convolution room correction and room curve

moedra · Sep 2, 2024

linuxonly said:
Hi,

I can see 1/6 is less detailed but why is it desirable? Less latency? Smaller files? Why 300 Hz? Is the difference audible? I lack the theory concerning smoothing.
No need to explain everything, just point me in the right direction, unless the explanation is very basic.

In my mind 1/12 represents 1 note of the musical scale resolution, 1/6 would represent 2. I might be wrong but intuitively that's what I underdand.

I already have those books:
Sound Reproduction The Acoustics and Psychoacoustics of Loudspeakers and Rooms Floyd Toole
Master Handbook of Acoustics
THE ACOUSTICS AND PSYCHOACOUSTICS OF LOUDSPEAKERS AND ROOMS – THE STEREO PAST AND THE MULTICHANNEL FUTURE

View attachment 73415

All we are trying to do is correct for the signal while ignoring reflections, which is why we need the smoothed response to be used instead of the raw data. Since the raw data contains the reflections, we would be over-correcting if we do the calculations with it. Hence the need to bake the smoothing into the measurements. We are just trying to not over-correct, so we can get better results by smoothing the highs more than the lows. Focusing more resolution on the low end is better, because the reflections there show up in a different form and are dealt with differently.

300Hz in this case was just an example. You should put the crossover (blend) frequency wherever your room's transition zone is, so that more resolution can be used to deal with the low end, where the crests in the waves are further apart. It seems backwards, but doing it this way almost always sound better. Var smoothing is following the same principle, but it's a little too smooth in the top for me. 1/12 nails the low end for me, and I like 1/6 up top. 1/6 is more in line with how our ears hear everything without being too vague. It's really a matter of what sounds best to you.

bixite · Sep 3, 2024

moedra said:
I think I will. In the meantime, here's a workaround that doesn't require editing txt files. It doesn't interpolate the values across the spectrum, but it does at least blend the two values together at a definable frequency over a very small fixed range. Unfortunately, we can't define the blend range, but at least we have something.

1 • Export the measurement at 1/12 smoothing
2 • Export the measurement again at 1/6 smoothing
3 • Import both smoothed responses
4 • Go to the arithmetic panel and merge B to A as follows with blend enabled:
- A is the smoother response, i.e. 1/6
- B is the more detailed response, i.e. 1/12
- Type in the frequency at which you want the blend to happen, then generate the result

The result of the math will leave you with a measurement with 1/12 smoothing below the blend frequency and 1/6 smoothing above it. All that remains is to generate a min phase copy of this measurement.

View attachment 73414

Great and thanks! I just tried this out and it seems that this approach can be iterated with more smoothing (e.g. 1/12 up to 200, 1/6 up to 8k, 1/3 above 8k). This works by using the "Merge B toA" result with the next smoothing.
This is much more comfortable than editing the TXT-Files and the "Blend" option automatically takes care for smooth transitions!
I notice a little variation in the SPL of the generated measurement which I do not understand.

linuxonly · Sep 3, 2024

moedra said:
All we are trying to do is correct for the signal while ignoring reflections, which is why we need the smoothed response to be used instead of the raw data. Since the raw data contains the reflections, we would be over-correcting if we do the calculations with it. Hence the need to bake the smoothing into the measurements. We are just trying to not over-correct, so we can get better results by smoothing the highs more than the lows. Focusing more resolution on the low end is better, because the reflections there show up in a different form and are dealt with differently.

300Hz in this case was just an example. You should put the crossover (blend) frequency wherever your room's transition zone is, so that more resolution can be used to deal with the low end, where the crests in the waves are further apart. It seems backwards, but doing it this way almost always sound better. Var smoothing is following the same principle, but it's a little too smooth in the top for me. 1/12 nails the low end for me, and I like 1/6 up top. 1/6 is more in line with how our ears hear everything without being too vague. It's really a matter of what sounds best to you.

Amazing. I found a detailed explanation there - https://www.digistar.cl/forum/viewtopic.php?t=794

Calculated transition zone to be 730 Hz in living room (4,22 m x 3,76 m x 2,29 m) and 790 Hz in music room (6.03 x 3.27 x 2.67m). Will experiment from that

@moedra Thanks a lot. I'm very curious about it. I can easily make A B comparisons in living room but not in stereo room where the speakers aren't in the same room as the computer, in fact they're not even on the same floor.

moedra · Sep 3, 2024

bixite said:
Great and thanks! I just tried this out and it seems that this approach can be iterated with more smoothing (e.g. 1/12 up to 200, 1/6 up to 8k, 1/3 above 8k). This works by using the "Merge B toA" result with the next smoothing.
This is much more comfortable than editing the TXT-Files and the "Blend" option automatically takes care for smooth transitions!
I notice a little variation in the SPL of the generated measurement which I do not understand.

