# Dipping my toe in the water: please check my sound card cal



## stevekale (Jan 19, 2013)

Hi there. I'm just starting out with REW and have just attempted to calibrate the sound card in my Mac Pro (mid 2010 running 10.7.5). The sound card calibration produced the following:










while the result of a measurement with the cal file in place is shown below:










I wanted to check the results as, while the variance is for the most part within +/- 0.1 db it would appear to have a lot more variance than the example depicted in the help files. 

Have I made my first tentative steps correctly?

Thanks in advance

Steve


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## fusseli (May 1, 2007)

It's probably fine, onboard soundcards aren't particularly great in general. For example, the heavy LF roll off that you can see in your calibration. The more that needs compensated for with the cal the more noise in the sound device is getting amplified... You'd get better measurement results from an external/better soundcard.


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## stevekale (Jan 19, 2013)

Thanks. One more basic question at this stage... Am I right in understanding that the basic setup for REW is to analyse one channel at a time or do people typically use a splitter and run stereo for their analyses?


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## fusseli (May 1, 2007)

Yeah it is nice to be able to measure L, R, and L+R if that's what you're asking. It shouldn't be hard to come up with a way to send output to one channel at a time.


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## stevekale (Jan 19, 2013)

Thanks. One of the bigger issues I face at the moment is that I don't have a laptop. Lugging a Mac Pro and display around to run tests isn't fun. I presume I will be able to set things up, capture all the measurements, return everything to my desk and then analyse at my leisure.


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## AudiocRaver (Jun 6, 2012)

stevekale said:


> Thanks. One of the bigger issues I face at the moment is that I don't have a laptop. Lugging a Mac Pro and display around to run tests isn't fun. I presume I will be able to set things up, capture all the measurements, return everything to my desk and then analyse at my leisure.


That's the theory, except when you forget a measurement, or decide you want to re-do a measurement just a little bit differently. It IS a pain lugging a desktop computer around.


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## stevekale (Jan 19, 2013)

I can imagine.

There's one other thing I'd like to check at the moment. (FYI I am using a Behringer ECM8000 microphone and an Avid Recording Studio by M-Audio, powered by USB and connected to my Mac via its right line out to my line in at the rear of my Mac.) 

I see I need to use another SPL to calibrate the SPL function in REW. (At the moment the dB levels measured are very low.) The only SPL I have is the Studio Six Digital SPL app on my iPhone. Is this good enough? I also presume that the Mic Gain on the M-Audio Avid needs to always be at the same level. Should I use zero? What about Output control on the Avid - leave at zero always also? (I see that the input level for the Line In (audio line-in port) on the Mac (set in System Prefs-> Sound) also always needs to be at the same level as when SPL was calibrated. When I did the calibration of my sound card, I found that I had to set the input level for the Line In (Audio Line-in Port) to maximum and also raise the Line Out level to reach -12dB.)

Apologies for the string of questions.

Steve


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## JohnM (Apr 11, 2006)

Nothing wrong with using the SPL app. Set your mic gain to whatever is needed to get a decent signal level, if you alter the gain setting just repeat the REW SPL calibration. Doesn't matter too much which output control you use to adjust the output level (REW, Avid, ...), but simpler to stick with one.


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## stevekale (Jan 19, 2013)

I am struggling with Check Levels. I generate a test signal, either sub or main, but even with the mic gain on maximum on the M-Audio and with line input volume set to maximum I can only get the input level in REW to -43dB. Also, the gain control affects the Left channel in REW even though everything is cabled for the right only (and right is set in REW prefs). I must have a bad cable?


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## JohnM (Apr 11, 2006)

-43dB is still useable depending on the level when the test signal is not playing - what are the levels when the input is silent?

What levels are you seeing on the unused channel?

Make sure the 'Thru' box is not ticked for the line in.


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## stevekale (Jan 19, 2013)

Replaced cable. I think the one I was using has the left right mixed between mini-plug and composite ends. Now have right channel responding.

According to my SSD SPL, my room measures approximately 37.4 with nothing playing.

I calibrated REW's SPL with pink noise measuring 75.3. Opening SPL Meter in REW it reads circa 73 with nothing playing and input -40 ish.

When I run Check Levels, I see -12dB on the input meter, -61ish on left and -43/44 ish (it bounces around a lot) for the right.

I am using the sound card calibration i did earlier and I downloaded the calibration file for the ECM8000 from this website.

I don't see a Thru box anywhere. 

For the output device selection I have Default Device (the only other option is Java Sound Device) and for Input Device I have selected Built In Line Input.


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## JohnM (Apr 11, 2006)

From your description it doesn't sound like the input level changes whether you have a test signal playing or not? For setup it is better to set the built-in line input as your default input in Audio/Midi prefs and set REW to 'Default input'.


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## stevekale (Jan 19, 2013)

In System Prefs -> Sound my input is set to Line In and output set to Line Out. If I open Audio Midi Setup I see these as active. (I can not control the master volume from Audio Midi Setup.)

If I set REW's input device to default device everything is the same.

When I Check Levels, adjusting the mic gain alters the right level. When I open SPL Meter, calibrate it, its level does not drop when the pink noise ends - it also doesn't read anything like the SSD SPL (even after calibration).

Thanks for the help


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## stevekale (Jan 19, 2013)

Thru is not checked in Audio Midi Setup


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## stevekale (Jan 19, 2013)

Ok, I don't know why but things seem more normal now. I decided to test the mic by simply plugging the output of the M-Audio pre-amp directly into my Theta. After a little puffing on the mic and switching inputs on the Theta (and then back to the original one) I could hear my voice out the speakers. Reconnected to the computer and things are very different from before. I can achieve -18dB on the right channel without having the input volume set to its maximum. I now have a very high degree of confidence in this mic and pre-amp. Not.


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## stevekale (Jan 19, 2013)

First run of right channel, left channel and then both together:

https://dl.dropbox.com/u/70685392/REW/First_Run.mdat

Now off to figure out what all the measurements mean....!


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## EarlK (Jan 1, 2010)

stevekale said:


> Ok, I don't know why but things seem more normal now. I decided to test the mic by simply plugging the output of the M-Audio pre-amp directly into my Theta. After a little puffing on the mic and switching inputs on the Theta (and then back to the original one) I could hear my voice out the speakers. Reconnected to the computer and things are very different from before. I can achieve -18dB on the right channel without having the input volume set to its maximum. I now have a very high degree of confidence in this mic and pre-amp. Not.


Chalk it up to "user-error" and the "learning experience" ( or the "Way We Learn" ) . :T

*On a different note;*

- Do you realize that you should be able to use your M-Audio soundcard to inject signal directly into the computer & therefore REW, avoiding your current kludge ?? 
- ( ie; this would do away with your current scheme of only using the M-Audio for it's pre-amp section where you then cascade signal into the Mac's soundcard ) 

- So far, I don't see any stated reason to use the Macs soundcard in your setup ( I assume it comes with its own "Core Audio" drivers, yes / no ???? ) .


:sn:


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## stevekale (Jan 19, 2013)

No, I'm not sure what you mean. I've been just following the instructions so far. 

(After "testing" the mic I returned the cabling to the way it was before (i.e. as per the REW help guide). This is why I have no idea why things should now work when they didn't before. Mic to Avid->Right channel to computer audio in + computer audio out to analog in on the Theta.)

The "Avid Recording Studio" comes with Pro Tools SE recording software and presumably any relevant drivers. This is what I have:

http://www.amazon.co.uk/dp/B0041OSWUG/?tag=hydra0b-21&hvadid=9550933629&ref=asc_df_B0041OSWUG


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## JohnM (Apr 11, 2006)

stevekale said:


> First run of right channel, left channel and then both together:
> 
> https://dl.dropbox.com/u/70685392/REW/First_Run.mdat
> 
> Now off to figure out what all the measurements mean....!


Sorry, but those are not valid measurements - big feedback/monitoring problem by the look of them, when set up to measure you should not be able to hear anything from the mic through the speakers.


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## stevekale (Jan 19, 2013)

Just to clarify the point I made above, I could not hear anything through the speakers from the mic when it was set up to measure. I wanted to see if the mic actually worked and so to test that I connected the analog output of the Avid to my Theta. I then returned the cabling to follow the REW online instructions.

i used 1M, 4 sweeps when measuring (although I only ever heard two per measurement).


