Comprehensive Review: " APL1 " - Advanced DSP / EQ / Phase Correction Unit - Page 15 - Car Audio | DiyMobileAudio.com | Car Stereo Forum

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Old 11-28-2016   #351
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Default Re: Comprehensive Review: " APL1 " - Advanced DSP / EQ / Phase Correction Unit

Quote:
Originally Posted by Raimonds View Post
Yes of course : ) There is potential for more than 2 channels!
I would like to see some voting to understand how many supporters APL will have in such direction ...
Fir processing is rediculously powerful but 95% of us dont understand how or why it works. If you bring fir processing to android. I think you might find a new market. Hard to say. I wish people on here would go more in depth in how it works. All the explanations ive seen are biquad this and biquad that and way over the general populaces head. Its almost like auto tune when used right though. Its totally cheating.



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Old 11-28-2016   #352
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Default Re: Comprehensive Review: " APL1 " - Advanced DSP / EQ / Phase Correction Unit

This is exactly correct. I have had an APL1 in my car for months and never could figure out how to tune it, and therefore I have it disabled and never used it.

The instructions are so vague. The tutorial that was posted here is a nice start but not in depth enough for me to understand how to get it working.


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Old 11-28-2016   #353
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Default Comprehensive Review: " APL1 " - Advanced DSP / EQ / Phase Correction Unit

In my case, it would be competing against something in the Helix camp of the Audiotec-Fischer line up. Currently an 8-channel DSP. Having experienced the limitations of IIR, I'm certainly intrigued. But I too would need some self-education how tuning the system would change.

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Old 11-28-2016   #354
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Default Re: Comprehensive Review: " APL1 " - Advanced DSP / EQ / Phase Correction Unit

In the Pro Audio world, FIR filters have been a big hit. Many of the Pro DSP units offer FIR now. They are typically used to flatten the response of a speaker cabinet with multiple drivers. For example, let's say A company builds a speaker with a horn and a bass driver in a single cabinet. You can buy that speaker and then tell the manufacturer which DSP you have and they will send you the FIR filters to make that cabinet perfectly flat in response with a linear phase crossover. Then you put that cabinet into your venue and use the input EQ to adjust the tonality to taste or EQ the room.

In the pro audio world the FIR filters typically don't have more than 1024 taps, however. This is because more taps = more delay in the sound. Since most pro audio setups are for live music, long delays in the sound make it difficult to perform. You would basically be singing and then hear yourself on delay from the speakers which can get distracting.

So 1024 taps is generally considered a limit for a live PA system with FIR.


For a playback system more taps is ok because the delay won't be noticeable. More taps is better because as you get more taps you can have better resolution of the EQ in the lower frequencies.
APL1 has 4092 taps.


It does not replace your DSP. You still need the DSP for crossovers.

My general idea for how it works is that the APL can go in front of the DSP and essentially act like an input EQ on a pro audio DSP. You then use the DSP output EQ and crossovers to build your active crossovers for your drivers and then the APL can be used to apply a target curve and fix phase across the spectrum.

My issue is that I've never been able to understand how to work the software well enough to get it working.



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Old 11-28-2016   #355
 
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Default Re: Comprehensive Review: " APL1 " - Advanced DSP / EQ / Phase Correction Unit

I think I have an idea of how to use my APL now, but with ha;f the gear out of my car, it'll be awhile before I can actually play with it.

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Old 11-28-2016   #356
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Default Re: Comprehensive Review: " APL1 " - Advanced DSP / EQ / Phase Correction Unit

The problem I was having was that I set up my audio system to use only digital inputs on my DSP, and therefore I set my amplifier gains based on the DSP receiving a full level digital signal from my mObridge DA1.

When I added the APL1 and ran the software, it created EQ boosts which caused the music to sound like distorted garbage.

I was told to lower the volume on the mObridge, but I don't think this is the proper solution because then I'm going to have to redo the gains throughout the entire system and that's going to completely change the balance of my speakers so they would have to be EQ'd all over again.

That's the part I just don't get. There needs to be a procedure to setting up a full system when using the APL1.


