Beginner trying to figure stuff out.

[Bobski321]

Hello

Forgive my inexperience, but I do have some things I'm trying to figure out.

I'm wondering if audiolense has a way to correct for phase issues caused by crossovers. I've used REW, and had to use re-phase to compensate for phase shifts. Does audiolense automatically do this?

Thanks.

 

[ogs]

Yes, the XO version does this on speakers with built-in crossovers. Time aligns drivers and corrects for crossover phase errors, Also corrects for any difference in speaker L/R distance from listening position.

 

[Bobski321]

Thanks so much for your reply. That's good to know.

Another quick question.... Is a measurement that says "good dynamic range"... Inferior to one that reads out "excellent dynamic range"?

Should I always have the volume that loud when taking measurements?

Thanks again.

 

[ogs]

I have experienced this but I can't say I totally understand the difference. I have achieved very good corrections with both "good.." and "excellent.." @juicehifi might want to chime in and tell us what it actually means.

 

[juicehifi]

The dynamic range assessment is just a comparison of the peak to the noise floor. This will typically be worse for a woofer than a tweeter (and a fullrange speaker), so I don't pay too much attention to it.

Sometimes I look at this, though:

This is a log measurement view that can be generated from the analysis menu. In this instance I disabled the noise reduction. As you can see, the visible noise floor is around -85 dB here (the flat region to the right of the peak). The calculated noise floor here is probably around 90-95 dB down here, which is plenty good for our purpose. This noise floor is in the typical region for an entry level microphone feed. Better figures can be had when a profesjonal sound card with embedded microphone input is used, but I doubt that it makes an audible difference in the end. But if I see this noise floor creeping towards -60 dB and higher I tend to suspect that something is wrong. And sometimes I see irregularities to the left of the peak that may also indicate some technical problems. The gradual rise prior to the peak is as it should be with the measurement used in Audiolense.

 

[Bobski321]

This is great information, thank you.

Just to finish off the measurement questions....I have one more.

I'm slightly confused about the length of measurement. I would assume that the longer measurement would be more accurate... This is something I always did in REW. But it seems recommended that 10 seconds is just fine. Is there any advantage to going longer? Like 30 seconds or maybe even a full minute?

Thanks again for your time.

 

[ogs]

Keep in mind that 10 seconds is a really long time for sound. Speed is 340 meters per second.. I'd say that the default 10 seconds is a good choice. Longer sweeps may give better measurements if your room is very difficult (lively and reflective) or speaker positioning is compromised, but I have never used more than 30 seconds.

 

[Bobski321]

But then wouldn't it just be better to use longer measurements then right? Maybe not necessary, but if you're saying sometimes it may be beneficial, then why not just use the higher number?

What are the downsides of using higher measurements?

The only thing I've observed is that the impulse response isn't as strong on longer measurements.

 

[juicehifi]

Yes, if everything works, longer sweeps will lower the noise floor, but this is unlikely to make a difference to the correction filters. On the other hand, shorter gives less time for external noise to interfere and for problems in the digital domain.

 

[Bobski321]

So....when drawing a target curve, how low should you draw the bass drop off? I see most examples are at -30 or so. Do you have any recommendations? Or is it really just personal preference?

As you can see, I went down to -42.

Edit: I'm actually finding it sounds a bit better closer to -30.

 

[Ofer]

Your target looks fine. A few notes from my experience.
It helps to have the target close to the measurement, in your case you have a drop at ~1.8khz and another drop at 11khz (which is strange, why is that?). Try to create a target that follows those drops.
For music I find the mixed phase targets most favorable. A 50% minphase and 50% linear phase sound great with TTD corrections. What audio interface do you have. Sometimes long sweeps cause errors that short ones don't.

 

[Bobski321]

So...when you say your target looks good, you didn't mention your opinion on how far down you would recommend to go with the bass before I complete the target. I'm really interested in people's experiences with that.

I've tried the 50%/50% thing. Thought it sounded great, but I prefered straight minimum phase for rock music, and thats mostly what I listen to. But yes, it does sound different and great.

I don't use treble boost, so the dip at 11khz with this target is un-touched and unaffected, so no point on following the dip in my target. And the dip at 1.8khz I could try a target that follows it, but it's mostly the one channel that does that.... I'm not sure I understand why you'd want to do that, unless you strongly prefer that characteristic that I think is being created by the room or something.

This measurement is a 10 second measurement. I'm not sure what you mean by my audio interface? You mean, what microphone I'm using?

Thanks for the input

 

[Ofer]

So...when you say your target looks good, you didn't mention your opinion on how far down you would recommend to go with the bass before I complete the target. I'm really interested in people's experiences with that.

My target goes down to -40db but I haven't experimented with that much.

I've tried the 50%/50% thing. Thought it sounded great, but I prefered straight minimum phase for rock music, and thats mostly what I listen to. But yes, it does sound different and great.