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## JohnM (Apr 11, 2006)

Use a single sweep and 256k


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## stevekale (Jan 19, 2013)

Ok. Is that because the higher sweeps don't work or to eliminate a variable to see if I have other setup issues? (I am running a 2x2.4 GHz Quad Core Xeon Mac with 24 GB of RAM so I presume computing power isn't the issue.)

I will have to wait until next weekend to redo the measurements. 

BTW I presume the impulse echoes are the biggest giveaway of a problem? I'm going through the pages in the Help again.


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## stevekale (Jan 19, 2013)

EarlK said:


> *On a different note;*
> 
> - Do you realize that you ...


I was thinking about this some more. Isn't this what I am doing? The Avid pre-amp converts the digital input from the Behringer and produces analogue sound which I then feed into the analog inputs on my computer? The analogue out of the computer feeds my Theta a sweep. 

If that's the case I think I have made a major error when doing the sound card calibration. I merely connected the analogue input of the computer to the analogue output. Presumably I need to create the loop in the M-Audio device, although how to do so with its limited inputs I don't know.


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## stevekale (Jan 19, 2013)

JohnM said:


> big feedback/monitoring problem by the look of them


The only thing I think I might have done wrong in the setup is to accidentally press "Direct Monitor" on the M-Audio Avid Recording Studio.


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## JohnM (Apr 11, 2006)

stevekale said:


> Ok. Is that because the higher sweeps don't work or to eliminate a variable to see if I have other setup issues?


Multiple sweeps do not work with some operating system/soundcard combinations - possibly linked to sample rate conversion in the OS. Time alignment is not maintained between the sweeps, so the averaging is no longer coherent. Longer sweeps are more likely to show up soundcard interface issues, but are not really needed unless there is a particular requirement to maximise the impulse response signal-to-noise ratio (which improves by about 3dB for each doubling of sweep length, so a 1M sweep has 6dB better S/N than a 256k sweep).


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## stevekale (Jan 19, 2013)

John, thanks a lot. I will try to find the time to run the measurements again. Any thoughts on the points raised by Earlk and my response?

Regards

Steve

PS: this is quite amazing software you have written!


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## JohnM (Apr 11, 2006)

Thanks 

I think you are on the right track with the setup. As an aside, the latest REW V5.01 beta now supports OS X (at least it runs under 10.5.8, which I have available for test). Might want to give that a try, but I'd make sure you can get things running with the V5.0 release first.


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## EarlK (Jan 1, 2010)

stevekale said:


> I was thinking about this some more. Isn't this what I am doing? The Avid pre-amp converts the digital input from the Behringer and produces analogue sound which I then feed into the analog inputs on my computer? The analogue out of the computer feeds my Theta a sweep.
> 
> If that's the case I think I have made a major error when doing the sound card calibration. I merely connected the analogue input of the computer to the analogue output. Presumably I need to create the loop in the M-Audio device, although how to do so with its limited inputs I don't know.


(i) You are currently emulating the setup where one uses a small analog mixer ( such as the Behringer 502 ) which then sends the audio signal to the computers audio line input for digital conversion . See; 










- You've essentially adapted/combined 2 examples ( into one ) from what is shown in this sites', help section .

(ii) One of the other setups mentioned in the help file uses a soundcard ( such as you have ) that includes the pre-amp & digital conversion , all into one unit . See; 










(iii) That "M-Audio" soundcard can do it all for you ( assuming it doesn't have issues coexisting with the Mac OS ) 

- It'll show up in your Audio/Midi control panel labelled something like this ( this is for a "M-Audio Mobile Pre" card );










- but not this ( this is what the labelling would show for a card that "triggers" the usage of the builtin, generic Mac USB drivers ) ;










(iv) The good news is you know where ( & how to use ) the help section .

:sn:

PS ; if what you're doing currently works for you, then simply shelf this info for awhile / until you fully understand what I just said . :T


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## stevekale (Jan 19, 2013)

Ah I think I understand what you mean. Run the Avid as a USB DAC and connect the analogue out from it directly to the Theta (rather than redo ADC and DAC via the computer)? 

When I plug the Avid in via USB it appears in the Audio Devices window of Audio MIDI Setup as Fast Track (2 in / 2 out). I can then select this for sound input and output. (The sliders are greyed however.) My Audio Devices pane looks a bit different from yours but perhaps that's simply because we are running different versions of OS-X (I am using 10.7.5).











BTW I was looking at the web-based help which does not have the second illustration you have shown unless it is tucked away somewhere that isn't so obvious:

http://www.hometheatershack.com/roomeq/wizardhelpv5/help_en-GB/html/gettingstarted.html#top

I only see the first. The second immediately makes sense. Thanks!

I guess I need to find an XLR to phone cable in order to calibrate for the Avid sound card (in my case as the only inputs are XLR or.....actually the "guitar" input is a large 1/4 in 2 connector phone socket - I may just be able to cobble something together!)


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## stevekale (Jan 19, 2013)

The great thing about this is that it should allow me to use an old Powerbook G4 that I have because I only need the USB connectivity. Hopefully it has enough horsepower to collect the measurements - it only has a 800 MHz Power PC processor and 1GB of SDRAM (how times have changed!). I can do any analysis on my Mac Pro.


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## stevekale (Jan 19, 2013)

Ok I seem to be having trouble getting the M Audio Avid to play with my old laptop - the input level seems to just pulse. I will keep working on it.

In the interim, I cabled the Avid via USB only to my Mac Pro and created a feedback loop to calibrate the Avid sound card (only). See below. Does this look sensible?










https://dl.dropbox.com/u/70685392/REW/soundcardcal_results_new.mdat

Thanks for the help.


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## JohnM (Apr 11, 2006)

Yes, looks fine.


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## stevekale (Jan 19, 2013)

Great - thanks. Next up I will have to lug my Mac Pro downstairs again. Maybe tomorrow evening...

Cheers

Steve


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## stevekale (Jan 19, 2013)

A couple of questions regarding microphone setup. I was reading the Dirac Live website's recommended microphones which says:

"If you already have a microphone pre-amplifier, we recommend an individually calibrated Behringer ECM8000: Information
(Please make sure that the Behringer microphone that you buy is individually calibrated)"

I purchased this microphone but have no idea if it is "individually calibrated". Is there a way to tell? I was planning to use the calibration file from this website with the microphone oriented vertically with a slight title towards the speakers (a guesstimate of 10-20 degrees). 

Should I have the foam cover on the tip of the microphone?

Regards


Steve


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## EarlK (Jan 1, 2010)

- There's no need to use that foam ( wind-sock ) .

- Aim the mic directly at the unit under test / the available ( generic ) calibration files require this orientation .

- *"Dollars to Donuts"*, your mic is not individually calibrated ( or else the seller would have provided a calibration file for it or at least a link for it's download ) . 



:sn:


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## stevekale (Jan 19, 2013)

EarlK said:


> - Aim the mic directly at the unit under test / the available ( generic ) calibration files require this orientation .


Thanks. Regarding the above, I read here that "for listening position measurements we recommend that the meter or mic be oriented vertically with a forward angle of about 10 - 20 degrees to capture a good mix of direct and reflected sound for 'room' measurements" despite the fact that "the following generic meter and microphone calibration files have been created at Cross Spectrum Labs [and] are all on-axis (facing the sound source - horizontal position) response measurement files." 

So I am now confused as to what is most appropriate given I am indeed trying to make listening position measurements to understand my room acoustics ahead of any treatment. If one should have the mic horizontal and pointed at the source, do you aim it straight ahead in the case of stereo runs?

PS: I guess one shouldn't really say microphone or sound card "calibration". Rather, it seems they are merely being "profiled".


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## EarlK (Jan 1, 2010)

- The logic within the info ( from the link you provided ) makes sense . Therefore, follow it's directions rather than mine .

- FYI, I'm usually EQing a speakers response ( in the near field ) rather than looking at the aggregate response within the far-field ( or room ) .

- You'll find advice suggesting both approaches . Play around with the orientations & save the results for each .

