44.1KHz in a software program (such as Audacity) skew (or somehow distort)the original wav?

My soundcard control panel (and I suspect many others) has no option to change samplerates. Therefore, I can only output 48KHz samplerate wavs (unless there is a universal way to change soundcard sample rates).REW gives you a sample rate choice (44.1k or 48k) in its soundcard menu, that's the one I was referring to.

A convolver has to have both its inputs at the same rate, if it gets an IR at a different rate than the audio signal it is convolving with it should either refuse to convolve with it, or resample it to the rate of the audio first. If it were to just use it anyway the effective filter characteristic it is convolving with would be frequency-scaled compared to the original according to the ratio of the sample rates.

Unfortunately I'm still getting a POP when trying to make measurements. This I think contaminates the results. I guess I'm going to have to wait till apple makes a release 6 available (so far the latest they have is release 3)

Unfortunately I'm still getting a POP when trying to make measurements. This I think contaminates the results. I guess I'm going to have to wait till apple makes a release 6 available (so far the latest they have is release 3)
You can't directly compare Apple's and Sun's release numbering, Apple make their own implementations. The measurement method is very resistant to noise, so it may well be OK. If you make a measurement, save the resulting measured data set and send me the zipped result I'll happily take a look at it. A screen shot of the captured data graph page after a measurement would also give a good indication of whether the measurement is likely to be valid if you fancy posting one or emailing it to me.

REW gives you a sample rate choice (44.1k or 48k) in its soundcard menu, that's the one I was referring to.

I overlooked that option, thanks. A little cumbersome having to switch back and forth, but seems to work.

One thing I noticed, is that when exporting the Filter Response as wav, and then loading it back into REW, I see the filters I exported PLUS the meter calibration curve. This means that the convolved MP3's are gaining a ton of top end correct? is there any way to export JUST the filters response without having to clear the meter and soundcard calibration?

http://www.avsforum.com/avs-vb/attachment.php?attachmentid=51761/response 2.jpg

One thing I noticed, is that when exporting the Filter Response as wav, and then loading it back into REW, I see the filters I exported PLUS the meter calibration curve. This means that the convolved MP3's are gaining a ton of top end correct? is there any way to export JUST the filters response without having to clear the meter and soundcard calibration?The export is only the filters response, the cal files get added when loading an impulse response.

Great, then I should have a fairly decent corrected response. But that means any time a filter response wav is imported for viewing, we have to disable C weighting, meter calibration and soundcard calibration to see the correct frequency response curve correct?

Many people say that using a wav file (or files) for convolution isn't a great thing to do. Do you have any plans on exporting filters as a text file for use with convolution? I don't know what would be required, as I haven't read that deep yet.

Great, then I should have a fairly decent corrected response. But that means any time a filter response wav is imported for viewing, we have to disable C weighting, meter calibration and soundcard calibration to see the correct frequency response curve correct?Correct, but for the next release I'll disable cal data application and C weighting compensation for imported impulse responses.

Many people say that using a wav file (or files) for convolution isn't a great thing to do. Do you have any plans on exporting filters as a text file for use with convolution? I don't know what would be required, as I haven't read that deep yet.WAV files might not be a good idea for convolution if they are 16-bit and not normalised, but normalised 16-bit files or 24 or 32-bit files should be fine.

Thats good news. Can you point me in the right direction on how to read Impulse Response graphs (sorry if I missed it)? I think the readings before 0s are noise, but I don't understand how the next 300ms expreses the whole frequency range (i.e. 20Hz-20KHz).

Here is an attempt at posting a screen capture of the measurement from my mac

http://www.atmos.albany.edu/facstaff/knight/screen1.tiff

There seems to be useful information in the lower frequencies.
I think there really is a peak at about 4-5K (though I hope
it is not that big!). I assume the fall off above that is a result
of using the RS spl meter for a Mic. But everything above a
few hundred hz is noisier than I expected - maybe this is meter
noise, or, I need to a better job of making sure the room
is absolutely quiet during measurements. Maybe I need to
do this at a higher volume - the spl meter gave a reading of 75 db
on the pink noise calibration, I'm leary to go much higher with
that pop going through my speakers.

Perhaps the high frequency noise is real and associated with echos in my room (although the clap test doesn't reveal anything nasty)

And here is a picture of the osciliscope trace

http://www.atmos.albany.edu/facstaff/knight/screen2.tiff

showing the little measured blip at the beginning of the sweep

Thanks
David

John,

Good news on the loud dc like noise on exiting. It's gone with this new release. So your fix works. I always seem to get a measured response is at -22db instead of 10db even though I have made the calibration at 75db and left it at that volume for the sweep. Should I increase my preamp volume another 10db?

Also I understand that there is sweep in etf that tells you if you have correctly covered the first reflection points of your speakers. Is there a similar sweep or graphs in REW?

I was wondering out loud. It seems like in my system, when I measure in stereo mode and in PL2 mode (whcih I assume is almost similar to DD/DTS) I get different FR which can be adjusted out using the phase or delay timing. Is there a way to do a tone sweep in DTS/DD so that we can burn it in a DVD and use REW to measure the response so that we can ensure the best result?

Oliver

Thats good news. Can you point me in the right direction on how to read Impulse Response graphs (sorry if I missed it)? I think the readings before 0s are noise, but I don't understand how the next 300ms expreses the whole frequency range (i.e. 20Hz-20KHz).
I had a quick trawl around and didn't come across an article explaining room impulse responses, so I'll give a summary myself. If someone knows of a good article on the subject, do post a link.

The impulse response is in essence a recording of what it would sound like in the room if you played an extremely loud, extremely short click - something like the crack of a pistol shot, but shorter. The reason for measuring the impulse response (by more subtle means than firing a gun in the room) is that it completely characterises the behaviour of the system consisting of the speaker(s) that were measured and the room they are in. An important property of an impulse, not intuitively obvious, is that it if you break it up into individual sine waves you find that it contains all frequencies at the same amplitude. Strange but true. :) This means that you can work out a system's frequency response by working out the frequency components that make up its impulse response. REW does this by Fourier Transforming the impulse reponse, which in essence breaks it up into its individual frequency components. The plot of the magnitude of each of those frequency components is the system's frequency response.

When an impulse response is measured by means of a logarithmically swept sine wave, the room's linear response is conveniently separated from its non-linear response. The portion of the response you see before time 0 is actually due to the system's distortion - if you look closely, you may see that there are small, horizontally compressed copies of the main impulse response there - each of those copies is due to a distortion harmonic, first the 2nd harmonic, then the third, then the fourth etc. as time gets more negative. The period after time 0 is the system's response without the distortion.

In a perfect system of infinite bandwidth with totally absorbent boundaries, the impulse response would look like a single spike at time 0 and nothing anywhere else - the closest you get to that is measuring the soundcard's loopback response. In a real system, finite bandwidth spreads out the response (dramatically so when measuring a subwoofer as its bandwidth is very limited). Reflections from the room's boundaries add to the initial response at times that correspond to how much further they had to travel to reach the microphone - for example, if the microphone were 10 feet from the speaker and a sound reflection from a wall had to travel 15 feet to reach the microphone, that reflection would contribute a spike (smeared out depending on the nature of the reflection) about 5ms after the initial peak, because sound takes about 5ms to travel that extra 5 feet.

When measuring full range responses from loudspeakers (rather than subwoofer responses) the reflections are easier to spot as the higher bandwidth of the full range system keeps the spike of the impulse (and the reflections) quite narrow, but you need to zoom in on the time axis to see them. They are easier to spot with a linear Y axis (set to %FS instead of dBFS) and also show up more readily if you select the "Show Full Range Energy-Time Curve" option.

Hope that helps.

Here is an attempt at posting a screen capture of the measurement from my mac

http://www.atmos.albany.edu/facstaff/knight/screen1.tiff

There seems to be useful information in the lower frequencies.
I think there really is a peak at about 4-5K (though I hope
it is not that big!). I assume the fall off above that is a result
of using the RS spl meter for a Mic. But everything above a
few hundred hz is noisier than I expected - maybe this is meter
noise, or, I need to a better job of making sure the room
is absolutely quiet during measurements. Maybe I need to
do this at a higher volume - the spl meter gave a reading of 75 db
on the pink noise calibration, I'm leary to go much higher with
that pop going through my speakers.

Perhaps the high frequency noise is real and associated with echos in my room (although the clap test doesn't reveal anything nasty)

And here is a picture of the osciliscope trace

http://www.atmos.albany.edu/facstaff/knight/screen2.tiff

showing the little measured blip at the beginning of the sweep
The noisy appearance is comb filtering, the effect of reflections from surfaces of the room adding to the direct sound and each other and generating peaks and nulls. To see the response more clearly, try setting the trace smoothing to 1/3 octave (or press ctrl+shift+3). The RS meter is no good for measuring above about 1kHz, it meanders generally upwards until about 5-6kHz then drops dramatically thereafter. You need a decent mic and preamp to make meaningful full range measurements. The pop will have very little effect on the measurement, generally the results look reasonable (from a measurement perspective, plenty to be gained from an acoustical perspective!)

John,

I've measure my room, determined my filters settings, turned up the amps, remeasured and am generally happy. The new response is really good, but I'm using a lot of cuts and therefore a lot more amp. I intend to play with my filters to see if I can decrease the amount of cuts for similar results.

I'm wondering if there is some way that REQW could estimate how much additional amp power is needed due to the cuts to maintian a certain level -- just a simple precent, like 25% more power needed. It would just a nice number to know to balance how flat we want it and how much amp we're willing to use to get it.

Thanks again for the great program. Looking forward to trying the new version.

Good news on the loud dc like noise on exiting. It's gone with this new release. So your fix works.Glad to hear that

I always seem to get a measured response is at -22db instead of 10db even though I have made the calibration at 75db and left it at that volume for the sweep. Should I increase my preamp volume another 10db?You mean you get a message saying you can increase the volume? The default settings are fairly conservative, to allow for potentially large resonances. It's OK to increase the measurement level as suggested (either on your preamp or using the measurement level setting in the signal generator panel), it will give better signal-to-noise on the measurements - but remember to re-measure the target level after changing the volume.

Also I understand that there is sweep in etf that tells you if you have correctly covered the first reflection points of your speakers. Is there a similar sweep or graphs in REW?That is the impulse response, see post above.

I was wondering out loud. It seems like in my system, when I measure in stereo mode and in PL2 mode (whcih I assume is almost similar to DD/DTS) I get different FR which can be adjusted out using the phase or delay timing. Is there a way to do a tone sweep in DTS/DD so that we can burn it in a DVD and use REW to measure the response so that we can ensure the best result?You get a different result because of the effect of the PL2 processing, you are measuring the room behaviour plus the PL2 response. Getting a good result with the Stereo setting will give a system that gives good results for the other modes.

I'm wondering if there is some way that REQW could estimate how much additional amp power is needed due to the cuts to maintian a certain level -- just a simple precent, like 25% more power needed. It would just a nice number to know to balance how flat we want it and how much amp we're willing to use to get it.
You can get a good idea by measuring the sub level using the pink noise test tone with/without the cuts applied, the difference in level is roughly how much boost you need to get back where you were - so if the level was 75dB before starting, and afterwards (before readjusting the sub level) it measured 69dB then you are having to add 6dB = 10^(6/20) = 1.995 times as much power as before. When using an AV processor to set the sub level, it is just the difference in the sub level trim results before/after. How much that is depends on what the original power was, which can be tricky to determine.

John,
Thanks for the reply. Yeah, I know that I can test it to figure it out. I was just thinking that it would be cool as I play with the filters to have the program tell me how much was needed - a running tally that would get added up as each filter is added.

Not a big need, just kind of cool info to know. Thanks for considering it.

John,

Thanks, I am starting to understand I think. Lets take your soundcard loopback measurement graphs you posted above. Before 0 seconds, we see harmonic distortion. At 0s, we see the 'frequency response' of about 20-20KHz. From 0s to 300s we see what, the decay of the impulse due to the soundcard not having high enough dampning to silence the impulse immediately?

Hello John,

One suggestion and one theoretical question.

1) How about adding a mic to speaker distance calculation? I'm assuming a electronic calculation will be more accurate than a physical measurement in getting your speakers equidistant. Plus if you must have speakers at unequal distances (my rears) it would allow you to confirm that your processor has been set to the correct distance differential.

2) From my reading it is apparently very difficult to electronically measure the distance of a subwoofer with the crossover engaged, 110Hz in my case. Are there any clever ways of doing this? Since the LP filter of sub can apparently use quite a bit of time and is different depending on the Fc and the slope, this would allow us to see precisely just how much time.