I don't use treble boost, so the dip at 11khz with this target is un-touched and unaffected, so no point on following the dip in my target. And the dip at 1.8khz I could try a target that follows it, but it's mostly the one channel that does that.... I'm not sure I understand why you'd want to do that, unless you strongly prefer that characteristic that I think is being created by the room or something.

Sometimes AL tries to compensate the measurement to the target curve, boost in your case. That is why it is recommended to have the target close to the measurement. Both Yellow and green have the dip at 1.8khz so it is probably both channels. Is it the driver characteristics, room?

This measurement is a 10 second measurement. I'm not sure what you mean by my audio interface? You mean, what microphone I'm using?

Thanks for the input

How do you measure? what do you use to play music? USB mic? Reciver? soundcard connected to something? in my case it is an RME ff-400 audio interface, that is why I asked.

 

[Bobski321]

I just use a umik-1 connected to a laptop.... something I'm looking into upgrading, but I'm not ready to spend the coin right now.

My system consists of Qobuz, through Roon, through HQplayer, out USB into Holo Spring 3 KTE DAC, into Kinki Studio integrated, out to Buchardt s400 mk2 speakers. Dedicated listening room with some acoustical treatment.

I'm very happy with the sound I'm getting right now....just learning Audiolense and how to maximize its potential.

 

[Ofer]

That is an impressive setup. The convolver is going through roon? It is impressive that such a small speaker goes all the way to ~40hz.

 

[Bobski321]

The convolver is going through HQplayer where it also is upsampling everything to DSD512.

The Buchardts are a remarkable speaker. Very small, but somehow manages to have decent bass extension with some punch.

 

[Mitchco]

So...when you say your target looks good, you didn't mention your opinion on how far down you would recommend to go with the bass before I complete the target. I'm really interested in people's experiences with that.

It is always good to inspect the actual correction filters themselves (i.e. chart view details: correction) to see if:

a) you are getting any narrow band artefacts in the frequency response, and if yes, then reduce the amount FDW cycles in the TTD sub window unitl it goes away.
b) with the frequency scale showing down to 1 Hz, where is the correction frequency response reaching 0 dBFS (actually Bernt limits to -1 dBFS to prevent any overs). What you are looking for is there any unnecessary "boosting" at the frequency extremes that is causing unnecessary filter attenuation. Typically seen at the frequency extremes.

So for both frequency extremes it is good practice for the target to follow the natural rolloff of both the bass and treble of your loudspeakers. There are some caveats, e.g. ones speakers can go lower and have the power (i.e. watts) to extend the response. Or you are using constant directivity waveguides and the room correction is also providing the necessary constant directivity HF eq boost.

Specifically to answer your question, I often go down to -60 dB or even lower below 10 Hz and straight line to the left to 1 Hz. There is no good reason to pass any significant signal below 10 Hz in the correction filter.

Another pro tip, once you have determined your target preference and CPD settings, a fine tuning step most people miss is to keep reducing the amount of max correction in the CPD down to like 3 dB or even 1 dB max. There is a sweetspot there that one can "bracket" and compare filters. It's a tradeoff between smooth frequency response and how dynamic the music it sounds.

Have fun and experiment!

 

[Bobski321]

Super awesome info... really appreciate it.

 

[Bobski321]

So... Do most people just set filter length to 131k taps and call it a day? Is more generally considered better?

 

[Bobski321]

Any input on the number of taps?

Another question. Is there a way to increase the accuracy of the filter or correction? And I'm not just talking about Max boost. I know the filter is designed in a way to not be absolutely accurate, by considering psychoacoustics and I suppose be a little more forgiving. Is there a way to increase the accuracy? Hope that question makes sense.

Thanks.

 

[juicehifi]

For regular speakers and below 96kHz sample rate, 65 k taps is usually plenty.

You will get more detailed correction by increasing the measurement window and the ttd window. Whether it will be more accurate is up to you

 

[Bobski321]

Resurrecting an old thread of mine from the dead.

I've been using Audiolense now for almost a year, and its been phenomenal.

I'd love to dig in a little more, and see if I can improve things ...so I need some help to get started.

In another recent thread, Juicehifi mentioned some things that are priority. Here is what he posted:

"1st priority is a best possible step-response, but with a reasonably clean entrance.

2nd pri is correction filters that does not have substantial pre-ringing."

I'd like to have a look at these things, but I'm not sure how to access them, and have a look.

Could someone please tell me where in the menu I should be looking for these? And I guess more specifically, what I'm looking for when I do pull up the right display.

Thanks so much.

 

[juicehifi]

There’s a mouse zoom with click, hold and drag a recysngle from top left to bottom right. That’s how you zoom in Ina few ms.

 

[juicehifi]

Post a couple of charts and well take it from there.

 

[Bobski321]

I know how to zoom, I'm just not sure where in the menu to go. I'm not sure how else to explain it better.

So....am I clicking "Analysis"....then "measurement" then....."step response measurement"???

Or is it "simulation" I want to go into?? Then what?

Am I making any sense?

This is probably super simple for most people....