- Use the orientation that provides a net result that you like the most ( you'll find one will give a bit more HF over the other ) .

- I really wouldn't over think this aspect ( since most are using the softare to tune subs ) . Also, consider you are using a generic mic calibration file ( inherently less accurate than the mic orientation ) .

:sn:


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## stevekale (Jan 19, 2013)

Yes the thread and particularly the post here are informative. I guess I should have bought a calibrated mic.


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## AudiocRaver (Jun 6, 2012)

All good comments above.

It is an omnidirectional mic. But even omnidirectional mics have some directivity at higher frequencies. But the amount of sensitivity change due to that directivity is minor (maybe 1/2 dB going from on-axis to 90° off-axis???) in the frequency range where you can EQ anyway, below 10 kHz or so. That along with the fact it is not a factory calibrated microphone leads me to suggest that the orientation matters very little. Just be consistent in how you use it. That is my opinion.:bigsmile:


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## stevekale (Jan 19, 2013)

Ok so I gave it another go. I encountered a few weird things in REW. Regardless of whether I selected Right or Left in preferences input channel, only the left channel responded in REW. Even when playing levels out of the right speaker, only the left channel in REW responded. Perhaps this is the default channel when a USB pre-amp is being used? I also couldn't simply switch from left to right or both speakers without quitting REW. Perhaps this is just my cheap pre-amp. 

Anyway...here are my results:

https://dl.dropbox.com/u/70685392/REW/Second_Run.mdat

The file contains 2 runs (256 x 1 pass and 1M x 4 pass) for each of left, right and both channels.

Did I get a valid set of measurements worth studying?

If so, how bad are they? I have a lot of reading/learning to do to understand all of the charts. Some pointers as to what to look for would be appreciated!

Thanks

Steve


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## AudiocRaver (Jun 6, 2012)

Let's start with the good news:

They look like real, meaningful measurements
Your lower midrange, 200 Hz through 1 kHz, looks _great_

Above 1 kHz is also very smooth, except for that steady rolloff, down around 20 DB at the top. Do the speakers have high frequency shelving controls? That is some serious rolloff. Were you straight on axis of the speakers with these measurements? How far away? That rolloff is almost too smooth to be acoustical, it almost looks like an electronic rolloff of some kind. Just a guess.

Below 200 Hz is going to need some work, a combination of speaker placement and working on room modes, then some EQ.

All that said, there is hope!

Edit: Correction, the high-frequency rolloff is 40 DB. Something is definitely wrong there. That is the first thing I would try to figure out. I'm hoping someone comes up with a brilliant suggestion, because I do not have one.


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## stevekale (Jan 19, 2013)

Hi

Thanks. 

I need to look at your comments with the results in front of me but to answer your questions:

1. Shelving controls. No. The speakers are Egglestonworks Andra III
2. How far away? 2.1 metres to listening position +/-
3. Mic orientation was vertical with circa 20 degree forward tilt towards the centre of the two speakers (I did not change the mic position between left and right speaker readings.

I wonder if it is because of the mic calibration file...

Here is a depiction of my room setup (the centre speaker was not on):

http://dl.dropbox.com/u/70685392/Media Room.pdf

Regarding the low end, something that may be relevant (I seem to have a hole at 30Hz when I glanced at the results), the crossover between mains and my Rel Stentor II sub on my Casablanca III HD is set to Phase Perfect with the frequency set to 40 and slope 12. The Andras are supposedly rated to 18Hz (-3dB) but the Stentor seems to add extension. 

Regards

Steve


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## JohnM (Apr 11, 2006)

Are you using the generic ECM8000 cal file? That file is for 0 degrees (i.e. pointed straight at what you are measuring) and may be significantly wrong for your (or any) particular ECM8000, especially at high frequencies. The speaker response also drops sharply at HF left and right of on-axis, so if they are not toed in to point directly at the measurement position that will contribute further roll-off.


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## stevekale (Jan 19, 2013)

Hi. I am using the calibration file downloaded from here

http://www.hometheatershack.com/forums/downloads-area/19-downloads-page.html#axzz2J1vWVD3E

oriented vertically with a 20 degree lean towards the centre of the two speakers, i.e. leaning straight forward, as discussed at the same place above.

Here is a depiction of my room layout (far from ideal): WITH FRONT SPEAKERS MOVED TO EQUILATERAL TRIANGLE CONFIGURATION










I will check the actual toe-in again but the graphic is 30 degrees. I also need to check the recommendation from Eggworks again but I recall 20-30 degrees. I may have opened it up a little to more like 25 degrees. (I am measuring an angle of rotation anticlockwise for the left speaker, clockwise for right speaker, from otherwise facing directly straight ahead.) A 30 degree toe-in does not quite point them directly at the mic which is sitting where the dark red oval is but rather behind this position. (At the moment the room has two Balzac chairs with listening positions where the pale red ovals are but I intend to replace these with a single "love seat" allowing one to sit centrally.


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## AudiocRaver (Jun 6, 2012)

Nice diagram.

With the 25° toe-in as you described it above, the listening position ends up a bit off-axis. No argument, an off-axis listening position can often give better imaging, but at the sacrifice of frequency response. Can you measure the off-axis listening angle? Then, for reference, you might run the frequency response plot exactly on-axis to see how flat that turns out. As John suggested, the off-axis set up could account for some of the high-frequency rolloff above 1 kHz.


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## stevekale (Jan 19, 2013)

Ok I spent a good junk of the afternoon running a number of tests. Firstly I made sure the speaker placement was as per the diagramme above. I then did a bunch of runs for the left and right speakers with the mic in various orientations: upright/70 degrees, horizontal, horizontal pointing at speaker and then horizontal 2 feet from speaker (on-axis). Lastly I did two runs with both speakers. Notes in each measurement run. To my mind the orientation of the mic did not make a massive difference to the general shape of the SPL charts. Still a lot of roll-off.

It's been rather frustrating. I have little faith in my mic - is there somewhere that will calibrate these in the UK (for a reasonable price)? 

I'm also thinking of bringing the speakers closer together. At the moment the listening position is dictated by the 38% rule and the speakers are 125% of the speaker-to-listening-osition length apart (a rule I found on the web somewhere - I will track down and post the reference).

mdat file downloadable from here:

https://dl.dropbox.com/u/70685392/REW/Test runs.mdat

There are lots of things that don't make sense to me in the attached. For example, even the measurements straight in front of each speaker. The right speaker is closer to the sub (crossover settings were posted above) and yet the differences in left/right low-end speaker response are around the other way.

Appreciate any help.

Steve

Re the toe-in being off-axis, the axis point is less than a foot behind the listening position.


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## JohnM (Apr 11, 2006)

Not sure about the UK, but IBF Acoustic in Germany offer a good value calibration service: http://www.acoustics.isemcon.com/shop.htm

When you did your on-axis measurement, was that on-axis with the tweeter? The baffle slopes back a little, so on-axis would be from a little above the tweeter.

I'm in London most weekdays, happy to lend you another ECM8000 if you'd like to try that.


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## AudiocRaver (Jun 6, 2012)

I am beginning to wonder if your microphone is damaged.

Can you describe in more detail the difference between plots three and four?


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## stevekale (Jan 19, 2013)

Hi guys. For the "in front of speaker" measurement I did not raise the mic to tweeter height. It would be a about 6in lower. So plot 3 is at relaxed ear height (my seats lay back somewhat) at the central point between the two seats, horizontal mic position but simply turned clockwise to face the speaker instead of directly forward. In this setup it is still off-axis. I've now brought the speakers closer together and set them back in more of an equilateral triangle configuration but haven't yet taken any new measurements. I will check out IBF. Doing this without a calibrated microphone is poking around in the dark. It was foolish of me to think the manufacturing tolerances were much tighter. John that's a very kind offer. Let me see if it will take long to get a more permanent solution.

Boy, I don't even feel like I have made it to first base. Calibrating my plasma display was much easier!


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## AudiocRaver (Jun 6, 2012)

The microphone definitely looks suspect, perhaps damaged/defective. John to the rescue.