Many thanks as always,
George

Thanks, I am starting to understand I think. Lets take your soundcard loopback measurement graphs you posted above. Before 0 seconds, we see harmonic distortion. At 0s, we see the 'frequency response' of about 20-20KHz. From 0s to 300s we see what, the decay of the impulse due to the soundcard not having high enough dampning to silence the impulse immediately?What you see peaking at 0s is the soundcard's attempt to produce a single-sample peak. The filtering on soundcards is such that there is generally a little bit of pre-ringing, so the response starts before the peak comes along, then the peak appears and decays, an initial very rapid drop then a much slower gentle curve. The two factors affecting the response shape you see are the upper and lower frequency roll-offs of the soundcard. The higher the upper limit (i.e. the less the high frequency roll-off), the narrower the peak is. The lower the low frequency limit is, the broader is that long curve that stretches out from the peak - it is caused by the high pass part of the soundcard's response which removes the very lowest frequencies. You will see a similar shape if you export the impulse of a High Pass filter with a low cutoff (in the Generic setting you can go down to 10Hz for the High Pass). If you set your input to "Stereo Mix" or similar to loop back internally you will sometimes find the response extends to DC, in which case that slow decay after the peak is absent. Note, however, that this is quite different to the slow decay of the overall response you see from a room measurement, which is caused by the rate at which the sound in the room dies away (essentially the echoes of the impulse fading into the background noise).

1) How about adding a mic to speaker distance calculation? I'm assuming a electronic calculation will be more accurate than a physical measurement in getting your speakers equidistant. Plus if you must have speakers at unequal distances (my rears) it would allow you to confirm that your processor has been set to the correct distance differential.
Unfortunately that can't be done without having an extra loopback connection, as the software has no reference against which to measure the time delay. There are variable delays in the soundcard output buffering and input buffering which cannot be determined without a local loopback. Having the local looopback does complicate some other parts of the setup process however, which is why I dropped it a while back. ETF does a good job of measuring time delays, but in most cases a tape measure will be all you'll need.

2) From my reading it is apparently very difficult to electronically measure the distance of a subwoofer with the crossover engaged, 110Hz in my case. Are there any clever ways of doing this? Since the LP filter of sub can apparently use quite a bit of time and is different depending on the Fc and the slope, this would allow us to see precisely just how much time.There's no easy way to handle that, the phase shift through the crossover filters is quite significant at those relatively low frequencies and variable phase controls on the sub add to the complication. Added to that is the complex interaction between the boundaries and the locations of the drive units and ports. Trial and error and lots of measurements are really the only way to find the best settings for the sub.

Guess I will have to stick to reading frequency response graphs. I am just trying to simplify the impulse response graphs into 3 parts and what they represent so I understand it.

Pre peak = harmonic distortion
Peak = frequency response of the speakers
Post peak = room response

or something like that. Otherwise I guess I am just used to thinking time vs. frequency and can't 'change modes' in my thinking.

Guess I will have to stick to reading frequency response graphs. I am just trying to simplify the impulse response graphs into 3 parts and what they represent so I understand it.

Pre peak = harmonic distortion
Peak = frequency response of the speakers
Post peak = room response

or something like that. Otherwise I guess I am just used to thinking time vs. frequency and can't 'change modes' in my thinking.
That's broadly correct. The period from the peak to the arrival of the first reflection is due to the direct sound from the speakers, but this time is usually so short that it only gives information about the speakers' response above a fairly high frequency (if the first reflection appears after 2ms, say, the lower limit for the direct sound is 500Hz). Below that frequency it becomes very difficult to separate the speakers' contribution from that of the room.

After Reading this, I know why I'm not an engineer...

So, no one has said how much better the sound is, or an example of something they listened to before and after, and what they thought.....Let's hear what you guys think about the sound now that you've been using the tool....I'm thinking of jumping in too!

How do I know if my laptop (DELL D800) has mic in or line in?

Does it support the FBQ3102 or 1502?

Stieger

stieger,

Using the program has made a huge difference for me although I don't use it exactly the way John intended. The thing that made the biggest difference for me, was just running a ton of sweeps with my mains in a bunch of different places until they were as flat as possible. Then I ran a ton more sweeps with my two 10" subs in different positions and using different phases. Finally, I used it as intended (basically), and programmed a bunch of filters in the program, then exported them for use in convolving engine. Even without the filters, just finding the best speaker positions made the sound better than it ever was.

And this is just for my main computer system. Eventually I will be tuning my theater with it, using it to program my BFG to 'tune' five 15" Infinite Baffle subs...



Look at the RED response graph. That is basically the way I was listening to my system (it was actually worse, I don't have the original response saved). It was always boxy sounding from the broad peek around 115Hz, and hollow from the broad dip around 250-300Hz.

Now look at the BLUE response. The peak is mostly gone, and the dip is totally gone. The overall response is much smoother. BOTH these response curves have NO filters active, but you can see the blue one is much better. With the filters I 'designed' in REW, I was able to totaly remove the peek at 155Hz, and lower the output from 30-70Hz leaving a small 'house curve' without the boxyness or hollowness I had before.

John,

I guess one of the confusing points in my mind is that you say you can measure the entire 'system' frequency response with a short click impulse response. But yet REW uses a log sweep to allow measurement of the rooms response. Since we have an approximate 5 second sweep, how is the impulse response derived? Reverse Fourier Transform? Otherwise I don't understand how we can get a short loud click impulse response to look at in REW from a 5 second frequency sweep.

The noisy appearance is comb filtering, the effect of reflections from surfaces of the room adding to the direct sound and each other and generating peaks and nulls. To see the response more clearly, try setting the trace smoothing to 1/3 octave (or press ctrl+shift+3). The RS meter is no good for measuring above about 1kHz, it meanders generally upwards until about 5-6kHz then drops dramatically thereafter. You need a decent mic and preamp to make meaningful full range measurements. The pop will have very little effect on the measurement, generally the results look reasonable (from a measurement perspective, plenty to be gained from an acoustical perspective!)

Thanks John. I know my room needs treatment :) Right now I'm trying to get a baseline measurement, and tweak speaker positions. This way I have something to compare against after I add treatment (my thinking is something for first reflection from walls and ceiling, and, as many bass traps (or really broadband absorbers) as the wife will allow).
Thanks for your work on this program, and, for making it freely available!

Okay guys... this is a really basic question ;)
I've got a Radio Shack SPL meter and AVIA DvD on the way direct from SVS, and i'm wondering if the Radio Shack SPL meter will work as a microphone, with a cable (what kind?) to my soundcard, to use with the RoomEQwizard?

The reason i ordered the SPL meter and AVIA was that, in discussion with a tech at SVS while trying to determine what sub of thiers would be most suitable for my situation, he suggested that i get em, take measurements, and report back ;) heh
... They seem very willing to help, A+ in that department for SVS!

Thanks fellas,
Mo

Have you read ANY page of this thread? The Radio Shack meter has been mentioned probably in just about 33 percent of the pages, also only requiring 2 mouseclicks from the software homepage, in the help file. I'm not trying to be an ass, but c'mon, a little reading isn't gonna kill ya ;) .

I guess one of the confusing points in my mind is that you say you can measure the entire 'system' frequency response with a short click impulse response. But yet REW uses a log sweep to allow measurement of the rooms response. Since we have an approximate 5 second sweep, how is the impulse response derived? Reverse Fourier Transform? Otherwise I don't understand how we can get a short loud click impulse response to look at in REW from a 5 second frequency sweep.All you ever wanted to know about system response measurement using sweeps (and more) can be found here (http://www.anselmgoertz.de/Page10383/Monkey_Forest_dt/Manual_dt/aes-swp-english.PDF) in an excellent paper on the subject.

Thanks John. I guess I was straying away from REW program usage and heading toward measurement practices and theory. I appreciate the time taken, and that paper seems to be geared just about right for my thought process.

EDIT: You just pointed me to that paper to delay me from asking anymore questions, because it is so long, right? ;) But it seems I did have a basic grasp, as it appears impulse response is infact derived from an Inverse FFT. Thanks!

At the time, i'd only read the first few posts. I figured i'd cut straight to the chase and ask a direct question. Your answer could have been a direct one like 'yes' and 'cable XYZ' instead of an entire paragraph, but I digress; you had to refer a little vaguely to almost 20 pages of posts to sift through, heh :P

Anyhow, i'll look for the reference on the RoomEQ Wizard website.

Thanks.

There's a 'search this thread' tool in the upper right hand corner of the page. Try that with the word 'Shack'.

lol, i never saw that. Only ever used the search forum one.
I'll have to use that, later.

The radio shack spl meter has a RCA jack. You need a cable with an RCA plug on one end, and, whatever your soundcard has on the line level input.

Thanks Dknightd, much appreciated confirmation of what the Radio Shack SPL meter has for connectivity.

Hi folks,

Is the Behringer DEQ2496 supported?

Thx,

Eric

You get a different result because of the effect of the PL2 processing, you are measuring the room behaviour plus the PL2 response. Getting a good result with the Stereo setting will give a system that gives good results for the other modes.

My Rotel processor for some reason sets the crossover for the mains at a slightly different frequency (about 10-15hz higher in PLx mode) then in stereo mode even though there is only a global crossover setting for each speaker. Good thing is that I can use a different crossover with my sub as I can use a differnt preset from my SMS-1 to work that. Of course the other issue I have is that I do not know if in DD mode which cross over settings it uses! =p That was why I was hoping I can get a quick sweep burned in DD format so that it can be played via my DVD player and measured via REW.

Has anyone else encountered that their reciever or processor uses different crossover settings in stereo or PLx or DD mode even though there is no way to adjust it?

Thanks
Oliver

Is the Behringer DEQ2496 supported?
The filter shapes should match those of the FBQ2496, but Midi download of filter settings is not supported - for some reason Behringer like to use different Midi commands for every product :rolleyes:

At the time, i'd only read the first few posts. I figured i'd cut straight to the chase and ask a direct question. Your answer could have been a direct one like 'yes' and 'cable XYZ' instead of an entire paragraph, but I digress; you had to refer a little vaguely to almost 20 pages of posts to sift through, heh :P

Anyhow, i'll look for the reference on the RoomEQ Wizard website.

Thanks.


SOME of us do a lot of work in reading about different subjects. When someone comes along and asks us to do the work for them, especially with something so basic, we can tell the person isn't willing to do a little work for themselves. it just irritates us a bit. I know someone who has a computer and can use it, but rather than turn it on and use Google, he will call me and ask me the question. He does very little by himself, but relies on others because he is lazy. I am just explaining this because you may come across others that will be downright mean in comparison to me with their replies to you if it is obvious you didn't try and find the answer yourself.

Anyway, please read the help files on the authors website before asking anymore questions. Your question was answered on the second page, and even has pictures to help the beginners. The author works on this program for free, and the less already covered questions he has to answer, the faster he will be able to improve the program for all of us. Not to mention all the work he put into the help files to answer our questions.

That was why I was hoping I can get a quick sweep burned in DD format so that it can be played via my DVD player and measured via REW.


This may not be much help, but in the past, I was able to download or generate a 20-20KHz sweep, then encode it to 6 different files with a dolby digital encoder, each one having just 1 channel - Left, Center, Right, etc.

That was years ago, and I used a now discontinued Dolby Digital encoder from Adobe, and I don't remember the source of my sweeps. And unfortunately, it seems REW can't measure sweeps from an external source. I believe I used TrueRTA which has a sweep plotting mode and can use external sweeps.

JohnPM, a "concepts" question about results vs. the R+D functionality in ETF ...

My understanding of the R+D functionality in ETF is that it helps find "stable peaks" that are more a function of the room than the speaker/chair positioning ... essentially finding peaks that both recur often throughout many positions in the room and have long/high ringing. (The peaks it finds appear to be large in amplitude and very narrow in width.)

If I had those identified, would I get "superior" results if I eliminated them first via PEQ and then used your program to generate corrections for "broader anomolies", or, would that be redundant with what your program is doing anyway?

Put differently, is there a "synergistic" way to use R+D with REQ, or, is R+D largely redundant to a good REQ solution?

JohnPM, a "concepts" question about results vs. the R+D functionality in ETF ...

My understanding of the R+D functionality in ETF is that it helps find "stable peaks" that are more a function of the room than the speaker/chair positioning ... essentially finding peaks that both recur often throughout many positions in the room and have long/high ringing. (The peaks it finds appear to be large in amplitude and very narrow in width.)

If I had those identified, would I get "superior" results if I eliminated them first via PEQ and then used your program to generate corrections for "broader anomolies", or, would that be redundant with what your program is doing anyway?

Put differently, is there a "synergistic" way to use R+D with REQ, or, is R+D largely redundant to a good REQ solution?Both programs have much the same aim as far as room EQ is concerned, identifying and countering modal resonances. In R+D Doug emphasises the benefit of making multiple measurements around the room and has done excellent work on system identification to characterise modes. The last year or so on Room EQ Wizard has been mainly concerned with providing its measurement capabilities, now that the app has reasonable measurement functionality I'll be devoting time to improving the process of identifying and correcting resonances. As tools for aiding the setup of a PEQ either app can be used; R+D has a more comprehensive set of capabilities and also offers many other acoustic measurement and analysis features, REW is perhaps a little more accessible within its scope.