I'm starting to think maybe the way it needs to be done is to EQ your speakers one-by-one and make them perfectly flat. Then apply your crossovers. And then use the APL1 to apply the target curve and let it fix any phase anomalies.

I don't think we would be using the FIR filters in the same way that they are used for pro-audio setups. In pro-audio, the speaker manufacturer takes their cabinet and puts it in a large room and measures it. They window the response so that all reflections are taken out of the measurement, which basically means it's like an anechoic measurement. Then they use FIR filters to build a perfect linear phase crossover for that speaker cabinet. You can then load those FIR filters and your speaker will be perfectly flat and in phase. But then you stick that speaker in a room and need to apply more EQ for tonality and room EQ. They do this with the input EQ banks on pro audio DSPs.

But, in a car, we can't really do it that way. The pro guys can do this because they are making a fixed speaker cabinet, where the individual drivers will always be oriented the same no matter where you put the cabinet. If we take the speakers out of the car, then we can't really put them in the same orientation to one another to get an anechoic measurement.


Also, in order to do like the pro-audio guys, you would need the FIR filters on the output side of the DSP. That way you could apply different filters to the drivers and build a crossover. So, in this case, it would mean multiple APL1 devices going after the DSP outputs.

So that's why I'm leaning towards the APL being used more like an input EQ. And that's what input EQ is used for on pro-audio DSP. They use output EQ to correct the speakers. And Input EQ is used for correcting the room, and for applying a desired tonality curve to the sound.

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Old 11-28-2016   #357
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Default Re: Comprehensive Review: " APL1 " - Advanced DSP / EQ / Phase Correction Unit

Quote:
Originally Posted by subterFUSE View Post
In the pro audio world the FIR filters typically don't have more than 1024 taps, however. This is because more taps = more delay in the sound. Since most pro audio setups are for live music, long delays in the sound make it difficult to perform. You would basically be singing and then hear yourself on delay from the speakers which can get distracting.

So 1024 taps is generally considered a limit for a live PA system with FIR.
Interesting, I had no idea. But what is the typical delay induced to be that noticeable?



Quote:
Originally Posted by subterFUSE View Post
The problem I was having was that I set up my audio system to use only digital inputs on my DSP, and therefore I set my amplifier gains based on the DSP receiving a full level digital signal from my mObridge DA1.

When I added the APL1 and ran the software, it created EQ boosts which caused the music to sound like distorted garbage.

I was told to lower the volume on the mObridge, but I don't think this is the proper solution because then I'm going to have to redo the gains throughout the entire system and that's going to completely change the balance of my speakers so they would have to be EQ'd all over again.

That's the part I just don't get. There needs to be a procedure to setting up a full system when using the APL1.
I had the same big distortion at the beginning, since it could boost more than 10db in some places, but tweaking the EQ target levels fixed it (APL1 feeding minidsp C-dsp).


Quote:
Originally Posted by subterFUSE View Post
I'm starting to think maybe the way it needs to be done is to EQ your speakers one-by-one and make them perfectly flat. Then apply your crossovers. And then use the APL1 to apply the target curve and let it fix any phase anomalies.

I don't think we would be using the FIR filters in the same way that they are used for pro-audio setups. In pro-audio, the speaker manufacturer takes their cabinet and puts it in a large room and measures it. They window the response so that all reflections are taken out of the measurement, which basically means it's like an anechoic measurement. Then they use FIR filters to build a perfect linear phase crossover for that speaker cabinet. You can then load those FIR filters and your speaker will be perfectly flat and in phase. But then you stick that speaker in a room and need to apply more EQ for tonality and room EQ. They do this with the input EQ banks on pro audio DSPs.

But, in a car, we can't really do it that way. The pro guys can do this because they are making a fixed speaker cabinet, where the individual drivers will always be oriented the same no matter where you put the cabinet. If we take the speakers out of the car, then we can't really put them in the same orientation to one another to get an anechoic measurement.


Also, in order to do like the pro-audio guys, you would need the FIR filters on the output side of the DSP. That way you could apply different filters to the drivers and build a crossover. So, in this case, it would mean multiple APL1 devices going after the DSP outputs.

