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Finished treating room - graphs good enough?
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fetachin
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4th December 2012
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Finished treating room - graphs good enough?

Finished treating room. Good enough for a small home studio?

I've added all the bass traps and absorption panels that I can afford and realistically fit into my studio. I am growing weary of tweaking the speaker and subwoofer placement and their dials and obsessing over the graphs and wondering what else I can do.

Basically, I want to know if I can move on to actually making some music. Or are there some glaring problems I still need to fix?

Any comments much appreciated!


Room size: 10.5' x 11.5'
(There are a couple of graphs of the untreated room at the bottom of this post.)


Frequency left speaker and subwoofer (1/24 octave smoothing):
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Frequency right speaker and subwoofer (1/24 octave smoothing):
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Frequency left and right speaker and subwoofer (no smoothing):
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ETC left:
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ETC right:
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Waterfall left and right speaker and subwoofer:
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Reverberation left and right speaker and subwoofer:
Name:  reverb-left-right.png
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And this is a bit of what the room looked like before any treatment:

Untreated frequency left speaker and subwooer (1/48 octave smoothing):
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Untreated waterfall:
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Untreated reverb:
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Can you post the IR files?

If not:
Before posting your measurement results


From what can be seen; it looks good (especially considering the room size) but perhaps a little dry (short decay) in the highs compared to the lows but I´ll make a reservation until I´ve seen the IRs.
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Quote:
Originally Posted by fetachin View Post
Thanks Jens. Here you go...

Views: Size: ">Attachment-IRJDSUNE9932123321222xxeww-319771
Views: Size: ">Attachment-IRJDSUNE9932123321222xxeww-319772
Views: Size: ">Attachment-IRJDSUNE9932123321222xxeww-319773
I´ve downloaded your files and even if the audio files are 24 seconds long (too long), the actual data recorded is only about 250 ms (file “ir-left-sub”) and thus we cannot use it to evaluate decays longer than this:
Finished treating room - graphs good enough?-ir-left-sub-ir-cut-off.gif Name: ir-left-sub; IR is cut off.gif Views: 314 Size: 88.5 KB ID: 319813" style="margin: 2px" />

Please check you settings and record them again using long (6-12 seconds or 256-512k FFT block size is good) sweeps and also use multiple takes (averages) to further increase the S/N-ratio.

Also check your gain structure since the peak level in your current recordings is at about -15 dBFS.
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Jens,

I'm fairly new to all this, so please bear with me... but I guess I'm having trouble producing IR files all that different from the ones I already posted.

I'm using FuzzMeasure.
I did do 10s sweeps.
I did do multiple (3) synchronous averages.

So I'm not sure why the files are 24 seconds long with only 250ms of data.

As far as gain structure, I set my speaker volume to be fairly loud (subjective, I know) and then increased the mic input level until FuzzMeasure's meter clipped and went red, then dialed it back a bit, peaking near 0db on the graphs above. Not sure how this relates to -15dBFS and what I should be doing differently, or how to measure dBFS.

Again, somewhat new to all this and very much appreciate the patience and hand holding.
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Quote:
Originally Posted by fetachin View Post
Jens,

I'm fairly new to all this, so please bear with me... but I guess I'm having trouble producing IR files all that different from the ones I already posted.

I'm using FuzzMeasure.
I did do 10s sweeps.
I did do multiple (3) synchronous averages.

So I'm not sure why the files are 24 seconds long with only 250ms of data.

As far as gain structure, I set my speaker volume to be fairly loud (subjective, I know) and then increased the mic input level until FuzzMeasure's meter clipped and went red, then dialed it back a bit, peaking near 0db on the graphs above. Not sure how this relates to -15dBFS and what I should be doing differently, or how to measure dBFS.

Again, somewhat new to all this and very much appreciate the patience and hand holding.
I don´t use Fuzz, but the general problem seems to be that the rec level is too low (in conjunction with 16 bit resolution). Can you please do a recording and go approx. 10-12 dB past the clip level reported by Fuzz Measure, so we can check if it´s a bug of some kind since this seems to be a general problem for FM users:

Did my first measurement! Need help to analyse it.

My First Measurement. Am I doing it right?
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This would actually explain why a lot of plots from FM looks the way they do. If low rec level and 16 bit resolution, the low-level information is lost due to truncation and the decay rate on the waterfall seems to increase after about 300-400 ms or so when in fact it´s due to the lack of information after this time frame.

It would be nice if an experienced Fuzz user could do some tests.
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fetachin, you should try your recordings again without the synchronous averaging feature. It was added for situations where you couldn't use a longer sweep signal and need to improve the SNR using alternate means.