This may not be much help, but in the past, I was able to download or generate a 20-20KHz sweep, then encode it to 6 different files with a dolby digital encoder, each one having just 1 channel - Left, Center, Right, etc.

That was years ago, and I used a now discontinued Dolby Digital encoder from Adobe, and I don't remember the source of my sweeps. And unfortunately, it seems REW can's measure sweeps from an external source. I believed I used TrueRTA which has a sweep plotting mode and can use external sweeps.

Thanks Trymor. I did not know that REW is not able to measure a external source or sweep. But I will check to see if there is a free encoder available on the web.

John =p Any chance you are planning to make available a way for REW to measure a external response or sweep? I am in communication with Rotel on my apparent differnet crossover issue and apparently it is not that they change the crossover but they use different slope for stereo and for DD/DTS. :(

Oliver

John =p Any chance you are planning to make available a way for REW to measure a external response or sweep?
No, no plans for that.

Brucemck2,

I have not used R & D, and you may know the following about REW: You can take multiple position measurements and use the averaging functionality of REW before deciding on your filters. You also have the option of changing the measurement window, to allow more or less of the room interaction into your measurements.

That being said, and remebering that I have not used R & D and have only measured for 1 position, I would consider using REW first, due to it's ease of use for the targeted application, then decide if the results are satisfactory. If not, then try R & D on the equalized system if you think the response could be better.

Just my 2 cents.

John,

Have you given any advice yet, or could you discuss, how to eq two subs using two channels of the BFD?

I've done a pretty good job of eq'ing them together, with one set of filters. But, I'm thinking that I may get better results at more seats and with less eq if I eq them seperately.

I tried to do a little searching, but now that this thread is 20pages, it is getting difficult.

Thanks

Ryan,

I haven't tried EQ'ing a pair of subs, but I'd approach it in similar fashion to a pair of main speakers: measure each independently, then both together, then individually EQ the two responses ignoring any peaks in the individual responses that were not present when both were measured together.

As to the thread length, it is getting rather unwieldy. I've added an item to my task list to produce an FAQ from the contents of this thread and the emails I receive, will try to get to that sooner rather than later.

Thanks John,

You're the best.

I guess this is a basic and possibly idiotic question...but I've perused the manual and don't see the answer, though the reply two posts abovee seems to touch on this.. Iif I simply want to measure the room/speaker interaction of my front left/right speakers ,in order to determine which positions for them result in the flattest low frequency response at the listening position, do I connect the right soundcard output to *both* the L and R inputs of my AVR (via a y-cable) or do I deal with the left and right speakers sequentially, one at a time?

The measurement signal is generated on both left and right outputs, so you can connect the soundcard left/right to the AVR left/right for measuring with both speakers running and disconnect left or right to measure either speaker alone.

EDIT: oops, John answered while I was typing this, but I will leave it up anyway.

Both of the soundcard output channels carry the test signal. A 'Y-cable' used to refer to a cable that splits a single mono connector into 2 leads. Use a STEREO 3.5mm to phono adapter cable (which is now called a Y adapter cable, along with any cable that has a single end and a double end) OR a standard stereo phono to phono patch cable, depending on your soundcard. You can then use either your AVR's balance control, or the soundcard mixer's balance control to change between left and right measurements. Perhaps John would consider explaining this in a little more detail in the help files.

As far as your measurement technique, John covered it in his (second) most recent post (#564). Personally, I measure both channels together, considering most (not all) LOW bass is mono, and played out of both speakers at the same time. Thais will get you close enough if you do not feel the need to get everything as absolutely perfect as possible.

Hi folks,

I set up a Behringer 6200 yesterday for the front L/R of my theater (using Triad inRoom Golds), and although I don't have my 2496 yet, I hooked up the L/R and adjusted some areas. Added some mid-bass, and man, sound is really cool! I had a few of the higher frequencies turned up past -0-, and noticed a hissing noice which wasn't acceptable, but all in all I like the Behringer - no noticable hum or feedback on the system, and only hear what I added via the EQ.

Anyone here using the 2496 with this software? What were your impressions? (I have read every page by the way!!)

On another note, even though the RTA may indicate what we should turn up or down on the PEQ, isn't everybody's ears different and hence I may actually need a "boost" in one frequency in order for my ears to hear it, when in fact the RTA is saying"cut by 3"?

Just wondering if it would make 'some' sense to EQ the room based on the liking of the sound to our ears, vs. doing it based on a 'flat" frequency response that may hinder our own ability to hear somethings from the soundtrack...


Best,

Stieger

The measurement signal is generated on both left and right outputs, so you can connect the soundcard left/right to the AVR left/right for measuring with both speakers running and disconnect left or right to measure either speaker alone.


Thanks. Btw, I've made a .pdf compiling the current REW help html pages into one document -- let me know if you want to host it , and I'll send it to you.

EDIT: oops, John answered while I was typing this, but I will leave it up anyway.

Both of the soundcard output channels carry the test signal. A 'Y-cable' used to refer to a cable that splits a single mono connector into 2 leads. Use a STEREO 3.5mm to phono adapter cable (which is now called a Y adapter cable, along with any cable that has a single end and a double end) OR a standard stereo phono to phono patch cable, depending on your soundcard. You can then use either your AVR's balance control, or the soundcard mixer's balance control to change between left and right measurements. Perhaps John would consider explaining this in a little more detail in the help files.


more connection(s) diagrams would be a good addition to the help files.


As far as your measurement technique, John covered it in his (second) most recent post (#564). Personally, I measure both channels together, considering most (not all) LOW bass is mono, and played out of both speakers at the same time. Thais will get you close enough if you do not feel the need to get everything as absolutely perfect as possible.

OK, now suppose it's a 2.1 setup -- I realize that the mains should be measured with bass management enabled, but should the sub be measured separately from mains, or does one normally measure the whole system?

I've made a .pdf compiling the current REW help html pages into one document -- let me know if you want to host it , and I'll send it to you.That would be great, thanks.

stieger,

Yes everybodys ears are different, but grossly so, not selectively. The overall frequency curve may be different, but everyones hearing at say, 12KHz specifically doesn't vary by +/- 3db compared to rest of the spectrum (unless there is something wrong with their ears).

My opinion is to adjust flat (other than possibly a house curve, which is a different subject than a midbass bump), then get used to the sound. If you want to hear more details, get better speakers. You can have 2 different speakers, BOTH adjusted FLAT, and hear a LOT more detail from the very good speaker, compared to an inferior speaker.

The brain has good 'hearing memory'. If you are used to hearing say, a moderate midbass bump all the time, any system without one may not sound good to you. Adjust flat, listen to the system for a month, then change it back for a minute. If you still like the non-flat sound, then keep it that way.

As far as hearing things in a soundtrack, how do we know what the sound engineer wants us to hear? He may want some things barely audible compared to something else.

OK, now suppose it's a 2.1 setup -- I realize that the mains should be measured with bass management enabled, but should the sub be measured separately from mains, or does one normally measure the whole system?

I think I remember reading an answer to that in the last 5 or 6 pages, but if a person is not picky or is lazy, measure the system and call it good. If a person is anal about sound he/she should read this thread from page 1 (mabey even twice), then decide for themselves if it should be done another way.

EDIT: The above refers to EQing. For finding the best position for the sub, measure it separately from the mains (after positioning them), then measure them together to make sure the 'combined positions' don't create any big nulls.

Okay guys... this is a really basic question ;)
I've got a Radio Shack SPL meter and AVIA DvD on the way direct from SVS, and i'm wondering if the Radio Shack SPL meter will work as a microphone, with a cable (what kind?) to my soundcard, to use with the RoomEQwizard?

The reason i ordered the SPL meter and AVIA was that, in discussion with a tech at SVS while trying to determine what sub of thiers would be most suitable for my situation, he suggested that i get em, take measurements, and report back ;) heh
... They seem very willing to help, A+ in that department for SVS!

Thanks fellas,
Mo


The Radio Shack mic is useless. Rolls off big time in the highs and lows. I Compared several cheap mics. Radio Shack, one that came with my PC and one my daughter had with a Karaoke unit. The PC mic look like it was most accurate - it didn't roll off on either end. (It looked to be good below 30hz)

Oh so many questions arise from your statement. First and formost - what did you 'measure' the mics with? Do you have a perfect loudspeaker in an anechoic chamber that plays perfectly flat from 10Hz to 22KHz? Do you have a mic calibration system thats costs thousands of dollars?

C-Weighting and microphone calibration files exist to allow imperfect mics to measure correctly, and at least predictably. I'm not going to address any mic specifically, because thats been done before. There is also the option of sending a mic in to be professionally calibrated for around $40 US of course.

REW is designed to find room modes below around 500Hz. Mic selection is not as important as it would be for say, speaker building.

"My opinion is to adjust flat"

AFAI can tell, most everyone agrees, until they try it.

It's at odds w/most professional's opinions.

There's the X-xurve to compensate for the way different size rooms absorb high freq differently.

And what has come to my attention recently and I believe more important - unless you have constant directivity speakers with flat power response, the mike will hear the reduced treble output of the room response, but you will hear a hot on-axis response if you EQ to compensate.

Can I amend my statement to "My opinion is to adjust flat below 500Hz"? :) I didn't specify, but was focusing on frequency response anomalies, specifically the thought of boosting the midbass above flat in the post I was responding to.

I definitely advise all looking for full frequency system response perfection to read some of the threads that Noah participates in. This is not meant as a sarcastic remark, Noah!

I do not EQ above 400Hz myself, and my listening position is about 3.5 feet back from my mains. The blue curve in the response link below is my current response at the listening position BEFORE using REW to EQ the bump around 155Hz and lowering the response from about 27Hz to 70Hz, if anyone is interested. There is no room treatment or EQing at all.

Noah, do you believe my high end is hot? Personally, my sound always seems better when when the overall response is close to flat, but of course, sound correctness can be very subjective and personal.

System Response (http://www.avsforum.com/avs-vb/attachment.php?attachmentid=51796)

Hi folks,

Want to ensure what I'm doing makes sense...

I have a Behringer 6200 EQ, which does generate pink noise, so I'm using it to measure with a RS SPL meter set at 75dB. I have the SPL meter connected to my line-in on my computer. I am getting measurements with the pink noise generator on the Behringer 6200, and after getting readings from this software, I'm making corrections.

My computer does not have a line out, so I'm not connecting anything to my Anthem AVM30 Preamp/Surround Sound Processor - Does this Matter???

Also, does the software generate the sine wave, or other audible acoustic measurements, or is that supposed to occur ONLY through the FBQ? OR does the FBQ provide a sine wave (I really don't know what a sine wave is, unless it's a 20Hz - 20kHz sweep signal). I don't know of any processors that provide that, unless that's why you are using the FBQ1124...

I seem to only get waterfall plots, and can't get "find peaks" to work...


Stieger

The 'line out' on your computer is the jack your computer speakers would connect to. And yes it matters. You must have your computer output connected to your preamp input to use all the functionality of REW. The software generates all of the test signals. Do not use the generator of your EQ.

I'm going to try to use REQW for the first time this evening, solely towards finding optimal speaker positions (flattest low-[frequency room response at the 'sweet spot', by varying the speaker positions). I jsut want to run the setup by this forum to see if I've got hte right idea going in. I intend to use a C-weighted RS analog sound level meter as a mic, and run its ouput into the right RCA input of an M-audio 2496 soundcard, part of the PC running REQW. Output from the soundcard will be R ch out to the R or L channel in of my AVR; I will disconnect speakers that I'm not positioning if need be. I'm using the 'CD' input rather than the multichannel analog panel, since bass management is not applied to the latter. (I wonder if REQW can ever been enabled to output a bitstream-type digital output instead of analog, such that channel direction can be encoded in the signal -- thus the user can select which channel to test, at the software side, and use an optical or coaxial digital connect to the AVR). All DSP other than bass management, speaker level, and speaker distance compensaton will be turned off.

So far so good? I'm a bit unclear about what to do about levels and distance comp -- as I think about it, it seems I'm going to have to change at least the distance comp evry time I move a speaker. Would it be better to simply take the measurement with ONLY bass management on, with no level/distance comp?

Krabapple, setup should be fine. No need to adjust distance settings unless you are measuring sub and main together, irrelevant when measuring a single speaker on its own.

Thanks. Real life intervened, and I wasn't able to get to it last night. ...looks like it will be another few weeks before i can try again. ;<

Hi folks,

Just got my DEQ2496 in the mail - great unit!

Quick question for those on this thread who use it - in order to use the RTA on the unit, you need a mic (SPL meter).

Have you guys used the RS SPL meter with the internal RTA on the DEQ2496?

Does the unit itself create the sine wave or pink noise, or do I need to generate that through my preamp/SSP?

Thx!

Stieger

stieger,

That question does not belong in this thread. It has NOTHING to do with REW, which is what this thread is about. You should use REW instead of any of the tools built to that unit because REW is much more advanced in precision and functionality. The only thing you should be using are the EQ filters.