So that's why I'm leaning towards the APL being used more like an input EQ. And that's what input EQ is used for on pro-audio DSP. They use output EQ to correct the speakers. And Input EQ is used for correcting the room, and for applying a desired tonality curve to the sound.
Agree, and same goes for Dirac, but after trying different methods like:
1 - optimize the system to flat (or any target but flat is easier) with all the DSP power available (XO, TA, PEQ) then let the APL fix all other anomalies + the final target.
2 - do the minimum on the dsp (ball park TA, XO and no EQ) and let APL to fix everything.
I got better results with #2, almost like if some agressives PEQs applied first could either fix or create more problems for the APL (in the time domain I guess).
And it's also a lot faster…

But now I want to try another method, a bit of both, focused on XO.
I mean with either APL or Dirac I believe that the most important part is to create a baseline that can accept safely any boost anywhere, so XOs are the most important.
Next time I will try to use all the dsp power to be sure all drivers don't play even a little out of their bandpass, but won't EQ anything "in" their bandpass.
I just started by using aggressive filter slope like 48db but it is not always enough, and I still have to compare the effect with 24db.
Because it's kind of hard to finetune both units with measurement methods that are not exactly the same.
For this Dirac is a lot easier to reproduce with REW (8 sweeps).

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Old 11-28-2016   #358
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Default Comprehensive Review: " APL1 " - Advanced DSP / EQ / Phase Correction Unit

The calculation for amount of delay in an FIR filter is as follows:

(N - 1) / (2 * Fs) = Delay


where N = # of taps and Fs = sample frequency



So... in the case of the APL1 which operates at 48kHz and has 4096 taps.

(4096 - 1) / (2 * 48kHz)

4095 / 96,000 = .0426 s

.0426 s = 42 ms



That's way too much delay to be acceptable in a live sound reinforcement system. It's ok for a playback system like our cars or your home stereo, but it would be impossible for a live performer to deal with that kind of delay unless their hearing was totally isolated from the audio system. This is why you so often see performers wearing in-ear monitors. It helps them isolate the sound reinforcement system from their own voice and playing.

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Old 11-28-2016   #359
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Default Re: Comprehensive Review: " APL1 " - Advanced DSP / EQ / Phase Correction Unit

Makes sense, thx!

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Old 11-28-2016   #360
 
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Default Re: Comprehensive Review: " APL1 " - Advanced DSP / EQ / Phase Correction Unit

Quote:
Originally Posted by Elgrosso View Post
Agree, and same goes for Dirac, but after trying different methods like:
1 - optimize the system to flat (or any target but flat is easier) with all the DSP power available (XO, TA, PEQ) then let the APL fix all other anomalies + the final target.
2 - do the minimum on the dsp (ball park TA, XO and no EQ) and let APL to fix everything.
I got better results with #2, almost like if some agressives PEQs applied first could either fix or create more problems for the APL (in the time domain I guess).
And it's also a lot faster…
Isn't #2 pretty much what Raimonds recommends? Get it as close as possible with just the XO and TA on the processor(s) after the APL, then use the APL to EQ and deal with phase issues.

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Old 11-28-2016   #361
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Default Re: Comprehensive Review: " APL1 " - Advanced DSP / EQ / Phase Correction Unit

Quote:
Originally Posted by LumbermanSVO View Post
Isn't #2 pretty much what Raimonds recommends? Get it as close as possible with just the XO and TA on the processor(s) after the APL, then use the APL to EQ and deal with phase issues.
But wouldn't one need to balance out the individual driver sets? Especially those running only a L and R input from the source unit. As we all know the individual outputs of each speaker can be quite different.

Or is this essentially somehow taken care of using the APL or DIRAC offerings?

I know Elgrosso kinda hit on this already. I just dont see how

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Old 11-28-2016   #362
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Default Re: Comprehensive Review: " APL1 " - Advanced DSP / EQ / Phase Correction Unit

Quote:
Originally Posted by LumbermanSVO View Post
Isn't #2 pretty much what Raimonds recommends? Get it as close as possible with just the XO and TA on the processor(s) after the APL, then use the APL to EQ and deal with phase issues.
Yes it is!