To clear up the confusion, FuzzMeasure is indeed working with floating-point data (i.e. at 32bit resolution) internally. When fetachin exported his impulse, he likely stuck with the default value of 16-bits. Impulses can be exported at 24- and even 32-bit depths.

It's possible that the synchronous averaging is to blame for the lower level here because of the nature of my implementation, but it's hard to say here. I'd say to avoid it in favour of longer sweep durations if you're having trouble with SNR.

Jens: If you'd like to suggest some tests, please fire me an email at chris@supermegaultragroovy.com and I'd be glad to run them for you.
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Quote:
Originally Posted by liscio View Post
fetachin, you should try your recordings again without the synchronous averaging feature. It was added for situations where you couldn't use a longer sweep signal and need to improve the SNR using alternate means.

To clear up the confusion, FuzzMeasure is indeed working with floating-point data (i.e. at 32bit resolution) internally. When fetachin exported his impulse, he likely stuck with the default value of 16-bits. Impulses can be exported at 24- and even 32-bit depths.

It's possible that the synchronous averaging is to blame for the lower level here because of the nature of my implementation, but it's hard to say here. I'd say to avoid it in favour of longer sweep durations if you're having trouble with SNR.

Jens: If you'd like to suggest some tests, please fire me an email at chris@supermegaultragroovy.com and I'd be glad to run them for you.
Hi Chris and thanks for answering.

Since this seem to be a general problem for many users, I think there´s something that needs to be fixed. I´m happy to help finding the cause but since I´m not on a Mac, I cannot do much to help I guess, but the first thing to try, is to record above clipping (as indicated by Fuzz) to exclude a possible error with the input level indicator. Next would be (assuming the level indicator is correct) to check the gain structure of the internal processing and also the export.

Sincerely Jens Eklund
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Quote:
Originally Posted by Jens Eklund View Post
Hi Chris and thanks for answering.

Since this seem to be a general problem for many users, I think there´s something that needs to be fixed. I´m happy to help finding the cause but since I´m not on a Mac, I cannot do much to help I guess, but the first thing to try, is to record above clipping (as indicated by Fuzz) to exclude a possible error with the input level indicator. Next would be (assuming the level indicator is correct) to check the gain structure of the internal processing and also the export.
The internal processing and export are fine. When I use Soundflower on my Mac to create a digital loopback and record that, I get a perfectly flat frequency response with a perfect impulse having a magnitude of 1.0. The exported impulse response—even with 16-bit precision—is also reflecting this impulse perfectly.

As for clipping, this same digital loopback test only clips when I leave the amplitude of the sweep set to 0dB. This makes perfect sense because audio values of 1.0 are ambiguous: they're either clipped or full scale, and we err on the side of clipped.

Setting my sweep amplitude at -1dB eliminates the clipping, and still results in a perfect impulse response. Hell, even setting the sweep as quiet as -50dB will result in the same perfect impulse, because the swept sine method is measuring the correlation between input and output.

So, as far as I'm concerned, fetachin did everything pretty much right in terms of the measurement process, save for his use of synchronous averages (my fault, perhaps) and exporting at 16-bits (also my fault, for leaving that as the default.)
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Quote:
Originally Posted by liscio View Post
The internal processing and export are fine. When I use Soundflower on my Mac to create a digital loopback and record that, I get a perfectly flat frequency response with a perfect impulse having a magnitude of 1.0. The exported impulse response—even with 16-bit precision—is also reflecting this impulse perfectly.

As for clipping, this same digital loopback test only clips when I leave the amplitude of the sweep set to 0dB. This makes perfect sense because audio values of 1.0 are ambiguous: they're either clipped or full scale, and we err on the side of clipped.

Setting my sweep amplitude at -1dB eliminates the clipping, and still results in a perfect impulse response. Hell, even setting the sweep as quiet as -50dB will result in the same perfect impulse, because the swept sine method is measuring the correlation between input and output.

So, as far as I'm concerned, fetachin did everything pretty much right in terms of the measurement process, save for his use of synchronous averages (my fault, perhaps) and exporting at 16-bits (also my fault, for leaving that as the default.)
You´re probably right but I don´t think this user is using averages:
Did my first measurement! Need help to analyse it.

And his levels are also very low causing truncating of low level data when exporting in 16 bit.

I guess the first thing would be to advise users to always record and export in 24 bit for starters. If you find out that the averaging is the cause of the low level, I´ll stop recommending it until the issue is addressed.


Thanks again for listening.
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Quote:
Originally Posted by Jens Eklund View Post
You´re probably right but I don´t think this user is using averages:
Did my first measurement! Need help to analyse it.
I think you're right. Still, I advise against Synchronous Averaging when I can.