I you decide to try and use REW, go ahead and ask more questions. If you are just going to use hardware for measurement and EQ, please ask you questions in a different thread.

John,

Is there anyway that you could add the ability to send a pink noise signal to each of 5.1 channels through digital out?
There has been some discussion around the forum lately regarding the inadaquecies of the test DVD's to send the proper levels to each channel for calibration. I thought that by being able to send a fixed "proper" level using the same connection as the DVD input, we might be able to achieve THX reference settings more easily. Just an idea.

Thank you as always
Darren

Darren,

Quite tricky as would need a Dolby encoder to generate the relevant stream (with associated licensing issues) and a way to pass that stream to the soundcard for output bypassing any mixers/level controls that might modify it. I would have more faith in the receivers' built-in test signals.

I have a problem when I click "Find Peaks", at which point I receive an error msg that my 'target level is to low'. Could someone point out the (obvious) problem here? I followed the directions step by step until this happened.
Subwoofer - Cutoff 80hz - Target 75db
Room Size = 28Wx40Lx7H feet.
Post Ref Window Width = 500ms
Enable REW's: METER/Compensate for C weighting.
Calibrate SPL @ 75db w/ RS meter (C slow, range 80) from listening position w/ boom mic.
Mic is the ECM8000 w/ cal file loaded.
Preamp = Rolls Mini Mic
FR showed up on the graph with a large & wide 23db bump at ~40hz, and Find Peaks generates the error at this point.
This was my first attempt & I assume if I complete the setup w/ a 75db ref level, shouldn't the remainder of the testing adhere to that same ref level, or does that large of a peak require a change from 75db?
Regards,
Ward
--Aha, rereading the help files:
Set your meter to C weighting. If you are using a microphone and preamplifier for measurement, turn OFF the Wizard's C weighting compensation in the Meter menu.
-Perhaps that was the problem. I'll start over again & give it another go.

That it is not an error message, it is a warning that is issued if a peak is found that is more than 24dB above the target curve (which includes the bass management response). Peaks that large are not so common, so the software warns that the target level may be low. C weighting compensation would have accounted for some of that, so as you do not need it with your setup turning it off will help. If the warning still appears and you are happy the target level is correct then just ignore the warning and carry on.

Hoping to try REQ finally tonight, just to measure room response to various speakers and different sub positions. I've been reading the manual , two questions that I couldn't find answers for ina thread search here --

1) is it better to use a sweep or pink noise to measure the subwoofer? Which is better for
'main' (small) speakers?

2) manual says RS meter is fine for measuring low freqeuncy response...but what about mid/high frequency measurements? Is the RS meter no good for that?

With REW you can only measure the response with either a sweep or stepped sine waves. Use the sweep.

The RS meter can measure the full range, but its response is nowhere near flat once you get above 1kHz or so. It rises about 5 or 6dB between about 2kHz and 6kHz (not uniformly) then rolls off very rapidly above that. For full range measurement you really need a proper mic, preferably calibrated.

Thanks. I'll settle for low freq for now -- up to what frequency should I consider reliable? 1 kHz? I'm interested too to see what the readings look like with the Pioneer AVR's room compensation and standing wave correction off, versus on.

Hi folks,

Anyone know if the Behringer 2496 PEQ can equalize the two channels seperatly? I want to use one channel for my center channel, and one for the subwoofer. Obviously, I can't EQ them equally as a "pair" but I can't figure out how to actually EQ each channel seperately (it's a digital eq, with no manual EQ sliders).

If you can do it, please let me know how - I have two of these, and if I can't EQ seperately, I'll have to sell this sweet unit!

Best,

Stieger

Yes, 2496 does that well.

(Bruce...)

Ok, please help me understand how I can EQ the Left and the Right channel seperately. I haven't figured out how to do that, and I'd really like to get it completed.

Thx,

Stieger

John - I'm unable to read the settings from my BFD. (I can send filter settings to the BFD with no problem - which is fabulous.)

Under "Equalizer", "Retrieve Channel Filter Settings From Unit" is greyed out. Under "Comms", both Midi input and output have "USB Audio Device" selected.

Am I doing something wrong?

John - I'm unable to read the settings from my BFD. (I can send filter settings to the BFD with no problem - which is fabulous.)

Under "Equalizer", "Retrieve Channel Filter Settings From Unit" is greyed out. Under "Comms", both Midi input and output have "USB Audio Device" selected.

Am I doing something wrong?
Nope, BFD doesn't support interrogation of settings via Midi (or if it does I haven't found out how).

This thread/software has me excited has me really excited about correcting my in-room low frequencies.

Sorry to ask, I've scanned the entire thread, but would someone mind summarizing the prefered hardware?

I plan on getting a ECM8000 mic, what is the better inexpensive preamp?
Which BFD, or is a Ultra curve better?
My Shuttle XPC has AC97 on board sound, do I need a sound card? Which one?

Thank you for your patience and help.

The ECM8000 is fine. There are other manufacturers that build similar mics around the same Panasonic capsule. Any small cheapo mixer with phantom power will work--it doesn't even have to be 48V because the ECM8000 doesn't need it to be that high. I use a Mackie 1202VLZ Pro, but the Behringer UB802 will work and it's much cheaper. Ditto for Phonic, Yamaha, Samson, etc. The Ultracurve DEQ2496 is a nice unit but might be overkill if you just want to EQ your sub. I like mine.

Your sound card might be good enough. You can download RightMark Audio Analyzer (RMAA) and perform a loopback test to see how flat it is (asuming playback performance is flat--a big if). Otherwise the Chaintech AV-710 is only $25 mail order and has bit-perfect digital I/O. Much better: the E-mu 0404 or the M-Audio Audiophile 2496 (or the USB version).

Hmmm, it looks like you can get everything in one trip to Guitar Center, if you get impatient enough. Before you EQ, have you considered bass traps?

Thanks!

This is for my bedroom, no room for traps.

Nope, BFD doesn't support interrogation of settings via Midi (or if it does I haven't found out how).


Oooh ok. That's too bad.. Thanks John.

I recently moved into a new house and I am at a point where I want to start fine tuning my HT.

First off, let me briefly decribe the large, open room that contains my HT. It's dimensions are 40' long x 16' wide w/ 10'-18' vaulted celings. Also, there are no window coverings yet and the floor is polished concrete.

As you can imagine, this room is very lively, to say the least. My first desire is to tame down the echo in the room. I assume that REQW will help with this by identifying the particular frequencies that echo the worst. Is this correct?

Once that is finished, I would be using REQW in conjunction with a PEQ to adjust the response of my sub.

Am I on the right track?? Does anyone have any tips or suggestions for someone getting started? I am a fairly technical person (ME degree), so I have no problem with all of the graphs, numbers, etc. My uncertainty comes from never having done this before. Anyway, any help would be appreciated. Thanks.

I would think you could simply use a mirror to find the first reflection points on the floor and sidewalls and treat those with absorption (at the very least, a rug on the floor)...then move on the dealing witht he low frequencies, via REQ and traps.

Timv,

Congrats on getting to that point in your project!

I'm no expert - but I have done about 30 various theaters for friends, etc., over the last several years.

What I would suggest, and what I believe most of the experts on this forum would tell you - is to treat your room first, then use EQ.

Regardless of how good your EQ settings or filters are, they cannot remove a concrete floor, the 18' ceilings for you. My opinion is Treat first, EQ second.

I believe if you try to EQ the reflections without treating the room, you're putting the cart before the horse and you'll never be able to do it.

I have been using EQs, but until I treated my front wall of my theater (no where near your size) and first 3ft of ceiling and side walls, the EQ alone didn't make the impact I was hoping for.The treatments and EQ made the audible difference.

Best,

Eric

I have the DEQ24/96 with a ECM800 mic. The DEQ supplies phantom power for the mic. Is there anyway possible REW can pull the mic readings over the MIDI conneciton? As I dont have a mic preamp to use for my soundcard.

Also is there any plans to fully support the DEQ24/96?

Thanx.


peace

I have the DEQ24/96 with a ECM800 mic. The DEQ supplies phantom power for the mic. Is there anyway possible REW can pull the mic readings over the MIDI conneciton? As I dont have a mic preamp to use for my soundcard.No, that is not possible

Also is there any plans to fully support the DEQ24/96?Probably, but I'll have to go out and buy one to do that. Some way off.

I'm using Room Equalizer Wizard....

There is a C-weight compensation option in REW that is selected by default.

You can also import a correction table for the RS SPL meter...

Does the "usual" correction table for the RS SPL meter includes this C-Weight and the option must be disable in the software ?

Correction tables would usually include the C weighting compensation, but you can still leave the C weighting compensation turned on in REW as it is only applied outside the range of any correction table values (i.e. below the lowest value and above the highest value).

What a well designed software !

Are there any differences between the analog and digital RS Meter when using it with the REW ?

I see there are plenty of references to using the MobilePre USB sound card with REQW, but what about using the less expensive Transit USB device?

Probably will stick with the RS meter for the mic since there are O.K. cal files already developed.

If the MobilePre is "that" much better, I could go that way; but I don't want to spend more if I don't have to.

I like having the real level controls on the front of the MobilePre rather than using the software controls. Also, the MobilePre has power for a mic but that's not your concern, at this time.

Kal

I think it's probably wise to go with mic power for future considerations.

One issue that's holding me back is something I read about the MobilePre's USB power requirement being relatively high and it not working with a Dell laptop. The fellow was saying that the Dell laptop was a bit low on maximum current from the USB port. This concerns me because I have a Dell laptop. Sure wish I could find that post.

Suppose I'll have to call Dell to find out the USB spec and then M-Audio about the MobilePre's current draw.

Perhaps it's nothing to be concerned about...

It may depend on the Dell. I used the MobilePre briefly with a Dell C800 and it was fine. Now using it with a Fujitsu, also fine. One problem I have heard about (and I do not know if it is true) is that some laptops will not supply any or much power when running on battery.

Kal

I used MobilePre with Dell's tiny/thin 300m laptop and it worked just fine.

Two for two so far; that is some reassurance.

The Dell I have is a D505. I've not yet had an opportunity to ask them about the USB port capabilities, other than I know it is USB 2.0, fwiw.

Man oh man....

Chatted with Dell to see if I could get the USB port current spec; you'd think I had asked Willy Wonka for the Everlasting Gobstopper formula...or had asked NSA about ECHELON!!

Actually, it's worse than that; just the typical million questions but no answer.....

John,

Is there an easy way to 'reset' REQW to its initial state, erasing all the calibration/setup settings etc.? Or do I have to uninstall and reinstall?

Two for two so far; that is some reassurance.


I'm using a MobilePre on an Inspiron 5150 (about 2.5 years old) without any apparent problems. It does seem to suck the battery dry pretty quickly, but maybe it's just all the FFT's that are hammering the processor instead.

To divert for just a minute...has anyone gotten RMAA to play nicely with the MobilePre? Any answers should probably be kept off this thread.

--Greg

Is there an easy way to 'reset' REQW to its initial state, erasing all the calibration/setup settings etc.? Or do I have to uninstall and reinstall?
Just delete this registry key:
HKEY_CURRENT_USER\Software\JavaSoft\Prefs\room eq wizard
then start REW. That deletes all settings. If you just want to clear soundcard or mic cal data can do that from the menus.

Thanks, John.

Those using the M-Audio Mobile Pre, how much are you turning the knobs on the front up?
I just starting messing with it last night and when I tried to calibrate the soundcard, the line wasn't flat at all. Looked more like a 9.4 on the richter scale. I couldn't turn the knobs up much at all before the clip lights came on so they are all the way down. This was with a 1/8" stereo cable being used to loop the input and output.

Just realized I may be plugged into the mic input. I'll have to check when I get home.
On the M-Audio Mobile Pre, what should I be plugged into?
http://www.m-audio.com/images/en/callouts/big/MobilePre_USB_callouts.jpg

I think I've answered my own question. I had bought a couple of the 1/8" to stereo RCA outs and pretty sure I used one from the RS SPL meter to the "mic" input. I should have gotten a 1/4" to RCA adapter and gone from the SPL meter to the 1/4" input on the mobile pre.

Those using the M-Audio Mobile Pre, how much are you turning the knobs on the front up?
I just starting messing with it last night and when I tried to calibrate the soundcard, the line wasn't flat at all. Looked more like a 9.4 on the richter scale. I couldn't turn the knobs up much at all before the clip lights came on so they are all the way down. This was with a 1/8" stereo cable being used to loop the input and output.

Just realized I may be plugged into the mic input. I'll have to check when I get home.
On the M-Audio Mobile Pre, what should I be plugged into?
http://www.m-audio.com/images/en/callouts/big/MobilePre_USB_callouts.jpg

I think I've answered my own question. I had bought a couple of the 1/8" to stereo RCA outs and pretty sure I used one from the RS SPL meter to the "mic" input. I should have gotten a 1/4" to RCA adapter and gone from the SPL meter to the 1/4" input on the mobile pre.