Yes of course! Just don't forget to generate a minimum phase copy of your final smoothed response.

moedra · Sep 3, 2024

linuxonly said:
Amazing. I found a detailed explanation there - https://www.digistar.cl/forum/viewtopic.php?t=794

Calculated transition zone to be 730 Hz in living room (4,22 m x 3,76 m x 2,29 m) and 790 Hz in music room (6.03 x 3.27 x 2.67m). Will experiment from that

@moedra Thanks a lot. I'm very curious about it. I can easily make A B comparisons in living room but not in stereo room where the speakers aren't in the same room as the computer, in fact they're not even on the same floor.

Cool. Honestly, I think 1/12 smoothing over the entire range sounds amazing. If you don't notice any difference with different smoothing levels, save yourself some time and effort and just use 1/12 for everything. It's detailed enough to control the lows and smooth enough to correct the highs in a very pleasing way.

linuxonly · Sep 4, 2024

The audio quality is unbelievable and the procedure is very simple to follow. I have, in my mind, a quality stereo system and it never sounded like that. All controls, tone, balance, loudness are kept flat for best results. That is something I never did before.

My stereo system evolved from a custom made preamplifier build with quality op amps, a Quad 33 303 as main power amp and 2 9 cu in ft bass-reflex loudspeakers using LC crossovers, to a 6 channels individually driven speaker where the tweeter pair is driven from one amplifier, the mids from a second (the Quad 33 303) and the woofers, by another stereo amp. The loudspeakers were build in 1975 and the 15 in woofers reconed in 2000.

https://sound-au.com/project09.htm presents the 24 dB/Octave 3-Way Linkwitz Riley Crossover to place between the preamp and the 3 stereo power amps. I got rid of the LC crossovers. The Quad 33 303 being an inverting amplifier, it's outputs must be inverted (i.e. - to +). The Quad, being an over 50 years old beast, was recapped 8 years ago and a 60 Hz or 120Hz hum issue was fixed. All in all, apart from the Quad 33 303, the system was designed and build from scratch, resulting in a high performance, low budget stereo system.

It's quite an experience to listen to a bass being played and being able to hear every single note with the same intensity and coming from the same point in space.

I couldn't acheive something comparable with tone controls where the 2 crossover points could be varied all over the range as well as the gain, graphic equalizers and so on. It was better, but still far from what we're acheiving here.

I lost 15 dB in the correction process, but the system is now linear from 13 Hz to 22 kHz.

@moedra Thanks again for your hard work. You made someone very happy, and I'm sure you will continue to make many others more.

And indeed, my heartfelt thanks and kind regards to @John Mulcahy

moedra · Sep 4, 2024

linuxonly said:
The audio quality is unbelievable and the procedure is very simple to follow. I have, in my mind, a quality stereo system and it never sounded like that. All controls, tone, balance, loudness are kept flat for best results. That is something I never did before.

My stereo system evolved from a custom made preamplifier build with quality op amps, a Quad 33 as main power amp and 2 9 cu in ported loudspeakers using LC crossovers, to a 6 channels individually driven speaker where the tweeter pair is driven from one amplifier, the mids from a second (the Quad 33) and the woofers, by another stereo amp. The loudspeakers were build in 1975 and the 15 in woofers reconed in 2000.

https://sound-au.com/project09.htm presents the 24 dB/Octave 3-Way Linkwitz Riley Crossover to place between the preamp and the 3 stereo power amps. The Quad 33 being an inverting amplifier, it's outputs must be inverted (i.e. - to +). The Quad, being an over 50 years old beast, was recapped 8 years ago and a 60 Hz or 120Hz hum issue was fixed. All in all, apart from the Quad 33, the system was designed and build from scratch, resulting in a high performance, low budget stereo system.

It's quite an experience to listen to a bass being played and being able to hear every single note with the same intensity and coming from the same point in space.

I couldn't acheive something comparable with tone controls where the 2 crossover points could be varied all over the range as well as the gain, graphic equalizers and so on. It was better, but still far ftom what we're acheiving here.

@moedra Thanks again for your hard work. You made someone very happy, and I'm sure you will continue to make many others more.