Once you have a microphone you know you can trust, you can compare yours to it and see how far off yours is. If it is off by more than a few DB, I would not trust it even with its own calibration file. The chart from Cross-Spectrum Labs showing the distribution of frequency response curves shows nothing with the amount of high frequency loss you are experiencing. If your microphone's response does not fall into the range they consider to be normal, you might have to call it a loss and get a new one.


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## stevekale (Jan 19, 2013)

Btw I am also losing faith in my M-Audio pre-amp. First, even though I have only coupled the right analogue output to my Theta CB, it doesn't seem to matter whether I choose left or right in REW's preferences. Secondly, the pink noise is now starting to crackle and pop and on occasions cut out altogether. (This is whether the microphone is connected or not.) Lastly, I can get it operational on my Mac Pro but not on my laptop. The laptop sees the device but it is as if the USB bus can't deliver it enough power. Could this be possible? It's a rather old laptop (Apple Powerbook G4 bought in 2002). Lights come on, although if the gain is turned fully up they weaken in intensity, but I get nothing in REW.

Lastly, if I do start again on the equipment side, is it worth getting a better mic than the Behringer? i note that the site John linked to sells other microphones.


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## AudiocRaver (Jun 6, 2012)

You have a lot of variables going here: computer & operating system, preamp, and microphone. The situation is clearly frustrating, but you need to get one setup working properly and then change one variable at a time.

The preamp seems to work on your Mac? Leave it there until you have the microphone issues resolved and can get good measurements that way. Maybe that means moving the Mac temporarily, sorry. Then, with a known good preamp and microphone, you can try moving to the laptop. Reduce and simplify, take it one step at a time, it is the only way to get there. I _know_ it is frustrating.


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## stevekale (Jan 19, 2013)

Good advice. I think the main issue with the laptop (after a bit of research today) may well be because it is only USB 1.1. 

If the pre-amp was the cause of the high frequency roll-off it would have shown up (and been corrected) in the sound card calibration. But the crackling and interruption (on the Mac Pro not the old Mac laptop) to the pink noise is concerning. I may have nuked it trying the laptop.

I think I will start with a call to iSEMcon in the morning. Maybe I should just plonk for the EMM-13D082 and be done with part of the equation....


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## stevekale (Jan 19, 2013)

FYI here was the reference for my point about speaker placement (distance between speakers versus distance to listener)

http://audiophysic.de/aufstellung/regeln_e.html

(See the bottom of the page)


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## stevekale (Jan 19, 2013)

Top quality (over-the-top perhaps) on its way....


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## AudiocRaver (Jun 6, 2012)

stevekale said:


> FYI here was the reference for my point about speaker placement (distance between speakers versus distance to listener)
> 
> http://audiophysic.de/aufstellung/regeln_e.html
> 
> (See the bottom of the page)


At a glance it looks like good advice. Guidelines like these are always a starting point, a certain amount of experimentation will lead to the ideal setup for your room and listening preferences.

I personally lean towards a near–field setup, closer to the listener, wider apart, sometimes more off-axis for better imaging, but you have to decide what is best for your speakers, room, and application. Spend some time experimenting to find the best-sounding setup without EQ, including room treatment as needed to control reflections, room modes, etc., then use EQ for fine-tuning. A setup that has poor sound stage and imaging will not get better with EQ, they have to come first, then EQ may improve them some. That has been my experience.


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## stevekale (Jan 19, 2013)

Agree. The purpose of this exercise is to evaluate my room before purchasing any acoustic treatments. Eventually I will likely purchase the forthcoming Dirac Live module for the CB III HD but there's a lot to learn between now and then. I have a weird room (see graphic - I will post the modified setup later) and there's a limit to what can be done (I rent the property) but I have been looking at RPG BAD panels for the slots behind the front speakers and RPG Skyline for the side and rear walls. Before I starting buying anything, though, I wanted to see just what was going on in the room.


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## AudiocRaver (Jun 6, 2012)

You are wise. Good plan.:T


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## stevekale (Jan 19, 2013)

Ok new mic arrived and I got a Windows laptop from the office. I hate Windows with a vengeance.

I am trying to calibrate the soundcard in REW for Windows. I got several messages saying that the dB varied too much for a valid calibration. I am now getting a message which says the sound card provided no input and then another which says "Timed out waiting for space to write the fade out block to the sound card." 

Ideas? Sound card had it? (This is from a calibration generated via my Mac)










Also, how do I convert the text files that I have received with the mic into a mic calibration file?


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## AudiocRaver (Jun 6, 2012)

Just making sure I have my head on straight: this is a calibration curve for your audio interface, correct? What model again?

Have you checked buffer settings for the audio interface? If they are too short, you can have dropouts and/or noise bursts that would look like this. A setting that is too long can sometimes be bad, too, but too short is more often the problem. A setting of 512 samples is a good starting point.

Audio buffer settings can be finicky. Start with a clean reboot. You might even download the latest driver, disconnect the interface, uninstall the old driver, reboot, Install the new driver, reboot again, then plug in the audio interface and check the buffer settings.

Be sure all audio applications are closed when you change audio buffer settings. Depending on the driver, you may need to close the driver interface for a setting change to be completed. If this all sounds like a pain, it can be. Getting audio buffer and interface settings working right for a given application can be very easy and it can be very difficult. Might as well blame it on sunspots.

It is possible that the interface is bad, but I would work with a clean driver install and buffer settings first.

Edit: I could be wrong, but the calibration files usually come in a standard format that can be imported directly into REW.


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## stevekale (Jan 19, 2013)

Yes. This is directly comparable to the earlier posts here. It is an M Audio Fast Track. The pic I posted above was with REW on my Mac (and can be compared with what I achieved earlier by looking at earlier posts). I could not complete a calibration on Windows which was a fresh driver instal and new installation of REW. 



> Start with a clean reboot. You might even download the latest driver, disconnect the interface, uninstall the old driver, reboot, Install the new driver, reboot again, then plug in the audio interface and check the buffer settings


This is why I hate Windows. Drivers, what are they? Rarely even have to bother with a Mac.

Re buffers, I tried the default 32k, 64k and 128k. The setting in REW only goes to 128k.


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## AudiocRaver (Jun 6, 2012)

I am not a Mac guy, so I cannot speak to that, except that many people say drivers are easier with Macs than with PCs/Windows.

In Windows, the buffer setting in REW is different from the buffer setting for the audio interface driver itself. I have a couple of M-Audio interfaces. Perhaps I should clean up my terminology. In the M-Audio control panel in the notifications area on the lower right-hand corner of the screen, there is a _latency_ setting that ranges from 128 samples to 4096 samples. If this setting is too short, you can get dropouts and noise bursts that might look like what you are getting.


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## stevekale (Jan 19, 2013)

Ah got you. I saw that. It was set at 128. But the chart I showed is what now happens when I run the calibration again on my Mac. It's a mess. I think it's toast. Oh well, £80 down the drain. That will teach me for going cheap. I'm thinking of getting this:

http://www.thomann.de/gb/roland_ua55_quadcapture.htm


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## AudiocRaver (Jun 6, 2012)

There you go. My son-in-law has one for recording and really likes it. It might be overkill for just room measurements, but if you are going to use it for recording, it has some really nice features you will like.


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## stevekale (Jan 19, 2013)

Just for this. Underkill has cost more money thus far.


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## AudiocRaver (Jun 6, 2012)

stevekale said:


> Just for this. Underkill has cost more money thus far.


Understood. It is a top-quality piece of gear. And Roland provides good drivers and solid support.:T


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## fusseli (May 1, 2007)

A soundcard calibration file should not have "blips" like that all over the place, something isn't setup right on the computer side of things. Is the mic volume cranked all the way up or something? If it's a laptop it wouldn't hurt to re-try on battery power.


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## JohnM (Apr 11, 2006)

stevekale said:


>


All those spikes, and the huge peak at 50Hz, are mains hum pickup. What did you use as the loopback cable, TRS to TRS? Also looks to be some odd setup issue giving the roll-off below 1 kHz.

With REW on Windows you have a choice of using the Java drivers or ASIO drivers (if you installed the software for the Quad-Capture it likely did install ASIO drivers). Using the ASIO drivers might get around Windows sound setup issues, otherwise you would need to look at the properties for the input and output being used (right click on the Windows volume icon in the task bar, select Recording Devices to bring up the Windows Sound control panel open on the Recording tab, click on the input you are using and select Properties then check that "Listen to this device" is not checked on the 'Listen' tab and there are no effects active in the 'Advanced' tab). Then check the Playback output selected.