Quote:
Originally Posted by Beckerson1 View Post
But wouldn't one need to balance out the individual driver sets? Especially those running only a L and R input from the source unit. As we all know the individual outputs of each speaker can be quite different.

Or is this essentially somehow taken care of using the APL or DIRAC offerings?

I know Elgrosso kinda hit on this already. I just dont see how
Right I forgot individual levels!
And since there was no EQ I especially paid attention around the XOs while not caring about huge peaks in the bandpass.
Global L/R variations being managed by the APL or Dirac.

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Old 11-28-2016   #363
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Default Re: Comprehensive Review: " APL1 " - Advanced DSP / EQ / Phase Correction Unit

Quote:
Originally Posted by Elgrosso View Post
Yes it is!




Right I forgot individual levels!
And since there was no EQ I especially paid attention around the XOs while not caring about huge peaks in the bandpass.
Global L/R variations being managed by the APL or Dirac.
Ok that makes more sense.

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Old 11-28-2016   #364
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Default Re: Comprehensive Review: " APL1 " - Advanced DSP / EQ / Phase Correction Unit

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Originally Posted by LumbermanSVO View Post
Isn't #2 pretty much what Raimonds recommends? Get it as close as possible with just the XO and TA on the processor(s) after the APL, then use the APL to EQ and deal with phase issues.

The problem is, you can't really build a proper crossover unless the speakers have a flat response. Flattening the speaker response needs to happen first, then you can add the crossover filters and they will have a predictable nature.

Once you have flattened the speaker response, and then built the crossovers, now it's time to apply the tonality curve you want. Pro-Audio DSPs make this easy by having separate EQ banks on the Input side and Output side of the crossovers. The output EQ is used to flatten the response of each individual driver. Then you make your crossovers. And then you use the Input EQ to flavor the sound however you like, or to correct for room issues.

Car audio DSPs have not caught up to the pro-audio world yet. Some are getting close, like Helix. But they still have work to do. The tuning process on a pro DSP is so much easier because of the ability to break the process up into parts, and have separate EQ banks to handle different jobs.

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Old 11-28-2016   #365
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Default Re: Comprehensive Review: " APL1 " - Advanced DSP / EQ / Phase Correction Unit

I thought eqing outside the passband messed with phase after your crossover

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Old 11-28-2016   #366
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Default Re: Comprehensive Review: " APL1 " - Advanced DSP / EQ / Phase Correction Unit

Have you guys ever read posts about "Acoustic" crossovers vs. "Electric?"

What is meant by this is that sometimes when you apply an electrical filter to a speaker, the resulting acoustic response does not always equal the predicted result. For example, you have a speaker and you apply a 12 dB Butterworth filter at 500 Hz. But the resulting frequency response from that speaker ends up being closer to 24dB slope. That's because the speaker's natural response combined with the electrical filter you just applied yields a specific acoustic result that is different than the electrical filter.

Well, if you EQ the speaker flat then when you apply an electrical filter then the acoustic result will be the same as the electrical filter setting. This is very well known and understood among the pro-audio engineers of the world. That's why when it comes to tuning their systems, they always begin by isolating the speaker from the room and then making the response perfectly flat. Then they build crossovers, and the last step is to put the speaker in the room and do another round of EQ to correct the room issues and to make the tonality sound pleasing.

The first part of this process is pure science. The second part is art.

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Old 11-28-2016   #367
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Default Re: Comprehensive Review: " APL1 " - Advanced DSP / EQ / Phase Correction Unit

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Originally Posted by Jscoyne2 View Post
I thought eqing outside the passband messed with phase after your crossover

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Yes, it does. But it will actually help smooth out your phase response if the EQ is done to flatten the response.


In the most basic sense, that's what FIR filters are doing. They take a measurement of a speaker's response, and then make an inverted copy of that response and turn that inverted response into an EQ. When you perfectly invert a speaker's response, you have effectively made it flat.

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Old 11-28-2016   #368
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Default Re: Comprehensive Review: " APL1 " - Advanced DSP / EQ / Phase Correction Unit

Quote:
Originally Posted by subterFUSE View Post
The problem is, you can't really build a proper crossover unless the speakers have a flat response. Flattening the speaker response needs to happen first, then you can add the crossover filters and they will have a predictable nature.

