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used this thread to run sweeps instead of pink noise as i was having a hard time and it has worked out well... i think lol, new to this and i dont know anyone around st pete that i can bother for help..

let me know what you guys think and if i need more attention in certain areas. using SI 3 way with a SI 3.5 for center as well in 2015 f150 plat. hence why the 1500 mark is so high i think.
 

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Post graphs in 5db per vertical division and 6db or var smoothing plz


Looks like a good start. Might need to tweak levels on some drivers.
 

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Looks like you've got a great start. I'd adjust tweeter levels to match a bit closer. One side appears to be 5db louder, shouldn't be too difficult to bring that down to match.
 

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Great work Hanatsu, I really look forward to seeing this develop further.

Do you plan to talk about connecting the 2nd channel in a "loopback" configuration? I find it is exceptionally useful for time alignment.
Sorry to tag you Justin, can you provide a picture of how to cable up an interface for loopback measurement (or point me towards somewhere that has one), I have an m-audio usb mobile preamp on its way to me as I type and a Dayton mic as the current Behringer ones don’t get decent reviews...

I want to specifically do just what you speak of...

Or the same question for hanatsu...

I’ve seen various guides and I believe it’s routing the output of the device that comes from the left channel into the right channels input or something like that so it can see when it’s timing pulse is sent out and compare that to when it sees it on the mic and effectively zero that so you just get an accurate time of the impulse

Also what do you use for impulse response? Rew or holm?
 

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Sorry to tag you Justin, can you provide a picture of how to cable up an interface for loopback measurement (or point me towards somewhere that has one), I have an m-audio usb mobile preamp on its way to me as I type and a Dayton mic as the current Behringer ones don’t get decent reviews...

I want to specifically do just what you speak of...

Or the same question for hanatsu...

I’ve seen various guides and I believe it’s routing the output of the device that comes from the left channel into the right channels input or something like that so it can see when it’s timing pulse is sent out and compare that to when it sees it on the mic and effectively zero that so you just get an accurate time of the impulse

Also what do you use for impulse response? Rew or holm?
If your using a USB mic I think you'll need to use HOLMImpulse because REW will not do time-domain via USB from what I remember.

Also, I think subterFUSE posted a napkin sketch flowchart of a basic loopback setup earlier in the thread. But yes, you basically take an output from your audio interface and route it back to another input.
 

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I'm aware of that little issue. If you use the 1/3oct spaced filters I provided and generate EQ filters with 'Optimize Frequencies, Gain, Qs' instead of full auto it will generate filters that's at 10% +/- of the 1/3oct standard octave spacing which the Helix handles fine ;)
Great post Hanatsu - Thank you so much

Where can I get your "31 band graphic/parametric EQ Filter" file in order to use with a Match DSP (Same DSP tool as the Helix) ?
(the link does not respond anymore)
 

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Great post Hanatsu - Thank you so much



Where can I get your "31 band graphic/parametric EQ Filter" file in order to use with a Match DSP (Same DSP tool as the Helix) ?

(the link does not respond anymore)


I would like to know also


Enviado desde mi iPhone utilizando Tapatalk
 

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Using this guide. When setting my filters before autoeq, they’re all set at 10 on the q. My dayton dsp default is 2.515. Do I set all of the Qs to that or leave them alone?

Also the dayton is a 10 band. Do i take the high and low frequency of the driver im tuning between the xover and divide that by 10 and set my targets there for the pre auto eq points? Or just line em up normal on what I can fit? This is a 2 way setup, so ive got more than 10 bands on the mids and highs.


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Just tried again. Attaching all my auto eq curves...plugged them in and it sounds absolutely horrible...

Started over completely with pink noise and did each driver at a time. Then level matched and adjusted to as close to the curve as possible. Decent, but nowhere near where it could be.

Running pink noise through rew, L&R channel -22db
With the built in generator and my mic wont read anything over 10,000hz...

I got no idea what’s going on.


Im so tired of getting ear fatigue and having a headache from doing this. My ears burn.




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What was the mic problem (and solution)?

Those auto-eq values sure do seem strange. I don't think I've ever had REW recommend all filters with the exact same dB and Q values like that?

I haven't read back through the thread yet, but will do that now (curious to see actual measurement graphs, etc).

I will say this - as another noob to all of this SQ tuning and DSP stuff, I've found that tuning to a target curve produces very different sound than what I was used to with just setting a regular graphic EQ to the "smiley face" pattern, like I've always done before. I think that folks like us are used to over-emphasized bass and treble, so a "technically correct" system sounds a little flat and lifeless in comparison.

I would get things tuned in to be "technically correct" - and make sure the left and right sides match as much as possible and then use some sort of "global" EQ to adjust for preference. Or - you can create your own target curve to include some extra emphasis on bass and treble, for example (or whatever your preference is).

EDIT: actually, I don't see any graphs posted - can you post some graphs of what your system currently looks like? Actually, you really should start your own thread for this discussion since it's not really specific to Jazzi's tool....
 

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This thread isn’t about jazzi’s tool...

Range for pink noise was cutoff for some reason in the program.

I made a new set of filters relates to my dsp and load those now when using auto eq and select optimize q, gain and freq now.

I think my major problem is I need to add a 2.5” mid more focused to the driver seat.

Mids are mounted in the bottom of the doors and they fire directly at each other. My head is way off axis from the driver side.


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This thread isn’t about jazzi’s tool...

Range for pink noise was cutoff for some reason in the program.

I made a new set of filters relates to my dsp and load those now when using auto eq and select optimize q, gain and freq now.

I think my major problem is I need to add a 2.5” mid more focused to the driver seat.

Mids are mounted in the bottom of the doors and they fire directly at each other. My head is way off axis from the driver side.


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My bad - not Jazzi's tool (not sure why I said that!) - but I'm just saying that you'd probably get better/more focused feedback/help if you started your own thread - otherwise, there will end up being lots of un-related posts that get in between your posts and posts responding to you - and it can become hard to follow in a general "how-to" thread, that's all...

You should be able to get good sound quality even when using the factory speaker locations. You can definitely get better than "sounds like crap". :) Customizing the speaker locations, angles, etc is great and can definitely make things better, but there are a lot of people with great sounding systems using factory speaker locations. I don't think that is your main "bottleneck" right now - at least based on your descriptions of your issues.
 
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