It is unlikely there is anything wrong with the soundcard.


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## AudiocRaver (Jun 6, 2012)

Thanks for the clarification, John.:T


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## EarlK (Jan 1, 2010)

I suspect that the Roland Quad-Capture ( being considered for purchase ) will not work properly with REW (* on your Mac's OS*, due to the cards multi-channel input scheme ) .


:sn:


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## stevekale (Jan 19, 2013)

Thanks John. Let me come back to you on this when I have some time.



EarlK said:


> I suspect that the Roland Quad-Capture ( being considered for purchase ) will not work properly with REW (* on your Mac's OS*, due to the cards multi-channel input scheme ) .
> 
> 
> :sn:


Oh no, you're kidding me. Well hopefully it will work with the Windoze laptop. :gulp:


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## AudiocRaver (Jun 6, 2012)

No idea what the Mac side is like, but I am confident it will work fine on a Windows 7 or 8 machine.:bigsmile:


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## stevekale (Jan 19, 2013)

The setup here was the same I had used previously. The Avid's "guitar" phono socket connected to the RCA out via an adapter - USB to Mac Pro. I could not get a calibration with the Windows machine at all. Anyway, I will start again when the Roland arrives with the EMM-13D - what a cute little mic!


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## stevekale (Jan 19, 2013)

How do I take the text calibration file for my new mic and construct an REW calibration file for it?


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## JohnM (Apr 11, 2006)

REW can probably read it directly, just load the cal file from the mic/meter preferences then take a measurement and view the mic cal file on the plot to check it looks plausible.


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## Kyhl (Dec 15, 2012)

I am having a similar issue with a new CSL calibrated UMM-6.

I have pointed REW to the 0 degree .frd file but my graphs show the calibration file as a flat line.
I've tried manually creating a sample file in word and excel using 100 or so samples from the .frd and saving them as text files with .cal and REW won't read them either.

Is there a special trick?

Also tried renaming the .frd to .mic and .cal and that didn't work either.

One other point. When I open the .frd in notepad the file opens as one long string. Should it be a table? Word and Excel open it as a table but notepad see the calibration file as a long string.


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## JohnM (Apr 11, 2006)

Kyhl said:


> I am having a similar issue with a new CSL calibrated UMM-6.
> 
> I have pointed REW to the 0 degree .frd file but my graphs show the calibration file as a flat line.


Attach the file here and I'll look at it. Bear in mind that loading a cal file via the mic/meter preferences only affects new measurements, for existing measurements you need to use the Change Cal button on the measurement panel.


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## Kyhl (Dec 15, 2012)

It's working today. Don't know what the deal was.

Now onto my next issues. They will probably need their own thread. :wits-end:


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## stevekale (Jan 19, 2013)

Ok I am back with a Windows laptop, Roland Quad Capture and a new mic. I can't for the life of me get the Roland to create a feedback loop in order to measure the sound card.

Here is a pic of the front and back of the Roland:










I simply connected the two RCA jacks together, coaxial out 3/4 to coaxial in 3/4.

In Windoze, with Java selected as the driver (I installed the Quad Capture driver) I have a variety of choices for Output Device, including 1-2 Quad Capture and 3-4 Quad Capture. I selected the second. Below it I can select either Default Output or Line_Out. I tried both. I left sweep level and buffer at their default. Under input device I can select 1-2 Quad Capture, 3-4 Quad Capture or Main Quad Capture. I selected 3-4 Quad Capture. Below it the only option is Default Input (unless Main Quad Capture is selected in which case Line In (Master Volume) is available). Buffer was left at the default. I tried both left and right input channels.

When I go to Calibrate, I see the level in Out (in REW) but nothing in the In columns. I've tried the various Sens and output dials on the front. 

Ideas?


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## JohnM (Apr 11, 2006)

The coaxial in/out are digital channels, use either 1L in/out or 2R in/out depending on which you will use to measure.


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## stevekale (Jan 19, 2013)

JohnM said:


> The coaxial in/out are digital channels, use either 1L in/out or 2R in/out depending on which you will use to measure.


Ok that explains that then. I guess I should have read the manual! Ugh, I am going to have to find a 1/4in phone cable aren't I....


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## stevekale (Jan 19, 2013)

Ok I think I am - hopefully! - back up and running. Soundcard measurement:










Feedback loop measurement with calibration:










Is the "noise" at 8k and 18k anything to worry about?


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## JohnM (Apr 11, 2006)

stevekale said:


> Is the "noise" at 8k and 18k anything to worry about?


Nope - don't forget the highly zoomed in vertical scale, that noise is only about +/-0.02dB.


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## stevekale (Jan 19, 2013)

Ok I am getting increasingly frustrated. Here is what I get when I run a simple measurement of both speakers at once. I always get an error measurement saying the impulse response is not where it should be:










https://dl.dropbox.com/u/70685392/REW/Initial Tests.mdat

Ideas?

Thanks in advance

Steve


(Also, is it possible to use the digital out on the Roland rather than the analogue?)


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## EarlK (Jan 1, 2010)

(a) If you're now trying to use that multi-input Roland card with your 2010 Mac Pro, then your current ( questionable ) results make perfect sense ( ie; that card won't work with REW without a special work-around put into place ) .

Read the top-most sticky ( within this forum) for the work-around .

(b) If you're still using a Windoze machine of some sort, then your sketchy results might suggest it's time for a clean install of the OS ( or more time needs to be spent playing with the buffer settings for the card &/or REW ) . 

:sn:


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## stevekale (Jan 19, 2013)

I am using a Windoze laptop. But I think the problem may be the analog board in my Casablanca. I just tried again with a different analogue input (3 versus 2) and am getting something sensible.

The results were a bit surprising though. I had thought running a Phase Perfect xover to my sub (a Rel Stentor II) at a frequency of 40 and slope of 12 (+ phase) had provided extension to the Andras. REW suggests otherwise.

Here is a chart (1/6 octave smoothing) of the left and right measured separately, once full range and then crossed over to the sub:










Full range would appear to be better, no?

This second chart is puzzling me. It shows the left and right measured separately (full range) and then together) full range. Why would there be the drop off in the low end?










(The 2 together crossed over is even worse.)

https://dl.dropbox.com/u/70685392/REW/Tests with analog3 in.mdat

Any ideas as to how to improve this?

Also I now need to start looking at other characteristics from these measurements. Where best to start?

Regards

Steve


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## EarlK (Jan 1, 2010)

*RE : Your Questions ;*

Sound Waves of the same frequency ( being "waves" ) will add or subtract from each other ( based on their phase characteristics ) resulting in measurably more or less of that same frequency ( just as you are now seeing ). 



> Also I now need to start looking at other characteristics from these measurements. Where best to start?



- Buying a book on acoustics is a good start for one who needs to study the fundamentals . 
- The following may be overkill for those with only a casual interest ( though it is highly recommended ) ; 



:sn:


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## stevekale (Jan 19, 2013)

EarlK said:


> Sound Waves of the same frequency ( being "waves" ) will add or subtract from each other ( based on their phase characteristics ) resulting in measurably more or less of that same frequency ( just as you are now seeing ).


Okay. While I think I understand the concept I still haven't got my head around phase in degrees. If I look at the phase graph for each channel in those lower frequencies it looks broadly the same. I thought they had to be out of phase to cancel and reduce SPL. 










Thanks for the link to the book. I will check it out.


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## stevekale (Jan 19, 2013)

BTW one other thing is bothering me re the setup for measurement. I am using analogue out of the Quad Capture into my Theta Casablanca III HD. I have calibrated/profiled the sound card / Quad Capture. But I run a digital system without analogue input from any device. I feel like I should simply measure the response from digital LPCM input to the Casablanca i.e. not use a calibration file for the Quad Capture and use digital out from the Quad Capture. For outbound signal the Quad becomes a USB to S/PDIF converter but of course retains the ability to receive inbound signal from my mic. 

Any thoughts on this or the above would be greatly appreciated.