Once you have flattened the speaker response, and then built the crossovers, now it's time to apply the tonality curve you want. Pro-Audio DSPs make this easy by having separate EQ banks on the Input side and Output side of the crossovers. The output EQ is used to flatten the response of each individual driver. Then you make your crossovers. And then you use the Input EQ to flavor the sound however you like, or to correct for room issues.

Car audio DSPs have not caught up to the pro-audio world yet. Some are getting close, like Helix. But they still have work to do. The tuning process on a pro DSP is so much easier because of the ability to break the process up into parts, and have separate EQ banks to handle different jobs.
Well until now I used mostly Jazzi's method (and Hanatsu).
So always combinations of XO+PEQ to get to a specific target per driver, acoustically.
Usually filters were applied roughly one or two octaves below and above each crossover points to get something clean.
And it just appeared that too much optimization before APL/Dirac could do more harm than good sometime, I don't know why.

I understand the value of what you described but I don't see how to do it in a car.
Do guys in pro world have to manage less FR variations maybe?

Horny Froggy...

Last edited by Elgrosso; 11-28-2016 at 11:46 PM..
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Old 12-02-2016   #369
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Default Re: Comprehensive Review: " APL1 " - Advanced DSP / EQ / Phase Correction Unit

Quote:
Fir processing is rediculously powerful but 95% of us dont understand how or why it works.
Almost all manufactures, that offer FIR, do not know what to do with it and its power. It is just fashion to add FIR and make some „product” more attractive. It’s cheating.

Thus the use of FIR is reasonable only in case when its power and accuracy is requested – to create very accurate predistortions that exactly reflect to distortions of system. If they will not accurately reflect – you will have new additional distortions.
Therefore you need trustable measurement to implement such idea and it is possible when the measurement and FIR use are integrated in a one complete solution.


APL tools are not a panacea to errors you made in your setup and tuning. You should fix up everything you can with ordinary methods and make you system as nice as possible. It is good to use curves of Workshop and TDA to do that.
Some example. I have customer Peter. Hi is a doctor surgeon. And here are curves of Workshop’s first measurement of his BMW. Hi got this just by his own efforts.


I am not usually suggesting to use APL1 unit in such well tuned setups because the client do not ketch improvement. But not in case with Peter. Peter was very happy with the result served by APL1 unit.

Another story is with different BMW. It has two volumes connected by some kind of pipe for a midbass loudspeaker. And that volumes and pipe are creating resonator on about 200 Hz and making very sharp and deep notch. It is impossible to equalize it properly but the partial measure is to reduce correcting peak by 2 times. That gives as not ideal but usable result.


Quote:
The problem I was having was that I set up my audio system to use only digital inputs on my DSP, and therefore I set my amplifier gains based on the DSP receiving a full level digital signal from my mObridge DA1.

When I added the APL1 and ran the software, it created EQ boosts which caused the music to sound like distorted garbage.
It is very popular mistake and is caused by “volume war”. Every sound system must have signal headroom. Usually and normally it is 16 dB. The “volume war” is pushing everything to FS (full scale) making headroom zero. You can’t do any signal processing without headroom! You must supply it! It must be done by a proper gain distribution in your system. On of tools to do that is to reduce APL1 filter’s gain. That allows to feed APL1 input with FS signal.


Quote:
So, in this case, it would mean multiple APL1 devices going after the DSP outputs.
Quote:
The problem is, you can't really build a proper crossover unless the speakers have a flat response. Flattening the speaker response needs to happen first, then you can add the crossover filters and they will have a predictable nature.
Yes, it is very powerful idea to equalize each band after crossover or even put together crossover and eq filters in one FIR. But it is expensive : )))

Quote:
In the pro audio world the FIR filters typically don't have more than 1024 taps, however. This is because more taps = more delay in the sound. Since most pro audio setups are for live music, long delays in the sound make it difficult to perform. You would basically be singing and then hear yourself on delay from the speakers which can get distracting.