I just use channel 1 line input XLA from the back and turn up the volume about half. I then connect the Stereo line output to my preamp.

Oliver

John,

Any plans yet to add RT60 measurements to REQW?

Thanks
Oliver

What's the procedure for changing mic's? I set it up initially with the RS SPL meter but want to try the Behringer mic. TIA

John,

Any plans yet to add RT60 measurements to REQW?

Thanks
Oliver
It's on the list, but not near the top :)

What's the procedure for changing mic's? I set it up initially with the RS SPL meter but want to try the Behringer mic. TIA
Clear any cal data you have loaded for the RS meter using the entry in the menu, then repeat the input volume setting and spl calibation steps.

It's on the list, but not near the top :)

Just got to wait then :p

Anyway currently to increase the freq that the low freq spectral decay measures? Currently it measures to about 700hz I believe. This should tell us roughly which freq we are having trouble with for RT60 by eyeballing it right?

Thanks
Oliver

Can the gurus provide me some help as I get started to measure and analyze my room.
1. What is the most cost effective sound card for this application? I have seen the MobilePre recommended but at 175 it is out of my reach.
Thx.
Sid

If you are using a RS SPL meter than you won't need phantom power for a mic. Most sound cards (and on-board sound) will work.

BTW in you want a Mobile-Pre, Amazon has the best price I've found.

sidvee99,

I bought the "Creative Labs USB Sound Blaster MP3+ External Sound Card Sound System" from Amazon for around $38 plus shipping. It has excellent connections (RCA, optical, etc.) and works great for me.

I must be doing something wrong. But is there a way to save the "tabs name" into the measurement file (MDAT) ? I usually change the tabs name from left, right, center etc etc to something like With Traps, With ASC, Front little, Back Little etc. But then it then stays as what it currently has been changed to even when you bring up a previously saved measurement file which has different name tabs then. Is this the standard behaviour?

Oliver

That is the current behaviour Oliver, but for the next version I have already changed it so that the tab names are saved in the mdat file and will be restored when that file is subsequently loaded.

That is the current behaviour Oliver, but for the next version I have already changed it so that the tab names are saved in the mdat file and will be restored when that file is subsequently loaded.

Thanks John. Thats fantastic.

Oliver

Just wanted to post my congrats & kudos on a very good piece of software. It does what it does very well, which is one of those important things when it comes to software. Keep up the good work, and thanks for sharing this with us!

Not sure if this has been said or not, a search didn't turn anything up, but to launch the program on a Mac the Terminal isn't needed. Just use the Applet Launcher included with the OS. I kept getting errors when trying with the Terminal, but the Applet Launcher worked fine.

Mac OS X 10.4.5.

Does anyone know if the Edirol UA-1EX is suitable for use with REW? This looks like a newer version of the UA-1x.

I tried it and was getting feedback when setting up the initial level calibration using the RS meter as a mic. If the software is generating the test signals, doesn't this automatically setup an feedback loop?

Thanks.

Does anyone know if the Edirol UA-1EX is suitable for use with REW? This looks like a newer version of the UA-1x.

I tried it and was getting feedback when setting up the initial level calibration using the RS meter as a mic. If the software is generating the test signals, doesn't this automatically setup an feedback loop?
I guess you made sure that Line In was not selected in the output mixer? Even with the correct mixer setup, if it is like the UA-1A there is an internal hardware monitor path from output to input, on the UA-1A it is possible to remove this by removing a pair of surface mount resistors but I don't know if the same is true of the UA-1EX.

I ordered a Denon 2106 which has the built in RoomEQ in it. Does using RoomEQ
in conjuction w/ the built-in add any value? If so, how should I proceed? ie let
Denon 2106 do the grunt work and then tweak?

I just discovered RoomEQ (ETF,...) and want to know if there is any added
advantage before I start reading up a lot on how to use RoomEQ and/or what
aspects I should pay more attention too.

Thanks in advance

EDIT : Do I still need to purchase RS SPL and/or additional MIC to use RoomEQ?
I am not able to find the online HTML tutorial/help files that seems to exist :(

John, Thanks for the response. Your support here has been stellar.

The Edirol has a dip switch on the back that can be used to turn off the monitor thus eliminating the feedback. Now the only problem I have is noise in the output of the device. A bit too noisy for my liking - basically hum and some buzz when the output is connected to my processor and the volume turned up to get the level up to 75dB.

Does anyone else have any noise probelm with any of the other USB sound cards used, such as the Creative mp3+ card? Any advice on getting rid of the noise? Might have to return the Edirol and get something else.

Thanks.

Does anyone else have any noise probelm with any of the other USB sound cards used, such as the Creative mp3+ card? Any advice on getting rid of the noise?
Are you using a desktop PC? Hum/buzz are usually ground loop problems, very unlikely to be the sound card (the UA-1A is very quiet).

Hi John,

I am using a Dell laptop, it was plugged in to the powersupply at the time. I guess I should try it with just the battery - will do that tonight.

Thanks.

Most of the time when I use the laptop I will get a nasty hum unless I unplug the power supply. Its most likely not the sound card since I'm using the cheapest one I could find ($20) with no problem.

-Drew

You are correct. I did not use the power supply and the noise went away. Now I will have to try the calibration.

Will the sound card in my laptop work? It's a Crystal 4281? If it won't work could someone recommend one.

I am not familiar with that one, but if it has Line in/Line out it should work fine. Do a search in this thread for Crystal to see if anyone mentioned it.

I have an Edirol and I am still messing with it to make it work.

Can someone help me with this?

My laptop only has 2 jacks, mic and headphone, can I connect my SPL meter to the mic jack and connect the headphone jack to my receiver so that I can play the test tones from the wizard through my receiver and send the reading from my SPL back to the wizard?

:p :confused:

My laptop only has 2 jacks, mic and headphone, can I connect my SPL meter to the mic jack and connect the headphone jack to my receiver so that I can play the test tones from the wizard through my receiver and send the reading from my SPL back to the wizard?Afraid not. The headphone output is fine for playing the test signals, but the mic input is not suitable, you need a line input. If your laptop has a docking station that may proivde a line in connection, otherwise you will need a USB soundcard such as the Creative MP3+, Creative Soundblaster Live! USB external, M-Audio MobilePre USB, M-Audio Transit or similar.

Thanks for your help John :(

No one ever said this hobby was cheap....now I have MORE things to buy :)

I've been trying to get REW to work but have run into an issue. I installed the software on my Windows PC and am using the RS Analog SPL for a mic. I have the mic connected to the line input and the speaker output connected to the CD input on my HK 340 receiver. I went through the calibration steps without issue.

I selected the sub tab and then ran a sweep from 20-120. When I look at the graph, it shows the opposite of what you would expect. Instead of rolling off at the lower frequencies, it actually rolls up (the curve looks right, just in the wrong direction). Additionally, even when I zoom in, it shows a very smooth curve, which is not what I experience when I manually calculate the FR using test tones and the RS SPL.

I've tried various combinations of the C weighting and calibration offsets specific to the RS SPL but this has had no affect.

Anyone have any suggestions? Thanks!

Sure you have the correct input channel (i.e. left/right) selected in the SPL panel? Haven't inadvertently left a loopback connection in place? If you email me a screenshot of the wizard to show the settings you are using, plus a copy of the soundcard debug file (generated via the soundcard menu) I can investigate further. Could also save the measurement (using Save Measured Data Set), zip up the mdat file and send me that. Likely to be something simple though.

There's another possibility. Some receivers, instead of providing a separate phono input, allow an analog input to be configured as either a regular line-level (as for a CD player) or phonograph input. If this is the case, your CD input may have been inadvertantly set as a phono input. Phono inputs have the RIAA equalization curve, which boosts bass frequencies in the manner you describe. I'm not familiar with your receiver, and it may not do this, but it's worth looking into.

Hi John;
I bought a BFD recently which works fine manually, but I cannot get the REW and the EQ to correctly communicate via MIDI. All the settings are correct and the input light blinks when transferring the filter info, but none of the eq settings are stored in the preset. Filter #11 lights, all the rest are off. I am using the Maudio Midi -USB "UNO" interface. After a while I gave up and entered the filter values manually, which isn't all that difficult. Any idea if I have a defective BFD or could it be the interface?

sportsound,

Difficult to say where the problem might lie. Perhaps someone else has used the M-Audio UNO and could say whether it has worked for them, also worth posting the question on www.hometheatershack.com in the REW forum.

Regards,

Hi, John; is REQ written for a fixed resolution? I have a laptop that displays no higher than 800 X 600. It cuts off part of the display and I can't figure out how to see the rest.

Also I can't see the boxes to select the audio input and output. Any ideas?

Hi, John; is REQ written for a fixed resolution? I have a laptop that displays no higher than 800 X 600. It cuts off part of the display and I can't figure out how to see the rest.

Also I can't see the boxes to select the audio input and output. Any ideas?
It has been written for a min res of 1024x768, but the next version will allow the graph or the controls section to be collapsed, think it still won't fit in 800x600 though.

Hi John;
I bought a BFD recently which works fine manually, but I cannot get the REW and the EQ to correctly communicate via MIDI. All the settings are correct and the input light blinks when transferring the filter info, but none of the eq settings are stored in the preset. Filter #11 lights, all the rest are off. I am using the Maudio Midi -USB "UNO" interface. After a while I gave up and entered the filter values manually, which isn't all that difficult. Any idea if I have a defective BFD or could it be the interface?

I have exactly the same problem with a Dell c800 and Compaq 2500 laptop and the "UNO" USB MIDI. I suspect inadequate USB power. I have a new Intel Mac Mini which I will soon try. I'll post the results in this thread.

John,
I got to playing with RoomEQ again today along with my new DEQ2496.

I found out today that I could rename the speaker tabs, so I could name them something more relevant, since I was only testing differnt sub settings.

Is there any possibility that in the next version that the tab names can get saved in the data file. This would be extremely helpful.

Loving your program.

Is there any possibility that in the next version that the tab names can get saved in the data file. This would be extremely helpful.
Already done in the current dev version, so will be in the next release :)

John, can you explain the problem with adding the RDES unit from AV123 to your software?

Hi,
Is there an easier method other that hiring a Pro. to get a good room reading? I have looked at this thread, but don't think I can manage this myself. I have the fq. readings of my room, but that is all I have been able to do. I have tried bass traps and panels, but I probably have them placed in wrong positions for best results. If you can help, thanks.

John, can you explain the problem with adding the RDES unit from AV123 to your software?Supporting additional equalisers requires:

Details of the filter parameters, their ranges and resolutions
Details of the filter characteristics to accurately model them
Details of the communications protocol to set up the unit
Access to a unit to debug the comms interface and verify the filter modelling
Time to write all the code

Sometimes aspects of this can be fairly straightforward - in the case of SMS-1, for example, an owner had taken the time to measure and post plots of the filter responses, whilst info about the filter parameters was readily available, so adding support for emulating SMS-1 (excluding any unit communications) did not involve a lot of work.

I haven't seen similar info for R-DES, and R-DES already has a nice graphical interface to set it up thanks to a lot of effort the AV123 folk have already put in (and presumably are still putting in). On the comms side R-DES uses a USB interface, which REW does not support - there has been work recently to add support for USB comms in Java (the language in which REW is written) but this is something that would need to be added to REW to have the basic means of communicating with an R-DES.

Even if the folk at AV123 were interested in providing the required information to model the unit and/or support communication with it I think more people benefit from my spending time on improving other aspects of REW, or adding support for the DEQ2496 (for example), than would benefit from trying to support R-DES. I am not ruling it out, just looking to make best use of the time I spend on REW.

This is additional pics. with the fq. reading: Thanks for your help. Joe

Thanx for taking the time to respond to my question about RDES.

John,

Again thanks for your contined efforts.

I've been reading through this thread and went to the Room EQ Wizard home page. I am trying to figure out if this software is right for me. I live in a Condo and right now I have horrible acoustics (hard wood floors, exposed concrete cielings, and floor to ceiling glass walls). I just bought a Rotel RSX-1057 Reciever (no auto-calibration) and a full set of B&W 800 series 5.1 speakers. Needless to say, my speakers are some of the best out there, but i am afraid my condo acoustics will kill me. I dont have a BFD or parametric EQ, so will this software do anything to help me fix my issue? I am planning on getting some acoustic panels, adding more furniture, and whatnot to help my problem. I guess this software will help me figure out my problems, but short of buying an parametric EQ, will it help me fix the acoustics as I add furniture and what not? This software looks awesome, but will it be able to help me select delays times etc. to compensate for my room? The system setup in the Help shows the SPL meter being connected to a BFD in both pictures which leads me to believe I need one?

REW has a good range of measurement features and can be used without any form of equaliser, but you will need an SPL meter or microphone+preamp to make measurements. The impulse response and frequency response plots will show the effect of too little absorption in the room, the low frequency waterfall will show up modal issues. You should also look at ETF5 and RPlusD on www.acoustisoft.com, both have additional acoustic calculation features (such as RT60) which can be helpful in deciding how much damping you might need.