Regards

Steve

PS: I am reading the book on my iPad


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## EarlK (Jan 1, 2010)

Correction ( calibration ) files for soundcards are mostly effective at addressing the response deficiencies that are found in their input sections ( most specifically the mic preamp ) .

See  for the differences between response linearities for the line input & the mic input ( this is from an M-Audio soundcard of mine ) .



> Okay. While I think I understand the concept I still haven't got my head around phase in degrees. If I look at the phase graph for each channel in those lower frequencies it looks broadly the same. I thought they had to be out of phase to cancel and reduce SPL.


Phase differences do need to be ( mostly opposite ) for cancellations to occur . 

The apparent paradox ( of your situation ) is most likely explained because your pic ( showing the phases to be almost the same between the 2 speakers ) is likely not believable .

- If you ran REW without a "timing channel" setup & engaged ( created by connecting the unused output to the unused input ) then for sure it's not believable . By default ( when a timing channel is not used ) REW will re-arrange/situate all the captured impulses back to a T=O no matter what the actual time of flight was ( this auto-align to "zero" results in the false relative readings ) .

What is believable are the graphs showing the measurable bass being less ( at the measurement location ) when more sound sources were added ( all, having some overlapping LF content ) .

:sn:


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## stevekale (Jan 19, 2013)

Thanks! I think I understand this. I was wondering what was the purpose of the settings in REW that show on the right of the soundcard preferences pane when using ASIO. Should I set up a timing channel as a matter of course? (Presumably, yes, if the Phase information is to be of any use.)

Then, more importantly, how do I fix this problem? Presumably this is a matter of speaker placement (?) as no manufacturer would want this sort of error to happen in correctly positioned speakers and certainly not when they are sold as "matched pairs". 

(I am also surprised that crossing some of the low frequency to a Rel Stentor II subwoofer makes things worse but that's presumably a whole different permutation.) 

I would also like to understand where to start improving the decay stats but perhaps I should start a new thread as the title of this one is not suitable...

Thanks 

Steve


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## stevekale (Jan 19, 2013)

EarlK said:


> Correction ( calibration ) files for soundcards are mostly effective at addressing the response deficiencies that are found in their input sections ( most specifically the mic preamp ) .
> 
> See  for the differences between response linearities for the line input & the mic input ( this is from an M-Audio soundcard of mine ) .


Ok so if I switch to using the digital out of the Roland Quad Capture then I shouldn't need to recalibrate as I had calibrated the input ADC (the channel used by my mic)?

I am having to switch to using digital input as I have found that, for some unknown reason, if I am using an analogue input on my Casablanca and have the crossover settings set to send LF to the sub, the LF is diverted away from the front L/R but nothing comes out of the sub. Hence the major drop off in SPL I was witnessing.

How do I set up a timing channel so that phase information is accurately recorded? I've not seen a Help page describing the ASIO soundcard preferences page.


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## jtalden (Mar 12, 2009)

No need for loopback:

> Set the physical distances
> Turn on the REW RTA mode on using a Pink PN signal from the signal generator.
> Adjust the SW distance +/- the minimum amount necessity to maximize the SPL output in the XO range.
[You should find that adjustment to be less than ±1.5 m. If you find good SPL support adjusting about the same distance in both directions, favor the pos distance adjustment.]

That method provides the best phase alignment.

This may or may not help the dip at 50 depending on the reason for the dip.


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## stevekale (Jan 19, 2013)

Just so I understand this correctly, basically physically move the speakers while watching the RTA? 

"SW distance" is how far apart the speakers are? I realised the other night that my current setup of an equilateral triangle of speakers 2m apart and 2m to mid listening position means I do have the speakers about a foot and a half closer together than recommended by Eggworks. I was going to widen them out and also reduce the toe-in.

Good idea to do this with the RTA up and running. I can watch it bounce around with all my huffing and puffing trying to move 90kg speakers with spikes ...


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## jtalden (Mar 12, 2009)

If you haven't chosen main speaker and SW locations yet that is the first thing to do.

There are lots of methods suggested to find a good SW location with a little searching. I usually have no more than 3 possible locations in my room so I just move the SW to each and measure it there and chose the one I like best. You could also place the SW at the LP and move the mic to the possible SW locations. No need for RTA with the process. [You can use RTA if you like though.] 

In the above post I was assumed the main speaker and SW locations were already chosen. The distance is the tape measure distance between the speakers and the LP. Those values are loaded into the Pre-Pro. I was answering your question about how to adjust the system to achieve good phase agreement and SPL reinforcement of the SW and the main speakers. The adjustment is made only to the SW distance setting in the Pre-Pro. No need to physically move anything at the point.


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## stevekale (Jan 19, 2013)

jtalden said:


> If you haven't chosen main speaker and SW locations yet that is the first thing to do.
> 
> There are lots of methods suggested to find a good SW location with a little searching. I usually have no more than 3 possible locations in my room so I just move the SW to each and measure it there and chose the one I like best. You could also place the SW at the LP and move the mic to the possible SW locations. No need for RTA with the process. [You can use RTA if you like though.]
> 
> In the above post I was assumed the main speaker and SW locations were already chosen. The distance is the tape measure distance between the speakers and the LP. Those values are loaded into the Pre-Pro. I was answering your question about how to adjust the system to achieve good phase agreement and SPL reinforcement of the SW and the main speakers. The adjustment is made only to the SW distance setting in the Pre-Pro. No need to physically move anything at the point.


The "out of phase" condition I was observing is without a sub-woofer. (I finally clicked as to what you meant by SW.) It was just both front L and R speakers running together. (See the second chart only.) Distance to the listening position from each main speaker is the same. Hence my surprise to observe the difference between L and R independently and L+R together. This was explained as phase cancellation by EarlK but I noted that the phase appeared to be similar. He in return noted that without a timing loop "REW will re-arrange/situate all the captured impulses back to a T=O no matter what the actual time of flight was ( this auto-align to "zero" results in the false relative readings )."


(If I do deploy the sub-woofer for two channel listening I would likely place it centred on the back wall and run the Casablanca DSP as FULL/LP - full range to the front L/R but also sending LF to the sub for 3-point distribution of base around the room. But for now I am trying to figure out what's going on with the front L and R Andra IIIs.)


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## jtalden (Mar 12, 2009)

Okay, I now understand the question and have reviewed your data file. I relabeled the measurements so that the key at the botom of the chart is descriptive. The first 6 measurements are your originals and last 2 are calculated results that I added.

I was able to calculate substantially the same response that you measured. I did this by reversing the polarity of one of the 2 front speakers (also with a very small timing/distance adjustment).
See the resulting SPL below (Trace 7):








This proves that the relationship of the FL and FR speakers when measured together was that one of them was set with opposite polarity vs the other. This is confusing because the original FL and FR measurements (the first 4 measurements) showed they both had the same polarity. This means that the polarity of one of them changed between measuring the FL and FR separately and when measuring them together. I have no clue as to what changed the polarity, but something did. If you find the problem and correct it and then measure them again, you will get the SPL chart as shown below (Trace 8):


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## stevekale (Jan 19, 2013)

Thanks. This is very helpful and I have sent it to Theta who are helping me trouble shoot the issues with my analogue input board on the Casablanca III HD.

Can you clarify a couple of things? When you say "reversing polarity" do you mean inverting the phase? What is the difference between reversing polarity and a small timing and distance adjustment? Lastly, how did you do all this in REW?

I am completely dead in the water on this unless it is possible to get a *stereo* digital out signal from the Roland Quad Capture.

I do not trust my Theta Casablanca analog input card because when I use a crossover setting to direct some LF to my sub, the LF that is indeed diverted from my front L and R speakers does NOT come out my sub! It does with a digital input (e.g. SPDIF over coax), however I can not get a stereo digital signal from the Roland's coaxial digital out with REW. It is either left channel or right channel but not both and I do not have an S/PDIF splitter (does one exist?). 

As a result, I have no idea if the Analog Input board is also causing the phase error....

:crying:
*
Question for JohnM:* when REW generates a sweep or pink noise, is the signal that is sent to the pre-amp stereo or just one channel? I am surprised that I can't get a stereo signal out of the coaxial digital out on the Roland. I only get a left channel when selecting 3-4 and a right channel when selecting 3-4(4) in the output in the ASIO settings in REW.