So 1024 taps is generally considered a limit for a live PA system with FIR.
Quote:
So... in the case of the APL1 which operates at 48kHz and has 4096 taps.
(4096 - 1) / (2 * 48kHz)
4095 / 96,000 = .0426 s
.0426 s = 42 ms
It is very incorrect statement. 1024 taps can not serve LF equalization at all.
But the delay introduced by FIR filter depends on how it is created.
APL1 unit has 4096 coefficients FIR filters that are able to serve EQ even on 30 Hz and FIR filters generated by Workshop have 0 latency/delay.
The overall latency of APL1 unit is just 1.8ms caused by converters



Dear APL customers! You have option to use the online in person training that will give answers on any questions and give you first sounding result. If you did not use that, it is up to you …



BR,

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Old 12-04-2016   #370
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Default Re: Comprehensive Review: " APL1 " - Advanced DSP / EQ / Phase Correction Unit

I don't have a APL1 but I am getting a Dirac , but I am a FIR user and use open DRC like accurite
And I can imagine the issues you guys describe...

When I eq I have the best results turning crossover down two oactaves into the stop band .than eq each driver to be flat at least one oactave past crossover. That will make your acoustic slope match your electric slope, But the key is to eq the driver using a close mic measurement to speaker. Than do your FIR biz , than eq whole system with a single eq With mic at listening position. That way the harmonics of the room and the eq harmonic changes in between drivers are not un-balanced.


So maybe do your pre tune with close mic measurements on each driver. Than set TA than run your apL software Dirac or room eq ...

That's how I do it and it works. (And set your limits to not allow more than 3db boost. Or however you have to manipulate things so nothing is boosted a shit ton.) even a digital signal has a headroom and is not infinite.

I was having a hell of a hard time getting all my input/output Eqs to sound as harmonically balanced as a single 31band eq and nothing else with a mic measurement from listening position with a basic 30band RTA.

Yeah the old skool method sounded much much more balanced until I learned this routine .

im so in love with my speakers I found a song I sing to them

https://youtu.be/VgU6LXY-AjI

Last edited by oabeieo; 12-04-2016 at 10:36 PM..
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Old 12-04-2016   #371
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Default Re: Comprehensive Review: " APL1 " - Advanced DSP / EQ / Phase Correction Unit

And fir was supposed to make tuning easier...

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Old 12-04-2016   #372
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Default Re: Comprehensive Review: " APL1 " - Advanced DSP / EQ / Phase Correction Unit

What raimonds said is also true

You can have a 2020 tap filter that will be 10.51ms or more (or less if you hack the impulse to be band limited) depending on windowing and centering the impulse. Filter length isn't fixed on amount of taps. There is a formula out there that will give you the MAX length for a filter based on how many taps but that filter may very well be shorter depending again on window used and centering of the impulse.

1024 taps however can do some LF correction in phase only, pretty useless in phase plus mag , but to linearize a IIR crossover even a 1024t 96k can get down to at least 50hz

Much below 50hz 1024t just falls short unless you run at 44k than you could squeak into the high 30s I suppose.

im so in love with my speakers I found a song I sing to them

https://youtu.be/VgU6LXY-AjI

Last edited by oabeieo; 12-04-2016 at 11:05 PM..
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Old 12-05-2016   #373
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Default Re: Comprehensive Review: " APL1 " - Advanced DSP / EQ / Phase Correction Unit

Quote:
Originally Posted by oabeieo View Post
I don't have a APL1 but I am getting a Dirac , but I am a FIR user and use open DRC like accurite
And I can imagine the issues you guys describe...

When I eq I have the best results turning crossover down two oactaves into the stop band .than eq each driver to be flat at least one oactave past crossover. That will make your acoustic slope match your electric slope, But the key is to eq the driver using a close mic measurement to speaker. Than do your FIR biz , than eq whole system with a single eq With mic at listening position. That way the harmonics of the room and the eq harmonic changes in between drivers are not un-balanced.


So maybe do your pre tune with close mic measurements on each driver. Than set TA than run your apL software Dirac or room eq ...

That's how I do it and it works. (And set your limits to not allow more than 3db boost. Or however you have to manipulate things so nothing is boosted a shit ton.) even a digital signal has a headroom and is not infinite.

I was having a hell of a hard time getting all my input/output Eqs to sound as harmonically balanced as a single 31band eq and nothing else with a mic measurement from listening position with a basic 30band RTA.

