# The new 100$ FIR dsp



## oabeieo (Feb 22, 2015)

For 195$
I just got one. I will see how she does with fir and 
Rephase. 

It also is a normal 2x4 dsp on top of it . 

Pretty awesome and affordable. 

Now we all can have fir . Make us a dolby lake for 100$


Anyone else getting one? 


https://www.minidsp.com/products/minidspkits/2-x-in-4-x-out-hd


----------



## oabeieo (Feb 22, 2015)

Which brings up a question, 

If I'm running FIR on 2 in 4out and have 4 other channels going directly into amps not using FIR or minidsp will the subsequent delay from the FIR processing have a effect on the channels that don't have this? 

I guess what I'm asking is ;

FIR has taps and takes tons of processing power to compensate for the delay that would be inherited from the re phasing at lower frequencies, I'm sure the minidspHD compensates for its own outputs , but will it be in "time" with other channels that aren't running FIR? 

Thank you, 
Cheers


----------



## sicride (Oct 26, 2014)

8-/\/\/\/\D Phase response of old forum boner processors.

8=====D phase response of new forum boner.

MiniDSP has nailed the technology part of signal processing and saves consumers a lot of money by NOT creating software that makes it easily accessible. Remember when they came out with windows, so you didn't have to be a freaking genius to use a computer.... Yeah, when they have point and click software, I may be smart enough to use this.


----------



## unix_usr (Dec 4, 2013)

sicride said:


> 8-/\/\/\/\D Phase response of old forum boner processors.
> 
> 8=====D phase response of new forum boner.
> 
> MiniDSP has nailed the technology part of signal processing and saves consumers a lot of money by NOT creating software that makes it easily accessible. Remember when they came out with windows, so you didn't have to be a freaking genius to use a computer.... Yeah, when they have point and click software, I may be smart enough to use this.


HUH? - miniDSP uses Windoze or Mac to configure -> https://www.minidsp.com/applications/dsp-basics/minidsp-concept Have you ever used/tried one?


----------



## brumledb (Feb 2, 2015)

I am wondering about using two of these. One for left channel and one for right channel. I would be tying into an OEM system. So would go--JL Fix 82 (has two analog outputs)-- left channel to one mini and right to one mini-- amps

Or I could use the toslink out of Fix82--splitter--toslink into both mini's

One thing that is a bit cumbersome is have to use the minidc isolator. The problem is it doesn't come in a housing so have to fashion one somehow. I also need to find out if I could use one minidc for both minidsp's or if each DSP would need its own minidc.


Sent from my iPhone using Tapatalk


----------



## sicride (Oct 26, 2014)

Nope haven't actually used one yet but the website always refers to using other software to make adjustments and then export to this or that. So I guess I was mislead. I guess the website could be better then? I dunno it just doesn't sound "easy" to use to me. I thought I saw a photo of a user interface on one of the pages but then they kept talking about someone else's software.

I guess what bugs me is I WANT to use this hardware but I don't want to spend that kind of money on something I have to build a box around and still not know if I'll even be competent to use it and have it end up as a paper weight.


----------



## brumledb (Feb 2, 2015)

sicride said:


> Nope haven't actually used one yet but the website always refers to using other software to make adjustments and then export to this or that. So I guess I was mislead. I guess the website could be better then? I dunno it just doesn't sound "easy" to use to me. I thought I saw a photo of a user interface on one of the pages but then they kept talking about someone else's software.
> 
> I guess what bugs me is I WANT to use this hardware but I don't want to spend that kind of money on something I have to build a box around and still not know if I'll even be competent to use it and have it end up as a paper weight.


For the general eq'ing and time alignment there is no extra software needed. It is already built in just like every other DSP. If you look at the user manual, it gives a good explanation of how to use it.

I think the "other" software you are referring to is rePhase FIR Tool. It is the tool needed to properly implement the FIR filters. Minidsp has a rePhase tutorial. 

Oh, and the only part that would need a box for it would be the miniDC. They make a DSP that is car specific that doesn't require the additional miniDC but it doesn't have the FIR capabilities. 

rePhase Fir Tool


----------



## Hanatsu (Nov 9, 2010)

Looks interesting! I might get one.


----------



## Hanatsu (Nov 9, 2010)

sicride said:


> Nope haven't actually used one yet but the website always refers to using other software to make adjustments and then export to this or that. So I guess I was mislead. I guess the website could be better then? I dunno it just doesn't sound "easy" to use to me. I thought I saw a photo of a user interface on one of the pages but then they kept talking about someone else's software.
> 
> I guess what bugs me is I WANT to use this hardware but I don't want to spend that kind of money on something I have to build a box around and still not know if I'll even be competent to use it and have it end up as a paper weight.


MiniDSP got an awesome user interface...


----------



## thehatedguy (May 4, 2007)

You sure you can run rephase on that? I don't think you can.


----------



## Hanatsu (Nov 9, 2010)

thehatedguy said:


> You sure you can run rephase on that? I don't think you can.





> The filter coefficient file loaded in File Mode uses IEEE 754 single-precision binary floating-point format. The number of entries in the file must not exceed the allocated number of taps


rePhase can output to that format.


----------



## thehatedguy (May 4, 2007)

But the miniDSP by itself can't run rephrase and doesn't do FIR crossovers or EQ. Right now as far as functioning, this processor uses the same plug in and thus the same features as the regular/older minidsp.