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## jtalden (Mar 12, 2009)

stevekale said:


> ...
> 
> Can you clarify a couple of things? When you say "reversing polarity" do you mean inverting the phase? What is the difference between reversing polarity and a small timing and distance adjustment? Lastly, how did you do all this in REW?
> 
> ...


A polarity change/inversion/reversal is the same as swapping the wire connections on a speaker. Sometimes there is a switch on a SW to do this. A Pre-Pro does not normally have a control to change polarity of a main speaker or the Sub pre-out.

A Phase adjustment (on a SW for example) does not invert the polarity it just shifts phase near the XO frequencies. Again, a Pre-Pro does not normally have a Phase adjustment.

You can do a loopback measurement on the Roland analog outputs to assure that it is not the problem. The phase trace of both outputs (tested separately) should be flat a 0°. 

If the Roland is good you can do a similar loopback measurement from the Theta pre-outs. The Phase should be 0° for both FR and FL channels (again tested separately when set to stereo mode). If one channel measures 0° and other 180° then it's the Theta that need attention. 


Since the analog XO redirection to the Sub is not working there is a problem with the Theta anyway. It sounds like a repair is in order. You may want to first check that all your cables are working properly and that all settings in the Theta are correct.

To make both channels active for input to the Theta just assign the other channel to the REW "Timing Reference Output". I think you can do the same with the digital output selections.


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## stevekale (Jan 19, 2013)

jtalden said:


> A polarity change/inversion/reversal is the same as swapping the wire connections on a speaker. Sometimes there is a switch on a SW to do this. A Pre-Pro does not normally have a control to change polarity of a main speaker or the Sub pre-out.
> 
> A Phase adjustment (on a SW for example) does not invert the polarity it just shifts phase near the XO frequencies. Again, a Pre-Pro does not normally have a Phase adjustment.


Ok, on both my subwoofer (physical switch) and in the Theta Casablanca (software setup for FL/FR together and for SW separately) I can reverse "Phase" (+/-). I understood polarity to be an electrical construct. 



jtalden said:


> You can do a loopback measurement on the Roland analog outputs to assure that it is not the problem. The phase trace of both outputs (tested separately) should be flat a 0°.
> 
> If the Roland is good you can do a similar loopback measurement from the Theta pre-outs. The Phase should be 0° for both FR and FL channels (again tested separately when set to stereo mode). If one channel measures 0° and other 180° then it's the Theta that need attention.


I want to make sure I do this correctly. I think it can be done for the Roland but I doubt the Casablanca.




jtalden said:


> Since the analog XO redirection to the Sub is not working there is a problem with the Theta anyway. It sounds like a repair is in order. You may want to first check that all your cables are working properly and that all settings in the Theta are correct.


Yes have been in constant touch with them. All cables have been checked. Getting a stereo digital trace will help further isolate things. I don't use the analog input card for anything so if I can get digital in working then a repair can take its time.



jtalden said:


> To make both channels active for input to the Theta just assign the other channel to the REW "Timing Reference Output". I think you can do the same with the digital output selections.


Ok I assume this is simply set in the ASIO panel for soundcard preferences.

I'm still keen to understand how you reversed the 'polarity' for one of the traces.

Thanks for all the help!


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## jtalden (Mar 12, 2009)

stevekale said:


> ...
> 
> I'm still keen to understand how you reversed the 'polarity' for one of the traces.


Select the "Invert Impulse" box as shown below:


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## stevekale (Jan 19, 2013)

jtalden said:


> You can do a loopback measurement on the Roland analog outputs to assure that it is not the problem. The phase trace of both outputs (tested separately) should be flat a 0°.


Ok I did this and for each of the analog input/output pairs the phase was flat (1.5 degrees average with one, 1.7 degrees with the other).



jtalden said:


> To make both channels active for input to the Theta just assign the other channel to the REW "Timing Reference Output". I think you can do the same with the digital output selections.


Excellent! That works. Thank you.


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## stevekale (Jan 19, 2013)

I seem to be narrowing down the issue with the Casablanca analog input. It's fine with other sources but has problems with the Roland Quad Capture and REW. For example, I can plug in my iPod, play a song with high crossover to the sub (160Hz), turn off the main amp and here the sub powering away. If I then connect the Quad Capture into the same input with the same cable and same settings on the Casablanca and "Use Main Speakers to Check/Set Levels" I can only hear something from the sub if I turn all the input volumes to maximum and then only barely. If I select "Use Subwoofer to Check/Set Levels" then I can't hear anything out of the sub at all.

If John or others have any ideas re this I would appreciate it. Theta have all of this and continue to investigate. (I do fear, however, they will now say it's a problem with REW/Quad Capture and drop the support.)

Thanks to jtalden I did get things working via the S/PDIF out on the Quad. I have the known issues in the low end. What's notable with the digital input, though, is the roll off in the high end versus my previous measurements via analog input.










https://dl.dropbox.com/u/70685392/REW/SPDIF tests 7 March.mdat


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## jtalden (Mar 12, 2009)

stevekale said:


> Ok I did this and for each of the analog input/output pairs the phase was flat (1.5 degrees average with one, 1.7 degrees with the other).


Good, the Roland is fine.

If you measure the Theta FL and FR Theta pre-outs let us know:
> Are they the same polarity or reversed from each other? 
> Does this change when using the analog inputs vs the digital inputs?


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## jtalden (Mar 12, 2009)

stevekale said:


> I seem to be narrowing down the issue with the Casablanca analog input. It's fine with other sources but has problems with the Roland Quad Capture and REW. For example, I can plug in my iPod, play a song with high crossover to the sub (160Hz), turn off the main amp and here the sub powering away. If I then connect the Quad Capture into the same input with the same cable and same settings on the Casablanca and "Use Main Speakers to Check/Set Levels" I can only hear something from the sub if I turn all the input volumes to maximum and then only barely. If I select "Use Subwoofer to Check/Set Levels" then I can't hear anything out of the sub at all.
> 
> If John or others have any ideas re this I would appreciate it. Theta have all of this and continue to investigate. (I do fear, however, they will now say it's a problem with REW/Quad Capture and drop the support.)
> 
> Thanks to jtalden I did get things working via the S/PDIF out on the Quad. I have the known issues in the low end. What's notable with the digital input, though, is the roll off in the high end versus my previous measurements via analog input.


If you tested by listening using "Use Main Speakers to Check/Set Levels" there is no low frequency content in that signal. REW is using a higher freq signal in that case so the SW would be silent.

If you want the to measure the SW output measure the FR using stereo mode with the FR main disconnected. 

[Make sure the measurement sweep is set to run from 15 - 20k Hz. The redirected FR signal below the XO setting will be sent to the sub. Make sure you do not have the Theta FR/FL set to "full range". You should have an XO set and the FR/FL set to "small" (or whatever Theta calls it to redirect the low frequencies).]


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## stevekale (Jan 19, 2013)

jtalden said:


> If you tested by listening using "Use Main Speakers to Check/Set Levels" there is no low frequency content in that signal. REW is using a higher freq signal in that case so the SW would be silent.


I have the crossover in the Casablanca set very high (160Hz) to test this. I can flick between FULL and XOVR and note the massive drop in SPL. 



jtalden said:


> If you want the to measure the SW output measure the FR using stereo mode with the FR main disconnected.....The redirected FR signal below the XO setting will be sent to the sub.


No signal gets sent to the sub when the Roland is connected. This is the very problem. Everything I divert via the Casablanca's crossover setting simply disappears. This is not the case with another analog input device e.g. my iPhone over the same cable (using the headphone jack)



jtalden said:


> Make sure you do not have the Theta FR/FL set to "full range". You should have an XO set and the FR/FL set to "small" (or whatever Theta calls it to redirect the low frequencies).]


See below regarding the Casablanca's sophisticated range of available crossovers. I have a lot of control over crossover. The mains can be set to full (crossover settings are ignored) XOVR (in which case they are deployed) or FULL/LP in which case the mains run full range _and_ a LF signal is sent to the sub(s).