Thanks for your quick response. I have one of the SPL meters featured in the Help. I am hoping my laptop will have the right inputs/outputs. Would using a BFD help with my acoustic issues? If its cheaper to buy a BFD, I'd rather do that instead of buying acoustic panels and whatnot.

You only need a preamp if using a microphone, your SPL meter has a line output which you can use directly.

A BFD can only help with low frequency resonances and perhaps some general shaping of the response, with so much hard, reflective surface area in your room you will need to add absorption. Furnishings can certainly help, including rugs, wall hangings etc but acoustic panels will generally give the best result for a given surface area. Try reading through the info on the websites of acoustic treatment manufacturers, there's plenty of info on Ethan Winer's site www.realtraps.com, for example.

First let me say that as a sometime programmer myself, I think that the wizard is a terrific application. As someone who has written applications I also must admire your cheerful willingness to update the program and to cheerfully deal with all of us who clamor for help or have endless suggestions.

My question- When using the REW, with full range speakers, should it be in the default setting of C compensation? I am using a RS digital SPL meter, #33-2055, and after reading EQ Wiz help file, and various postings in this and other forums, I have to admit that I remain unclear on this.

I posed this question in Ethan Winer's forum and someone logically suggested that I ask you. Hopefully you, or someone who has put it to the test can give a dispositive answer. Thank you.

My question- When using the REW, with full range speakers, should it be in the default setting of C compensation? I am using a RS digital SPL meter, #33-2055, and after reading EQ Wiz help file, and various postings in this and other forums, I have to admit that I remain unclear on this.For low frequencies best is to use a cal file from www.hometheatershack.com. With the cal file loaded you still enable C weighting compensation, but it is only applied outside the range of the data in the cal file.

If you are interested in full range measurements, then for high frequencies you also want C weighting compensation to be on, but (you knew there'd be a but, right? :)) the RS meter doesn't do an especially good job of following the C weighting curve at high frequencies. Enabling the compensation gets the output closer than with it off, but ideally you need a full range cal file for the meter. You can generate something vaguely reasonable by looking at the printed response in the RS meter's manual, or tracking down an accurate response compared to a reference mic if someone has put one online somewhere. Beyond that you really need an accurate mic if you want to make meaningful full range measurements, ECM8000 is about the starting point, or the mic/preamp set IBF Acoustic sell seems to perform well and comes with a cal file, see http://www.ibf-acoustic.com/2005/shop-us/pd1041155380.htm?categoryId=8

And thanks for the compliments :D

OK...let me qualify this by saying I have no clue what I'm doing. Here's what I did:
1) Installed REQW
2) Plugged in my RS Meter as the mic and loaded the appropriate cal file
3) Ran the calibration on my input and output
4) Ran the auto EQ wizard for sub calibration and created several plots

The plot that seemed the most stable is what I'm attaching. By the way, this is a PB10 in a 18x16x7(height) room and the sub is near a corner (tho I also tried spots away from the corner). This room is as of yet untreated other than some curtains.

I also realized after seeing this that the x-over on my receiver was set to 100hz, which somewhat explains the sudden drop (tho its earlier than 100hz). I think its not as obvious at the zoom level the attached plot shows, but the plot seemed messier than I was expecting. The 10db range and many peaks and nulls didn't look too pretty. Am I right?

Here are all of the things I'm not sure about, and could use some guidence. I also plan to do some more reading about this - any suggestions on a good tutorial?
1) What is the meaning of the target response curve on the graph? Shouldn't my target response be a completely flat line???
2) Did I miss anything critical in the setup of EQW?
3) Is the graph meaningful at the zoom level I provided?
4) My PC was making a decent amount of background noise. I'm assuming since it was steady, it shouldn't really affect my relative curve results.

Thanks,
Jeff

Here are all of the things I'm not sure about, and could use some guidence. I also plan to do some more reading about this - any suggestions on a good tutorial?

1) What is the meaning of the target response curve on the graph? Shouldn't my target response be a completely flat line???
2) Did I miss anything critical in the setup of EQW?
3) Is the graph meaningful at the zoom level I provided?
4) My PC was making a decent amount of background noise. I'm assuming since it was steady, it shouldn't really affect my relative curve results.
Your plot is fairly typical, looks like the measurement is working OK.

To get the correct target response you need to tell REW what speaker type it is measuring, use the drop-down box just below the channel tabs to select either "subwooofer" if measuring the sub on its own or "full range" if measuring the sub and the mains together. The Full Range target is a flat line, the sub target has a roll off corresponding to the sub crossover frequency, which you can set next to the speaker type selection.

The PC background noise won't have much influence.

You should try changing the sub phase when measuring sub+mains to see if it improves the response through the crossover region.

For more advice and suggestions try the REW/BFD forum at hometheatershack.

Hi John, I just downloaded REW and was "playing" with it for a while. I am using a M-Audio USB amp/sound card and a Behringer ECM-8000 microphone.

All the places I looked only have setting and help with using the SPL meter. How do I set the Input Volume, Calibrate the SPL, and Set Measurement Level with the mic and amp? Also, REW says use the SPL meter when trying these things.

Any help would be very appreciated.
Thanks, WMD

Hi WMD,

You can find info on using the MobilePre here (http://www.hometheatershack.com/forums/bfd-rew-forum/962-m-audio-basic-setup-question.html).

The setup procedure is the same as when using an RS meter as the mic. You still need your SPL meter for calibrating the REW SPL reading as that provides the reference figure. You won't be able to do a soundcard loopback with that configuration, so just skip that. Otherwise all should work well.

Godd day all

for the past 3 hours i have been trying to get rew to work and configure my soundcard but nothing is working. REW is outputting no sound to the card but it is working cause i can hear windows sounds.

my card outpout is pluggued into my AVR tape in and my line in to the SPL

my sound card is a realtek HD the only option a have to select devices beside REaltek HD output is back line in/mic front, realtek HD rear output, line in front green, line in front pink, microphone

that's it

when i boot the computer i can hear the starting sound in the right front speaker wave in is not mute but as soon as i select the input and output and try to set the measurement level, i get the error that wave control is not found for the selected output and when i check the card properties the wizard has mute the wave , unmuting it does not solve the problem at all as no sound is coming out

it is as if REW is detecting all my input and output but not the card by itself

what am i doing wrong ??

Mon vieux, did you remember to select your card and the input from with the two drop down lists above the "Set Input Volume" button in the lower left part of the EQ wiz window? Click the arrow head button at the right side of each box to open the list.

Mon vieux, did you remember to select your card and the input from with the two drop down lists above the "Set Input Volume" button in the lower left part of the EQ wiz window? Click the arrow head button at the right side of each box to open the list.


i did and the on;y option i have are

REaltek HD output is back line in/mic front, realtek HD rear output, line in front green, line in front pink, microphone

Room EQ mutes the sound card when it loads to prevent feedback, so that is not a problem.

I have a Realteck AC97 chip set on the machine I am using right now, but not I think the same setup you have. When I select it in the Wizard, the input choices listed are mono mix, stereo mix, SPDIF, compact disc, line in, microphone and phone line. But let us ignore the diferences.

Is your SPL meter connected actually to the "line in front green"? If it is, did you try changing the channel selection from L to R (or the reverse) using the radio buttons located just under the large red button to the right of the dbSPL meter?

If that fails, can you get microphone, or headset to use as a mic, or DVD player or anything external to play through your card?

If you did and that did not work, try each of the other input choices in turn, both L and R.

As an experiment I connected a microphone to my PC, added it to the Windows sound control panel and un muted it (Room EQ is running). I selected microphone as the input source in REQ, and clicked on Set Input Volume, but not the run test dialog box and did a voice check. The meter in REQ clearly showed activity (and feedback).

We (meaning us or someone else on this forum who knows what he is doing) will solve this yet.

Tried all that even updating the drivers from Realtek website reverting to old drivers ...just look like the output does not have wave control for some strange reason i ca play sound theu media player itune ect no problem

guess i'll just forget about this program and manually configure the bfd without it

Room EQ mutes the sound card when it loads to prevent feedback, so that is not a problem.

I have a Realteck AC97 chip set on the machine I am using right now, but not I think the same setup you have. When I select it in the Wizard, the input choices listed are mono mix, stereo mix, SPDIF, compact disc, line in, microphone and phone line. But let us ignore the diferences.

Is your SPL meter connected actually to the "line in front green"? If it is, did you try changing the channel selection from L to R (or the reverse) using the radio buttons located just under the large red button to the right of the dbSPL meter?

If that fails, can you get microphone, or headset to use as a mic, or DVD player or anything external to play through your card?

If you did and that did not work, try each of the other input choices in turn, both L and R.

As an experiment I connected a microphone to my PC, added it to the Windows sound control panel and un muted it (Room EQ is running). I selected microphone as the input source in REQ, and clicked on Set Input Volume, but not the run test dialog box and did a voice check. The meter in REQ clearly showed activity (and feedback).

We (meaning us or someone else on this forum who knows what he is doing) will solve this yet.

John, has support for the Ultracurve Pro ever been added to the software? If not, is the communication across the midi channel the same as the BFD?

I am new to the Room Acoustic aspect of Audio. I have read a lot here and and other places and now want to try to improve my room acoustics.

As a starting point, I read up on Speaker placement, did some experimenting, and think I have found a reasonable blend of a setup that allows for a usable family room and listening/viewing room. It is a starting point anyway.

I downloaded and install the REW program to my laptop. I already had a Analog RS meter, and the required cabling. My Laptop/Soundcard is an IBM x31 that has a AC97 card that has Full duplex.

I ran the various calibrations as per the instructions and that seemed to go fine. But when I ran the Auto Measurement test, I got the message that said my system was clipping, and tells you to lower your output. I lower the output 10bd, and reran. I did this until I was at 35db.

Does anyone have any ideas as to what I may be doing wrong?

After the tests, I researched more and I see a note in teh REW instructions that says not to use the Mic Input. On my laptop, I thought that was my only input so I used the mic for the test. Apparently I have a line-out as well. Do you think my problem could have been because I used the mic input?

Thanks,
Darrell

Darrell,

It sounds like you're using your mic input so it's overloading the input. See if you can change it somehow to line. If not, you might consider picking up a cheap USB soundcard.

Ok, I ended having a good sound card, and a bad cable. Everything is working now, and I was able to take some measurements.

Thanks for your help in getting setup with REW. IT looks to be a very handy and well done program.

Regards,
Darrell

John, has support for the Ultracurve Pro ever been added to the software? If not, is the communication across the midi channel the same as the BFD?No to both questions I'm afraid. Life would be a bit simpler if Behringer kept any consistency at all unit to unit in their Midi comms, but seems they prefer not to.

What kind of microphone do I need to to view the frequency response produced by my subwoofer in my room Using Room EQ Wizard?

What kind of microphone do I need to to view the frequency response produced by my subwoofer in my room Using Room EQ Wizard?
An SPL meter with a line output will do the job, e.g. the Radio shack meter. Full details in the help files http://www.hometheatershack.com/roomeq/wizardhelp/help_en-GB/html/index.html

An SPL meter with a line output will do the job, e.g. the Radio shack meter. Full details in the help files http://www.hometheatershack.com/roomeq/wizardhelp/help_en-GB/html/index.htmlI don't need a calibrated microphone or anything like that?

Does this have a line-out port? http://www.radioshack.com/product/index.jsp?parentPage=search&summary=summary&cp=&accessories=accessories&productId=2103667&kw=spl&techSpecs=techSpecs&currentTab=accessories&custRatings=custRatings&features=features&origkw=spl&support=support&tab=summary

What are "A" and "C" weightings?

What is a "BFD"?

I don't need a calibrated microphone or anything like that?No

Does this have a line-out port? http://www.radioshack.com/product/index.jsp?parentPage=search&summary=summary&cp=&accessories=accessories&productId=2103667&kw=spl&techSpecs=techSpecs&currentTab=accessories&custRatings=custRatings&features=features&origkw=spl&support=support&tab=summaryYes

What are "A" and "C" weightings?See http://www.hometheatershack.com/roomeq/wizardhelp/help_en-GB/html/meter.html#top

What is a "BFD"?See http://www.hometheatershack.com/forums/bfdguide/

John I plan on getting the Model 33-2055 Digital RS Meter, but would you recommend not using the internal SoundMax sound card on an IBM T42p Thinkpad or will it be fine for this process?

John I plan on getting the Model 33-2055 Digital RS Meter, but would you recommend not using the internal SoundMax sound card on an IBM T42p Thinkpad or will it be fine for this process?If the Thinkpad has a line input it should be OK, mic input alone isn't suitable.

There is a thread in the Sub forum that talks about the importance of getting thephase at the crossover correct. That it is more important then a flat FR. Does REQW have a phase curve measurement? OR is there a way that it can do this?