Love it. Spread the word around to anyone you know who can take advantage of it. I will begin working on a video.

JStewart · Sep 10, 2024

moedra said:
If anyone else has a moment to try out the procedure and let me know what your experience is, I'd appreciate it.

@moedra , I did a run though your procedure. Followed it step by step looking for anything amiss. The procedure is error free. I haven't upgraded to the latest REW version so the UI wasn't 100% the same, but that’s to be expected.

The filter's sound very good! Once again. nice piece of work! Very easy. Great results. Wish my car dsp did convolution.

I reserved a few dB headroom in Roon for the little filter boost at 250Hz. I should have dropped the target a touch more to adhere to the exact instructions.

bixite · Sep 10, 2024

JStewart said:
... I reserved a few dB headroom in Roon for the little filter boost at 250Hz. I should have dropped the target a touch more to adhere to the exact instructions...

How did you set MaxGain in the TraceArithmetic of A/B. Since you have a boost, you probably have allowed a few db there, correct?

Did you measure, if the dip went away? When boosting you do not know in advance, if it will have an effect.

JStewart · Sep 10, 2024

bixite said:
How did you set MaxGain in the TraceArithmetic of A/B

I didn’t. Since the target is at or below the response everywhere but 250Hz gain is effectively unchanged or reduced. While I doubt I’d run into digital clipping at 250Hz, it just seemed good hygiene to take the gain down a couple of dB in Roon. I‘ll turn it off sometime and see if the Clipping indicator lights.

bixite said:
Did you measure, if the dip went away? When boosting you do not know in advance, if it will have an effect.

No and II didn’t bother measuring after because I already knew what to expect from prior experiences over the years. It’s caused by SBIR and will mitigate some, but not go away with EQ.

JStewart · Sep 10, 2024

My speakers have always sounded better to my ears by not EQing too high if I'm using min phase filters. For reasons I don't understand, Dirac and Focus Fidelity, which do impulse correction, sound better full range.
It occurred to me that instead of merging, for example, a 1/6 and 1/12 octave response, why not just merge the target with the response? Then no correction will be made whatsoever in the chosen region. In this case, above 235Hz.

Here is the original and "target merged" response. Somehow REW took care of getting the measurements relative SPL between the two correct on it's own.

And the merged responses with filters.

This sounds better to me with my speakers and room which, I will admit, may just be because I'm more used to it.

Enough fun for now.

Edit:

moedra · Sep 11, 2024

@JStewart You are probably very used to the way your speakers sound. The response seems flat enough to warrant that. Dirac has a proprietary way of handling the top frequencies. It is very odd, what it does. I can't really explain it but I have measured the filters it generates. They seem somehow phase manipulated in the top. Focus Fidelity does true time domain correction, so that is one sure-fire way to account for better full range filtered sound.

It's unclear to me how you are getting any boosting over zero on the scale in REW with your filters. Following my guide, you should not see anything reach up over 0.. If Max Gain is left at 0.0 you should have no positive amplitude adjustments. There must be a discrepancy based on your version of REW.

JStewart · Sep 11, 2024

moedra said:
It's unclear to me how you are getting any boosting over zero on the scale in REW with your filters. Following my guide, you should not see anything reach up over 0.. If Max Gain is left at 0.0 you should have no positive amplitude adjustments. There must be a discrepancy based on your version of REW.

When I have some more time I’ll upgrade and start over.
The trace arithmetic control panel on the version I’m using has a “regularisation” (Sp?) control and not “max gain”. Not sure what “regularisation “ does.
I also used measurements created in Impala and imported to REW. Perhaps there is some unwanted small change taking place on the impulse response import. I’ll take them in REW next time.

linuxonly · Sep 11, 2024

JStewart said:
The trace arithmetic control panel on the version I’m using has a “regularisation” (Sp?) control and not “max gain”. Not sure what “regularisation “ does.

That is correct. IIRC Example of regulation that I tested a few times: 8% = 5dB max

Capture d’écran du 2024-09-11 10-12-57.png

moedra · Sep 11, 2024

linuxonly said:
That is correct. IIRC Example of regulation that I tested a few times: 8% = 5dB max

View attachment 73575

Right on, so update your version of REW and it should be alright.

JStewart · Sunday at 9:15 PM

moedra said:
Right on, so update your version of REW and it should be alright.

Updated REW to current beta and all of the inversion filter is below 0 dB now.

convolution room correction and room curve

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