Yeah the old skool method sounded much much more balanced until I learned this routine .
Wow, so you mean you do your XOs with mic in front of the cones/mouths?
That’s interesting, and must be much easier.

Horny Froggy...
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Old 12-05-2016   #374
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Default Re: Comprehensive Review: " APL1 " - Advanced DSP / EQ / Phase Correction Unit

Quote:
Originally Posted by Elgrosso View Post
Wow, so you mean you do your XOs with mic in front of the cones/mouths?
That’s interesting, and must be much easier.
NO, I flatten the responce of the driver with the crossover either off or two oactaves past XO, I only work on the stop band , the pass band I use global eq for... the actual crossover point is determined at listening position by what sounds the best based on stage height depth and overall accuracy and by what measures good and is safe for the driver.


The close mic measurements are strictly for stop band . I find it sounds better to flatten the curve with crossover off/moved away than apply the filter. It will make the acoustic slope match the electrical slope

Before I did the same think except I was using measurements at listening position. That added room issues to measurements so a close mic measurement eliminated that so it didn't mess up the harmonic balance with rest of system. If I kept the old way I wouldn't be "fixing the acoustic slope" it would be fixing the slope with room stuff added. At that point it wasn't truly fixing the slope.

Close mic for adjusting your crossovers with eq. Listening position for gloabal eq. if I do make an adjustment to a drivers passband in driver peq than I add that to all other speakers. But I try to avoid it and just use input eq for making flat FR and speaker eq for fixing slopes. Both different mic locations for different things.

Sorry I don't post much screenshots. Maybe I should . Just time consuming...


im so in love with my speakers I found a song I sing to them

https://youtu.be/VgU6LXY-AjI

Last edited by oabeieo; 12-05-2016 at 01:56 PM..
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Default Re: Comprehensive Review: " APL1 " - Advanced DSP / EQ / Phase Correction Unit

Quote:
Originally Posted by oabeieo View Post
NO, I flatten the responce of the driver with the crossover either off or two oactaves past XO, I only work on the stop band , the pass band I use global eq for... the actual crossover point is determined at listening position by what sounds the best based on stage height depth and overall accuracy and by what measures good and is safe for the driver.


The close mic measurements are strictly for stop band . I find it sounds better to flatten the curve with crossover off/moved away than apply the filter. It will make the acoustic slope match the electrical slope
Ok I think I see, kind of what described subterFUSE then. But mixing nearfield and listening position.
But first, question for confirmation: bandpass = between XO points, stop band = below & Above XO points?

In fact there's a lot of similarities with what I was trying to do with Dirac recently (and APL before).
Meaning focusing PEQ+XO to get to the acoustic slope, on the stop & transitions band, and leave the FIR device focusing on the bandpass.
Well with these actual words it seems much simpler.

But I never used near-field measurements for anything else than just checking the driver.
I'll have to give it a go, it will simplify the process, less steps, less time.
(yesterday with all these sweeps on such a large area I got lost at one time, what to EQ what not)



Quote:
Originally Posted by oabeieo View Post
Before I did the same think except I was using measurements at listening position. That added room issues to measurements so a close mic measurement eliminated that so it didn't mess up the harmonic balance with rest of system. If I kept the old way I wouldn't be "fixing the acoustic slope" it would be fixing the slope with room stuff added. At that point it wasn't truly fixing the slope.

Close mic for adjusting your crossovers with eq. Listening position for gloabal eq. if I do make an adjustment to a drivers passband in driver peq than I add that to all other speakers. But I try to avoid it and just use input eq for making flat FR and speaker eq for fixing slopes. Both different mic locations for different things.

Sorry I don't post much screenshots. Maybe I should . Just time consuming...

How close do you measure?
And how do you do for the HF drivers? (for flat)

Also I must say, I'm getting confused.
- after all these readings from Andy on how we should EQ the whole system (drivers+car reflections).
Yet here you're removing some of these effects of the equation.
- but in the same time I can see some value, maybe just trade-offs or for better balance like you said.
- and I'm close to think that these points were the ones blocking me before.

Horny Froggy...
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