Oliver

There is a thread in the Sub forum that talks about the importance of getting the phase at the crossover correct. That it is more important then a flat FR. Does REQW have a phase curve measurement? OR is there a way that it can do this?REW does not have a display of the phase of the response, however the statement about phase is incorrect. The reason getting the phase correct is important is in order to achieve a flat response, if the sub is in anti-phase (for example) with the main speakers at the crossover frequency when measuring at the listening position there will be a deep dip in the FR. Sub phase should be adjusted for flattest FR through the crossover region at the listening position.

REW does not have a display of the phase of the response, however the statement about phase is incorrect. The reason getting the phase correct is important is in order to achieve a flat response, if the sub is in anti-phase (for example) with the main speakers at the crossover frequency when measuring at the listening position there will be a deep dip in the FR. Sub phase should be adjusted for flattest FR through the crossover region at the listening position.


Thats what I thought in the begining. But Sometimes getting the mains and subs totally in phase over a wide range of the FR does result in more dips and peaks but better sounding bass. At least thats what I have found out during both listening and measurement test I have done.

Lets for for instance the mains itself already has a bump every 20hz from the 60 to 120hz. At certain phase setting on the sub, when both are played, you can get the 60-120hz to totally smooth out. But then it sounds lifeless. But at certain phase, the bump is still at the same area. I take it that the 2nd measurement actually shows that the sub and Mains are totally in phase over the region they are overlapping. Wouldnt this be better? Using EQ then to smooth the bump somewhat should result in better sound as well.

Oliver

Not really the same thing. In looking at the sub/mains integration you need to focus on a fairly narrow region around the crossover frequency and avoid a dip there. This gets more complicated if you have a room resonance around that point, but you can see that by measuring sub and mains independently. You can also adjust the sub phase to counter modal resonances where the sub and mains overlap, which can be a much broader range, but you may not then find the best response at the crossover. If the overall impression is a bit lifeless when the response is reasonably flat you might want to look at having a "house curve" for a smooth lift at the lowest frequencies. It is also that case that when modal resonances are suppressed you can miss the extra energy they injected at low frequencies, some time is needed to acclimatise to a more accurate reproduction of what was recorded. In any case, the best setting is the one that gives you the sound you like most :)

Not really the same thing. In looking at the sub/mains integration you need to focus on a fairly narrow region around the crossover frequency and avoid a dip there. This gets more complicated if you have a room resonance around that point, but you can see that by measuring sub and mains independently. You can also adjust the sub phase to counter modal resonances where the sub and mains overlap, which can be a much broader range, but you may not then find the best response at the crossover. If the overall impression is a bit lifeless when the response is reasonably flat you might want to look at having a "house curve" for a smooth lift at the lowest frequencies. It is also that case that when modal resonances are suppressed you can miss the extra energy they injected at low frequencies, some time is needed to acclimatise to a more accurate reproduction of what was recorded. In any case, the best setting is the one that gives you the sound you like most :)

Really love the last sentence :p

My room seems to be hell to tame. Bass is pretty good I should say. Just the mid bass or punch region seems pretty difficult to get right. Explosion like the blast in U571 jsut does not seem to hit you. But WOTW is fantastic from the ground rumble to the laser blast. I also notice that in my room using BFD or SMS-1, the REQW "suggested" EQ jsut does not seem to work right. I have to adjust the levels or the bandwidth to get it right. Last week I just helped a friend with his BFD and when I followed the suggestions and remeasured. I was surprised that the EQed measurement came out exactly as the "corrected" graph on REQW. That has nevered happend in my room. My guess is cancellation and peaks in my room make it more of a trial and error to get it right.

What you are suggesting is that I just concentrate on the area where both the mains and sub crossover meet (within 5-10hz) and ensure that there is no dip at that region? What happens if there is a setting where that happends but it results in a pretty nastic dip or peak further up or down the crossover area?

Oliver

What happens if there is a setting where that happends but it results in a pretty nastic dip or peak further up or down the crossover area?Then you make sure you follow the last sentence :D

Does anyone know if the Griffin iMic (http://www.griffintechnology.com/products/imic/) would be adequate for using REW? I have an Apple PowerBook and its audio IO isn't playing well with REW. The iMic offers line-level input/output. Is that all I need? Of course, I already have an SPL meter.

Does anyone know if the Griffin iMic (http://www.griffintechnology.com/products/imic/) would be adequate for using REW? I have an Apple PowerBook and its audio IO isn't playing well with REW. The iMic offers line-level input/output. Is that all I need? Of course, I already have an SPL meter.Line level I/O is all that's required, but if the on-board audio isn't accessible then its likely the same problem would occur with a USB interface. I guess you have set sample rate to 44.1k already? You may find that the next version of REW (currently going through beta testing) will be easier to interface on the PowerBook, in the meantime you can try posting in the REW forum at Home Theater Shack where there are some other Mac users who may be able to help with what worked for them.

Line level I/O is all that's required, but if the on-board audio isn't accessible then its likely the same problem would occur with a USB interface. I guess you have set sample rate to 44.1k already? You may find that the next version of REW (currently going through beta testing) will be easier to interface on the PowerBook, in the meantime you can try posting in the REW forum at Home Theater Shack where there are some other Mac users who may be able to help with what worked for them.Sorry I wasn't clear enough. REW is able to see the audio interfaces, and I've been able to get test signals to play through my receiver, but when I use my RS SPL meter, I get errors telling me the highest measured volume is way too low (I'm getting in the -60s). For some reason, I thought I'd read somewhere that the build-in audio on a PowerBook isn't adequate for REW.

According to System Profiler, I have line-in but headphone-out.

I use the Griffin iMic with a pc and REW..and it works fine.

Later
RayJr

Sorry I wasn't clear enough. REW is able to see the audio interfaces, and I've been able to get test signals to play through my receiver, but when I use my RS SPL meter, I get errors telling me the highest measured volume is way too low (I'm getting in the -60s). For some reason, I thought I'd read somewhere that the build-in audio on a PowerBook isn't adequate for REW.

According to System Profiler, I have line-in but headphone-out.REW can't access the mixer controls on the powerbook so you'll need to make sure that the line in is selected and the line in volume is set correctly, using the volume adjust tools on the powerbook. Also make sure the channel you've connected to (i.e. left or right) is the same one you have selected in REW on the SPL Meter panel.

REW can't access the mixer controls on the powerbook so you'll need to make sure that the line in is selected and the line in volume is set correctly, using the volume adjust tools on the powerbook. Also make sure the channel you've connected to (i.e. left or right) is the same one you have selected in REW on the SPL Meter panel.Thanks, I'll work with it some more tonight.

Ok, i've got what i'm sure is a newb question.
I've got my computer connected to my subwoofer, and am able to output all kinds of stuff from REQW to it and hear the sound, which is a good thing, heh.

When i connect my RS spel meter using an RCA to the mic jack on the back of my PC, the RS meter goes to the maximum reading, no matter what dial setting i've got it on. What the heck am I doing wrong here? As far as I know, and i've tested it using my headset, i've got it connected to the Mic jack.

So, next step is logically to try hooking up the RS meter to the same jack and try speaking into it and see what i hear coming out of my headphones. I tried this, but my voice sounds very very quiet... in the windows sound level control area, i've got the Mic volume set to max, and the +20db boost is on.

Am i doing something wrong here?

Thanks for any help you can provide.

EDIT: using my headset mic, with the same settings, my voice sounds like a slightly lower than normal level, MUCH louder than through the SPL meter.

oh bah. nevermind. I found the problem... was using the wrong side of the RCA. LOL.
me = audio tech newbie :)

Ok, forget those last two posts entirely.
When i click the Calibrate button just below the big BIG db SPL readout numbers, it creates 100db of noise from my little sub. I cannot match the number on the screen with the number on the RS meter, as the number on the screen goes only to 90db.

It seems to me that there has to be something wrong with this setup =/

Anything glaringly obvious?

OK, so this is totally new to me, I just realized that my system volume was cranked, i can adjust (lower) this to produce less SPL from the Subwoofer.

For some reason, i figured it was supposed to be cranked. blahrg.

OK fellas, here's what i've exported to a text file.

Can i get some specific tech support with regards to Meter / Mic Calibration files?
I applied the one i got from... i'm sorry, i forget where. But here's the contents of the calibration file!
copy / pasted:
58 -0.50
59 -0.51
60 -0.53
61 -0.55
62 -0.57
63 -0.59
64 -0.61
65 -0.63
66 -0.65
67 -0.67
68 -0.68
69 -0.72
70 -0.75
71 -0.77
72 -0.79
73 -0.79
74 -0.80
75 -0.82
76 -0.83
77 -0.84
78 -0.86
79 -0.85
80 -0.86
81 -0.87
82 -0.88
83 -0.91
84 -0.94
85 -0.94
86 -0.93
87 -0.92
88 -0.92
89 -0.91
90 -0.90
91 -0.89
92 -0.88
93 -0.87
94 -0.86
95 -0.85
96 -0.84
97 -0.83
98 -0.83
99 -0.82
100 -0.82
101 -0.81
102 -0.80
103 -0.78
104 -0.76
105 -0.73

Now, here are the results that i exported, which were recorded WITHOUT the calibration file loaded, and were exported WITH the calibration file loaded. Herein lies my question, are the results below taking into account the cal file or not?
before i get flamed, the only reason I exported after applying the calibration file is because i'm literally learning as I go here. As i'm sure you can tell !!

SPL measurement data saved by Room EQ Wizard V3.29
Source: SB Audigy Audio [DF80], MICROPHONE, Left channel, volume: 0.138
Format: Manual Stepped Sine measurement
Dated: Dec 1, 2006 9:09:55 PM
Wizard Settings:
Tracking: On
C-weighting compensation: On
Measurement level: -16.0 dBFS
Target level: 75.0 dB
Note: RMS level at -16dB, master system volume @ max, taken with 33-4050 rs meter, taken without correction file loaded, exported after correction file loaded
Channel: Left, Cutoff 80Hz

30.50 86.20
35.00 86.63
40.00 86.10
45.00 85.70
50.00 85.42
55.00 85.20
60.00 85.03
65.00 84.88
70.00 84.77
75.00 84.53
80.00 84.59
85.00 84.55
90.00 84.48
95.00 84.41
100.00 84.38
105.00 84.33
110.00 84.27
115.00 84.26
120.00 84.22
125.00 84.20
130.00 84.24
135.00 84.22
140.00 84.12
145.00 84.07
150.00 84.16
155.00 84.15
160.00 84.17
165.00 84.17
170.00 84.13


Is it just me, or is the response relatively flat?

OK, one last post relative to my above calibration file questions:

Here's the results from my RIGHT MAIN speaker sweep... the only difference in connections here is that these are taken running through my receiver. I've set the volume on it to my normal LOTR listening level for personal reference.

SPL measurement data saved by Room EQ Wizard V3.29
Source: SB Audigy Audio [DF80], MICROPHONE, Left channel, volume: 0.136
Format: Automatic Stepped Sine measurement
using mic/meter cal file:
C:\Documents and Settings\Mo\Desktop\BFD REW\newrsanalog.cal
Dated: Dec 1, 2006 9:39:47 PM
Wizard Settings:
Tracking: On
C-weighting compensation: On
Measurement level: -16.0 dBFS
Target level: 75.0 dB
Note: RIGHT main speaker recalibrated RS SPL, calibration file loaded?
Channel: Left, Full range

30.00 82.65
35.00 82.88
40.00 82.46
45.00 82.11
50.00 81.92
55.00 81.93
60.00 81.99
65.00 82.10
70.00 82.21
75.00 82.29
80.00 82.34
85.00 82.40
90.00 82.38
95.00 82.33
100.00 82.30
105.00 82.20
110.00 82.17
115.00 82.15
120.00 82.14
125.00 82.10
130.00 82.00
135.00 82.08
140.00 82.07
145.00 82.05
150.00 81.97
155.00 82.02
160.00 82.03
165.00 82.02
170.00 82.01
175.00 81.99


As you can see, it states that it loaded the same cal file as the one in the previous post where I wasnt sure if the results were retroactively tweaked to reflect the calibration file loaded after the measurements were taken but before they were exported.

SO, let me get this straight so i can stop driving myself, my girlfriend, and all of you insane:

The measurements that I'm posting now, they should be accurate, assuming that i've calibrated the RS meter correctly, and all my REQW settings are correct... right?

few!

Basic problem is you can't use the mic input, you need to use a line input. The reason the meter went to max was the mic input includes a power supply output for a microphone. Connect the Rs meter to a line input and start again from there. For more detailed support visit http://www.hometheatershack.com/forums/bfd-rew-forum/

I got it to work using the jack my headset mic is currently plugged into.
I simply switch from one side of the RCA-Y to the other, and had no power output problem with the RS meter going to max.

I'm a little confused as to what the other jacks on the back of my audigy are, to be honest. The one it's currently plugged into has a little picture, etched into the gold plate, of a microphone.
Two have the 'speaker' picture, and say either -2 or -3 below i think.
The last normal jack is a digital only jack.
And there is a Firewire Jack too.