Quote:
And his levels are also very low causing truncating of low level data when exporting in 16 bit.

I guess the first thing would be to advise users to always record and export in 24 bit for starters. If you find out that the averaging is the cause of the low level, I´ll stop recommending it until the issue is addressed.


Thanks again for listening.
Recording isn't always possible in 24-bit, but that's not usually a concern. The levels are fine as long as you're 'tickling' the clip range. It's the export that we need to educate users about.

Anyway, I have some backlog to get through with my other products, but FuzzMeasure is up for some more TLC really soon. I've got about a year's worth of improvements all bunched up that I need to clean up and get out into the world…
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Seems to be some kind of noise going on. Frequency response is not all that bad.
How are things translating when you mix or listen to music in the room?
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Finished treating room - graphs good enough?-noice.jpg  
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Haven't started mixing yet, but I will say that as far as listening to music, reference tracks, etc., there has been a world of improvement from the untreated room. The low end in particular is much tighter and well-articulated. Also no ringing when I clap my hands. In general the listening experience is getting closer to what I hear with a nice pair of headphones.
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In the end that is the test. Room treatment is tool to let you work in the room faster and more accurate. If that is what you have going on then you did the right thing.
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Quote:
Originally Posted by Glenn Kuras View Post
In the end that is the test. Room treatment is tool to let you work in the room faster and more accurate. If that is what you have going on then you did the right thing.
Thanks Glenn. The GIK bass traps really made a noticeable difference. I'm afraid to ask, but how worried should I be about the noise you mentioned in your previous post?
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What is the reading on the level meter before taking your measurements, i.e., baseline room noise level? Do you have a heater or AC running or something?
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Quote:
Originally Posted by fetachin View Post
Here are my IRs again: 10s sweep, exported at 24-bit, and with no synchronous averaging.

Attachment 320008
Attachment 320009
Attachment 320010
So;


First of all; the rec level is still low (-12,7 & -13,0 dBFS) but since it´s now a 24 bit file; loss of low-level data is no longer a problem, but I´m still curious what the rec level meter in Fuzz indicated when recording? The noise floor could be improved by using averaging, but as of today, I won´t recommend it to Fuzz users since Chris (Fuzz) thinks there might be something wrong with that in FM. Stay tuned.


Finished treating room - graphs good enough?-ir2-left-sub.gifFinished treating room - graphs good enough?-ir2-right-sub.gif
The waterfall plots show a quite rapid decay except for a few modal frequencies: (25, L), 29, 41 & 50/54 (R/L) Hz primarily. The 120 Hz trail is noise, presumably the first harmonic of the mains if you´re in a 60 Hz AC county?


Finished treating room - graphs good enough?-etc-ir2-left-sub.gifFinished treating room - graphs good enough?-etc-ir2-right-sub.gif
The ETC reviles a strong early reflection at about 1,4 ms (after direct) and it´s a lot stronger in the left channel. There’s also some very early reflections (possibly caused by diffraction from the speaker itself or nearby obstacles or screens) within the first ½ ms or so. Except for that, the ETC is quite clean and indicates a close to anechoic speaker-room response (but this is not necessarily a bad thing, it depends on the design concept desired).
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Originally Posted by fetachin View Post
Thanks Glenn. The GIK bass traps really made a noticeable difference. I'm afraid to ask, but how worried should I be about the noise you mentioned in your previous post?
I really would not worry about it. It is something in the reading. If you want to give me a call at the office we could try to track it down. Wish you could get rew to work for you.
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Quote:
Originally Posted by fetachin View Post
I'm afraid to ask, but how worried should I be about the noise you mentioned in your previous post?
The stuff you see still present after 1500ms is the noise floor of your gear and/or room:
Finished treating room - graphs good enough?-ir2-left-sub-1500ms.gif
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Quote:
Originally Posted by Jens Eklund View Post
The waterfall plots show a quite rapid decay except for a few modal frequencies: (25, L), 29, 41 & 50/54 (R/L) Hz primarily. The 120 Hz trail is noise, presumably the first harmonic of the mains if you´re in a 60 Hz AC county?
Yes, U.S., 60Hz AC. Do you think the rapid decay plus the modal frequencies you noted are acceptable for a small home studio? Especially now that I am aware of them and pay special attention to those frequencies (headphones, etc.)?


Quote:
Originally Posted by Jens Eklund View Post
The ETC reviles a strong early reflection at about 1,4 ms (after direct) and it´s a lot stronger in the left channel. There’s also some very early reflections (possibly caused by diffraction from the speaker itself or nearby obstacles or screens) within the first ½ ms or so. Except for that, the ETC is quite clean and indicates a close to anechoic speaker-room response (but this is not necessarily a bad thing, it depends on the design concept desired).
I ran a few tests with absorption material on my desk and the early reflection at 1,4 ms consistently goes away.