I am troubleshooting two problems using analogue input to the Casablanca. One, if XOVR is used to divert signal to the sub it never arrives and the phase issue when running with FULL. It's a very weird problem. 



> Each input has these industry-proven crossover options:
> 
> Phase Perfect
> Butterworth
> ...


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## stevekale (Jan 19, 2013)

jtalden said:


> If you measure the Theta FL and FR Theta pre-outs let us know:
> > Are they the same polarity or reversed from each other?
> > Does this change when using the analog inputs vs the digital inputs?


Presumably this would require a loop using analogue out of the Theta into the Roland. I will have to take a look at this in a week or so as tomorrow all my gear gets packed up while the room it's all in is redecorated.


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## jtalden (Mar 12, 2009)

Another thought:

My pre-pro, as most all others, does not apply any of SW management to the multichannel inputs.

Are you connecting the Roland analog outputs to a Theta Stereo input or to the Theata multicahnnel inputs?


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## stevekale (Jan 19, 2013)

Into the 2-channel analog input. The crossover controls in the Theta Casablanca apply to any of the many digital and analogue inputs and regardless of mode. (Basically the first thing you do is configure your speakers - levels, crossovers, number/config - for each 'source'. Any source can draw from up to three of the connections in the back (and can easily switch between these). So I can configure a 2 channel "CD" input and a "Blu Ray Movie" input that draw on the same input device through either the same connection, e.g HDMI, or a different connection for each, e.g. coax for "CD" and HDMI for "Movie". I can then configure the speakers completely differently for "CD" versus "Blu Ray Movie" e.g. no surround for CD and no sub versus 5.1/7.1 for the "Blu Ray Movie" source. If my primary input for CD is coax and the second input is HDMI I can readily flick between the two to test the differences between the two input paths etc. Of course, you also assign modes to each source, e.g. DTS or Matrix or a whole host of others, and can easily switch between each.) 

If I plug my iPhone into the same analogue input, with the same cable (but without the 1/4in phone adapter) it all works fine. Theta think it may have to do with the voltage the Roland operates at. (I have no idea why.)

I think my digital connection (S/PDIF) measurements are valid - hopefully. Do you notice any phase issues etc? 

(BTW if I do want to use the timing loop to correctly record phase, how do I do this? I selected the other digital channel in soundcard preferences as a "timing reference output" to get stereo on the digital out (as you suggested). It's not clear to me how I would create the loop back when using digital out unless I use an analog channel on the Roland. Is it even worth bothering with?)

In the hope that my digital measurements at least are valid, how should I start thinking about dealing with the 50Hz dip? I was planning on placing some RPG BAD panels in the front corners of the room (and RPG Skylines along the rear wall and side wall reflection points) but before I start putting down cash I'd like to have a strategy for what's likely needed.


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## jtalden (Mar 12, 2009)

stevekale said:


> Into the 2-channel analog input. The crossover controls in the Theta Casablanca apply to any of the many digital and analogue inputs and regardless of mode. (Basically the first thing you do is configure your speakers - levels, crossovers, number/config - for each 'source'. Any source can draw from up to three of the connections in the back (and can easily switch between these). So I can configure a 2 channel "CD" input and a "Blu Ray Movie" input that draw on the same input device through either the same connection, e.g HDMI, or a different connection for each, e.g. coax for "CD" and HDMI for "Movie". I can then configure the speakers completely differently for "CD" versus "Blu Ray Movie" e.g. no surround for CD and no sub versus 5.1/7.1 for the "Blu Ray Movie" source. If my primary input for CD is coax and the second input is HDMI I can readily flick between the two to test the differences between the two input paths etc. Of course, you also assign modes to each source, e.g. DTS or Matrix or a whole host of others, and can easily switch between each.)


Okay.



> If I plug my iPhone into the same analogue input, with the same cable (but without the 1/4in phone adapter) it all works fine. Theta think it may have to do with the voltage the Roland operates at. (I have no idea why.)


The adaptor needs to be a 1/4" *TS* to RCA (mono Adaptor). If it is a 1/4" *TRS* to RCA then it won't work. I can't relate to a "Voltage" issue as all audio interfaces and all pre-pros comply with standards that assure compatibility. I still feel there is a simple setting or wiring issue.



> I think my digital connection (S/PDIF) measurements are valid - hopefully. Do you notice any phase issues etc?


Looks good - the HF SPL drop-off is just because the FR and FL are slightly different distances from the mic.



> (BTW if I do want to use the timing loop to correctly record phase, how do I do this? I selected the other digital channel in soundcard preferences as a "timing reference output" to get stereo on the digital out (as you suggested). It's not clear to me how I would create the loop back when using digital out unless I use an analog channel on the Roland. Is it even worth bothering with?)


You should focus on the setup and calibration problems you have. Phase analysis is for measurement/calibration hobbyists. It is not needed for making standard system settings. It is best practice to only measure one channel at time that is why REW is setup the way it is. Running 2 or more main channels together causes the signal interference and misleading charts.



> In the hope that my digital measurements at least are valid, how should I start thinking about dealing with the 50Hz dip? I was planning on placing some RPG BAD panels in the front corners of the room (and RPG Skylines along the rear wall and side wall reflection points) but before I start putting down cash I'd like to have a strategy for what's likely needed.


Check with those in the Acoustics Forum. My approach would be to implement EQ, or get 1 or 2 SWs and locate them the to reduce the dip and then apply EQ.


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## stevekale (Jan 19, 2013)

jtalden said:


> The adaptor needs to be a 1/4" *TS* to RCA (mono Adaptor). If it is a 1/4" *TRS* to RCA then it won't work. I can't relate to a "Voltage" issue as all audio interfaces and all pre-pros comply with standards that assure compatibility. I still feel there is a simple setting or wiring issue.


This may be where I am coming unstuck. I am using a simple Monster cable with a 3.5mm stereo headphone jack at one end and L/R RCA jacks at the other. The adapter is a simple stereo adapter:

http://www.amazon.co.uk/gp/product/B000LATMQW/ref=oh_details_o02_s00_i00?ie=UTF8&psc=1

Theta were thinking it's because the analog out of the Quad is balanced while the RCA in on the Theta are not. 




jtalden said:


> Looks good - the HF SPL drop-off is just because the FR and FL are slightly different distances from the mic.
> 
> 
> 
> You should focus on the setup and calibration problems you have. Phase analysis is for measurement/calibration hobbyists. It is not needed for making standard system settings. It is best practice to only measure one channel at time that is why REW is setup the way it is. Running 2 or more main channels together causes the signal interference and misleading charts.


Okay



jtalden said:


> Check with those in the Acoustics Forum. My approach would be to implement EQ, or get 1 or 2 SWs and locate them the to reduce the dip and then apply EQ.


Theta will be implementing Dirac Live later this year. I wanted to deal with any core problems with acoustic treatment first then leave the balance for Dirac to mop up. That said, I am also building an audio server (like the Bryston BDP-2 but better) which will run JRiver. It would appear that I could feed it filters from REW analysis. That may be a fun interim solution. 

I was chatting with a retailer of acoustic products here in the UK. He thinks it's not the dip that the problem but rather room modes boosting frequencies around it. The room is 4.9m x 4.1m x 2.3m. His advice was to add a couple of RPG Modex Plates.

I appreciate all the help.

Steve


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## stevekale (Jan 19, 2013)

Is this the sort of adapter I should use (with a splitter on the other end of the RCA jack-terminated cable for stereo runs)?

http://www.amazon.co.uk/Phono-6-35m...computers&ie=UTF8&qid=1362779025&sr=1-1-spell

Oh dear, I am beginning to feel incredibly stupid and embarrassed...


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## jtalden (Mar 12, 2009)

stevekale said:


> Is this the sort of adapter I should use (with a splitter on the other end of the RCA jack-terminated cable for stereo runs)?
> 
> http://www.amazon.co.uk/Phono-6-35m...computers&ie=UTF8&qid=1362779025&sr=1-1-spell
> 
> Oh dear, I am beginning to feel incredibly stupid and embarrassed...


Yes that is the correct adaptor.

The one you linked above is shorting out the signal from the Roland and yes, it is because the Roland has a balanced output.

Many of us had some initial connection issues when setting up the measuring system for the first time. Welcome to the club.


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