I beleive the jack i'm using is the only one that's a line-in, does that sound right?

EDIT: I've linked a picture of the back of my soundcard...
picture (http://www.imagemonster.net/is.php?i=15007&img=audigy.jpg)

The connector to use on the original Audigy is 2nd from the top, just above the Mic connector, that should be the line input.

by from the top, you mean the second from the left in the picture linked, right?
Does this make a difference in the measurements taken?

Ok, i've got the RS meter plugged into the Line In jack as suggested.
Only one problem... if i hookup my subwoofer to the output jack as is, it's an open connection.
ie; as soon as I un-muted the Line In, i could hear everything through my headphones, coming through the RS meter.
Seems to me I would get a ton of feedback in this case.
What am i missing? i'm sure it's something fundamental and obvious to you ;)

Your soundcard is probably set up with monitoring enabled. You can find a pdf from Creative about setting up the Audigy 1 for measurement on the RightMark site here: http://audio.rightmark.org/downloads/audigy-setup.zip

If your card is not an Audigy 1 check the user manual to see what you need to do to turn monitoring off, it might be in an advanced controls box accessed via a red + next to the line in or analog mix controls.

Thanks for the info.
I dont have the EAX Control Panel - AudioHD apparently.
All that came with the driver download was the Creative Audio Console, which doesnt have step #4 in that PDF file you linked. At least, if it does, i'm not seeing it.

And do you think the EAX Control Panel - AudioHD would be an easy thing to go and download? ha! ya right.

I guess i'll see if I can find the installation CD that came with it.

In regards to your second comment there John, I can go into the recording volume control section, and select Analog Mix, click the advanced controls button, and there is a checkbox there that will 'Record without monitoring' ... perhaps that's all I need to check, and forget the rest of the stuff in the PDF you linked?

Yes, just select "record without monitoring".

That's great to hear... it's that simple, eh?

so I dont have to do all the other stuff?
thank goodness... having nothing but trouble trying to install from the cd i found.

OK. So i've connected the RS SPL meter to the line input, i've told it to record without monitoring.

So i launch REQW, and try the calibration. I first do the set input volume, which works fine, or so it seems.
Next i try the 'calibrate' button, and the readings on the SPL meter are @ around 100db... since the scrollable menu that i'm told to match to what my SPL meter shows only goes to 90db, I'm thinking something is wrong.

Should I simply turn down the master volume control on my computer? wouldn't that be screwing with the calibration?

I mean ... when i turned the system volume down to 1/2, the 'set input volume' wouldn't work. So i turned the system volume up to about 3/4, and it did work. Then i used the calibrate button, and had to set it to 88db.

Something just doesn't seem or feel right here.

Thoughts johnPM?

I've posted over on the BFD REQW forums @ hometheatreshack, hopefully I can get some more specific help there.

Thanks for your help, johnPM.

So if I run into trouble, can i post screen captures here and get some pointers?

It is best to post them on the hometheatershack REW forum as there are more moderators there monitoring and responding to the posts.

Hi John,

After several weeks learning ETF5 and RPlusD (at least their basic operation), have just finished reading this entire thread.

Can't wait to get REQW up and running on my PC, laptop and eventually the Mac that drives my studio. My goal is to test and treat the new basement songwriting/recording room I am just completing construction on.

I do have a Behringer DEQ2496 between my monitor outs and my power amps, and as others have requested, would like to see that functionality added to REQW. However, I can't very well loan you my unit from Toronto, Canada.

So here's my first question:

Since REQW is quickly becoming ubiquitous, and benefits several of Behringer's other EQ-capable products, would it help if several of us contacted Behringer and requested that they loan you a unit (perhaps through a local dealer), so that you have one to work with. Ultracurve Pro DEQ2496 users unite!!!

Appreciate your thoughts, and those of any DEQ2496 users who might want to take up the cause in your locale ...

--Wayne :)

Hi Wayne,

Wait a while before trying REW on your Mac, a new version of REW will be released shortly then I'll spend some time getting it to run properly on the Mac (bought a Mac Mini just to do that).

Not sure you'd get too far with Behringer, but can't hurt to try :) DEQ2496 is on my list to be supported.

Best regards,

John

Hi John,

It'll be a while before I get REQW onto my Mac (Dual 1.4 GHz G4), because I'm still finalizing construction (floors, doors, trim, etc.) on my new studio room, and intend to do the room setup using my Windows PC's.

I will make a concerted effort to get you a DEQ2496 (albeit the effort will be from Toronto, Canada). I think it's ridiculous that you haven't been offered one before now.

I'll begin by contacting Behringer (you never know), but hopefully some of the local frequenters of this forum will chime in with some offers or suggestions. :D

--Wayne

John, I contacted Behringer last night, by means of their international website. Looks like the message was sent to Behringer Americas. So now we just wait.

In the mean time, let's hear from all you DEQ2496 owners out there -- let's help John add capability for this unit to his software ... :)

Cheers,

--Wayne

Room EQ Wizard V4.0 has now been released, available from the download section of the BFD/REW Forum at http://www.hometheatershack.com/forums/

New REW (Room EQ Wizard) user. Had no issues setting everything up, calibrating, measuring, adding CAL file, etc.. running with SB Live 24 USB and RS digital SPL meter.

Regardless of what I attempt to measure, and despite nearly all changes in volume, input/output levels, when I take a measurement, I receive a message in the middle of the measuring process that says "Clipping is detected. Continue or cancel.." I have to turn it down to such a degree that the test tones are inaudible to me to get a reading that doesn't give me a clipping error message.

This happens when testing all speakers, or simply the subs. I find it extraordinarly difficult to believe this is really the case, as all of my speakers are powered by their own 200 wpc monoblock, and my subs are both PB12+/2s running at 1/4 gain with a 75db reference/calibration signal.

Any idea what could be going wrong here?

Thank you.

The clipping detection is on the input signal. With the SB Live and the RS meter on the 80dB range it would be unusual to have the input level high enough to clip, even with the input volumne set to 100%. You can see what the input signal looks like by selecting the scope tab, that shows the raw signal that has been captured. It is shown relative to digital full scale, so anything that reaches 100% (positive or negative) would cause clipping to be detected. It may be that something in your system causes clicks/pops during acquisition (on my desktop PC the wireless card has that effect, I have to disable it off measurement), if so you will see the spikes in the scope plot.

Thank you. Apparently it was caused by using the mic-in instead of the line-in. Changed that, it went away. I have other issues now, but that one is gone. Thanks again.

John, Thanks for the response. Your support here has been stellar.

The Edirol has a dip switch on the back that can be used to turn off the monitor thus eliminating the feedback. Now the only problem I have is noise in the output of the device. A bit too noisy for my liking - basically hum and some buzz when the output is connected to my processor and the volume turned up to get the level up to 75dB.

Does anyone else have any noise probelm with any of the other USB sound cards used, such as the Creative mp3+ card? Any advice on getting rid of the noise? Might have to return the Edirol and get something else.

Thanks.

I installed the last version of REW last Friday, and must say it is a beauty!. Awesome job, John.

Machaibar: have you experienced additional feedback/level problems with your UA-1ex? I switched the dip in the back to AUTO (supposed to be OFF in normal operation) but the feedback is still there. Surprisingly, the Edirol drivers do not provide access to the line-in mixer in Windows, it has to be adjusted on the input volume knob at the device itself. Also, I am pumping a lot of juice from the mic/mixer setup ECM8K and Behringer 1204FX, even before the feedback appears (~26 db on the REW input meter scale). That shouldn't be right, that mic/mixer setup outputs a strong signal and very clean through both the crtl/room and headphones. Yet the noise getting into REW seems high and the overall level input remains low. Push the input a little further and the feedback happens. All other output/freq/cabling setup is exactly as described along this thread, and seems to be working fine. I am close to call it quits with Edirol and try something else, just wanted to see if your UA-1ex has been successful, or if there is something else I am missing here.

Thanks!
Tsound

Edirol UA-1ex update: if you are using this card with REW...
Finally found the problem. The "monitor" function has to be disabled with the bottom dip switch as Machaibar mentioned. Then the card has to be "disconnected", and "reconnected" from/to the computer to reset the drivers. But sometimes it won't work, the drivers will not update (as it was my case). It finally worked by manually uninstalling and re-installing the drivers. Seems to be working fine now, levels are also better. Emailed Edirol support about this problem but no answer yet. :mad:

tsound

Hey guys,

I have finally gotten fed up with "flying blind" with no measurements other than an SPL meter. I have decided to download Room EQ Wizard for my laptop.

However, I have no idea what type/kind of microphone to purchase to take measurements into the subsonic freqs.

Can anyone please suggest a microphone to buy, perhaps from Parts Express or similiar retailer that would be suitable for these type of measurements/callibration that is common place in these forums?

Thanks in advance!
Joe

If you have a Radio Shack SPL it will work fine. It includes a microphone and correction files are readily available. The correction files are necessary because the response is not flat. See here: http://www.hometheatershack.com/forums/bfd-rew-downloads/19-bfd-guide-room-eq-wizard-downloads.html

Can anyone tell me where I could purchase the cable that connects between my laptop and a BFD to transfer my filters from the Room Eq software to the BFD? I am currently reading the help me file on the wizard and it seems that I will be able to transfer my filters info from the software directly to the BFD. Can anyone give me their experience with this? If there is a link or something that talks about it that will do fine as well.

Thank you

Transferring filter settings to the BFD requires a Midi interface, there are quite a few options including the Edirol UM-1X and M-Audio Uno mentioned in the REW help files. Note, however, that recent BFD's have a firmware bug that prevents filter setting over Midi, Behringer have produced a fix but it requires swapping a device in the unit. More info here: http://www.hometheatershack.com/forums/bfd-rew-forum/2769-revised-firmware-v1-4-dsp1124p-v1-3-only.html

If you have a Radio Shack SPL it will work fine. It includes a microphone and correction files are readily available. The correction files are necessary because the response is not flat. See here: http://www.hometheatershack.com/forums/bfd-rew-downloads/19-bfd-guide-room-eq-wizard-downloads.html

Thanks! Now all the references to the RS meters make sense. I will try that! I have another generic brand meter, but I can certainly obtain the RS model easily enough. I thought it would be much more difficult than that. Thanks a million!

Thanks John, I will check out that thread and also the Midi units you mentioned.

Is the Creative SoundBlaster MP3+ USB (http://www.soundblaster.com/products/mp3+/) sound card a good one to use with this program? I wasnt able to find if its full or half duplex. It needs to be full, correct?

Yes, the MP3+ works well.

Is the Creative SoundBlaster MP3+ USB (http://www.soundblaster.com/products/mp3+/) sound card a good one to use with this program? I wasnt able to find if its full or half duplex. It needs to be full, correct?

As John says, it works. I can personally vouch for that 'cause it's the one I use. A tip: you'll have far less setup issues if you *don't* use the Creative drivers. I use the generic USB Audio drivers that come with XP and it's been smooth sailing. In particular, the Creative drivers' installed-by-default "What U Hear" mixer setting gives REW fits. :(

As John says, it works. I can personally vouch for that 'cause it's the one I use. A tip: you'll have far less setup issues if you *don't* use the Creative drivers. I use the generic USB Audio drivers that come with XP and it's been smooth sailing. In particular, the Creative drivers' installed-by-default "What U Hear" mixer setting gives REW fits. :(

Once you connected it to your pc, XP found it and installed the drivers it had?

Yep, it sure did. Actually, I had installed Creative's driver and app package (this was before I started using REW) and was having issues with the bundled apps. I uninstalled the Creative stuff and was going to manually install the drivers only from the Creative CD. I plugged in the card before inserting the CD and was greeted with Windows' "Installing New Hardware" screen. Out of curiosity, I let it install the recommended XP-bundled drivers and have been pleased with them ever since. :D

Thank you Rolls-Royce, you saved me from a headache :)

I believe this card is only half-duplex (at least I havent been able to find anything to say otherwise). Unless I am mistaken, full duplex is nessesary for this software, right? If so, can one use this soundcard as the input and the soundcard already in the laptop as an output? Or does this not need full duplex?

The MP3+ is full duplex (virtually all soundcards are nowadays).

Ah..okay. Thank you John!

I am hoping someone can help me with this. Last night I attempted to use the REM and I was following the help file to make sure I did everything correct. I am using a laptop with the Soundblaster MP3+ external soundcard. I checked the levels and calibrated the soundcard and spl meter as per the help file but my problem is that once I got to the point of actually measuring the frequency response my soundcard was sending out a constant signal that sounded like a heartbeat. It seemed to get louder the longer I kept trying to take my measurement. When I initially checked my levels, I was getting 75db on the meter when my receiver was at -1 but when I went to measure the room, due to the heartbeat sound, I was at 75db on the meter when the receiver was at -16.

Does anyone know what could be going on?