I had a few extra 2' x 4' absorption panels that I mounted in addition to the ones I places at first reflection points (left and right sides, back wall, ceiling). Should I consider removing them to make the room less anechoic-room sounding?
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Quote:
Originally Posted by fetachin View Post
Yes, U.S., 60Hz AC. Do you think the rapid decay plus the modal frequencies you noted are acceptable for a small home studio? Especially now that I am aware of them and pay special attention to those frequencies (headphones, etc.)?




I ran a few tests with absorption material on my desk and the early reflection at 1,4 ms consistently goes away.

I had a few extra 2' x 4' absorption panels that I mounted in addition to the ones I places at first reflection points (left and right sides, back wall, ceiling). Should I consider removing them to make the room less anechoic-room sounding?
“Acceptable” is rather subjective, but I would at least try and get that 54 Hz mode under control. Anechoic speaker response or not is a matter of personal preference but you do want an even (as good as practically possible) decay thought the frequency range and it´s currently too long in the lows and short (too short or not depends on the design concept) in the highs. If you don´t like a dead sounding space, you can try and remove absorbers that don’t contribute in a positive way, or perhaps replacing them with diffusers:
1D diffusor - vertical or horizontal?

Something you do not want, no matter what design concept commonly used for control rooms, is strong early reflections that will cause image shifts, comb filter etc. degrading the reproduction quality. Assuming all early reflections from nearby walls/ceiling have been dealt with; increasing the distance between the speaker and desk, raising the speakers (and if so, a slight tilt so the acoustic axis is still pointing towards the ears), minimizing the desk size and also tilting it, redirecting the reflection away from head height, are some measures one can do to minimize these early reflections.
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Originally Posted by Jens Eklund View Post
“Acceptable” is rather subjective, but I would at least try and get that 54 Hz mode under control. Anechoic speaker response or not is a matter of personal preference but you do want an even (as good as practically possible) decay thought the frequency range and it´s currently too long in the lows and short (too short or not depends on the design concept) in the highs. If you don´t like a dead sounding space, you can try and remove absorbers that don’t contribute in a positive way, or perhaps replacing them with diffusers:
1D diffusor - vertical or horizontal?

Something you do not want, no matter what design concept commonly used for control rooms, is strong early reflections that will cause image shifts, comb filter etc. degrading the reproduction quality. Assuming all early reflections from nearby walls/ceiling have been dealt with; increasing the distance between the speaker and desk, raising the speakers (and if so, a slight tilt so the acoustic axis is still pointing towards the ears), minimizing the desk size and also tilting it, redirecting the reflection away from head height, are some measures one can do to minimize these early reflections.
Thanks Jens. I really appreciate all the help.
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Thanks Jens. I really appreciate all the help.
No problem!
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Hi,

Fetachin

what kind of sound card do you use?
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Hi,

Fetachin

what kind of sound card do you use?
Apogee Duet 2.
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Levels

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I´m still curious what the rec level meter in Fuzz indicated when recording?
It reads 0-94dB by default. This can be calibrated to actual SPL using an SLM or Calibrator. However, as a default it does a pretty good job. When the Sweep level gets uncomfortably loud, to the point of ear protection, we tend to read 90dB or so.

Simple advice would be the same for both FM and REW users. Push actual Sound Pressure Level in the room to ear protecting heights. Then adjust the mic pre to hit close to full level on the input meter.
This is using the software as designed.

I am very curious as to why Chris does not recommend using Synch Averaging.

24 Bit recording doesn't happen in REW. afaik input capture is 16Bit, which is not a problem unless level setting is done really badly.

Normalising the Files and Exporting with 24 Bit resolution seems wise.
But I am not sure how REW will treat a 24 Bit Import.

DD
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It read 0-94dB by default. This can be calibrated to actual SPL using an SLM or Calibrator. However, as a default it does a pretty good job. When the Sweep level gets uncomfortably loud, to the point of ear protection, we tend to read 90dB or so.

Simple advice would be the same for both FM and REW users. Push actual Sound Pressure Level in the room to ear protecting heights. Then adjust the mic pre to hit close to full level on the input meter.
This is using the software as designed.

I am very curious as to why Chris does not recommend using Synch Averaging.

24 Bit recording doesn't happen in REW. afaik input capture is 16Bit, which is not a problem unless level setting is done really badly.

Normalising the Files and Exporting with 24 Bit resolution seems wise.
But I am not sure how REW will treat a 24 Bit Import.

DD
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