----------



## 1fishman (Dec 22, 2012)

Subed


----------



## brumledb (Feb 2, 2015)

"This tiny powerhouse is jam-packed with miniDSP’s tried and proven audio processing functionality: large FIR banks, flexible parametric EQ, Butterworth and Linkwitz-Riley crossovers, advanced biquad programming, and delay on each output channel. miniDSP’s “one hardware many plugins” concept allows for many interesting future applications!"


----------



## Hanatsu (Nov 9, 2010)

thehatedguy said:


> But the miniDSP by itself can't run rephrase and doesn't do FIR crossovers or EQ. Right now as far as functioning, this processor uses the same plug in and thus the same features as the regular/older minidsp.


I don't understand you now... The MiniDSP GUI don't run rephase by itself but we should be able to upload from rePhase to this unit, it certainly seems so.


----------



## brumledb (Feb 2, 2015)

Hanatsu said:


> I don't understand you now... The MiniDSP GUI don't run rephase by itself but we should be able to upload from rePhase to this unit, it certainly seems so.


Yeah, I mean that is what is says in the user manual.


----------



## brumledb (Feb 2, 2015)

5.5.1 FIR filtering overview

FIR ("finite impulse response") filtering differs from the IIR ("infinite impulse response") filters used in the PEQ
and crossover blocks. Technically speaking, IIR filters are recursive, meaning that each output value is partially
calculated from earlier output values as well as from input values. An FIR filter is specified by a large array of
numbers, whereas an IIR filter requires only a fairly small of values to be specified.
These numbers are conventionally referred to as "taps." The 2x4 HD can compute a total of 4096 taps.
These taps can be distributed as you wish across the four output channels, with the limitation that each output
channel must have 6 or more taps and can have no more than 2048 taps. The decision on how many taps to
allocate to each channel up to you, and should be determined after working with an FIR filter design program
(see below). The number of taps is set in the lower right corner (click on the text entry box and type the desired
number of taps, then press Tab or Return):

5.5.2 FIR filter design software

The filter coefficients must be created with the aid of filter design software. miniDSP does not provide any such
software, instead referring you to the many software packages available for this purpose (both freeware and
commercial). Please see the FIR filter tools page on our website.

5.5.3 Filter file format

The filter coefficient file loaded in File Mode uses IEEE 754 single-precision binary floating-point format. The
number of entries in the file must not exceed the allocated number of taps.
In Manual Mode, the coefficients must be plain text in this format:
b0 = 1,
b1 = -1,
b2 = 0.5,
b3 = -0.5,
b4 = 0.2,
b5 = 1,
miniDSP Ltd, Hong Kong / www.minidsp.com / Features and specifications subject to change without prior notice 34

5.5.4 Loading filter coefficients

In File Mode:
1. Click Browse, navigate to the file containing the filter coefficients, and open it. A dialog will appear
confirming the number of coefficients loaded.
2. Confirm that the response curve is as you expect.
3. Press Send to DSP. This will write the coefficients into the DSP's memory.
4. To clear the filter coefficients, click Unload FIR and then Send to DSP.
In Manual Mode:
1. Cut and paste the coefficients from the text output of the design program.
2. Press the Process button.
3. Confirm that the frequency response graph is as you expect.
4. Press Send to DSP. This will write the coefficients into the DSP's memory.
5. To clear the filter coefficients, click Clear Taps and then Send to DSP.


----------



## thehatedguy (May 4, 2007)

Where and how are you going to get the rephrase data to input manually to the minidsp?


----------



## thehatedguy (May 4, 2007)

Pretty sure you have to at this time use a miniSHARC as a convolver engine to run the rephrase software.


----------



## Hanatsu (Nov 9, 2010)

Input manually? Export the bin file from rePhase and load it in the 'FIR' tab in the minidsp software. I don't see the problem here...


Sent from my iPhone 6 using Tapatalk.


----------



## brumledb (Feb 2, 2015)

The *upgraded* on-board 400 MHz *Analog Devices SHARC processor* also enables substantial processing upgrades previously available only on more expensive platforms, such as 96 kHz internal processing for true high-resolution audio capability and *assignable FIR filter taps for sophisticated equalization*, crossover, and *room correction capabilities*. *All to be accessed and programmed with miniDSP’s easy-to-use interface software.*


----------



## thehatedguy (May 4, 2007)

Thank again for copy and pasting.


----------



## brumledb (Feb 2, 2015)

thehatedguy said:


> Thank again for copy and pasting.


You are welcome. It just doesn't seem like you are reading the same material I am and I surely can't explain it better than the manual does. I will send a message to minidsp to make sure it can be used with rePhase without any other equipment.

If it can't be, then they surely did a piss-poor job of making that clear.


----------



## thehatedguy (May 4, 2007)

That processor will run out of taps before you will get everything programmed. The home guys are using multiple openSHARCs and DACs to get a multiway system that will do FIR filtering in the bass.


----------



## brumledb (Feb 2, 2015)

Just comparing it to the APL1, the APL1 has 4096 taps per channel whereas the 2x4HD only has 4096 total for a combined 4 channels. But the taps can be allocated however you want with a few stipulations. The max taps you can use on any one channel is 2048.

*Hanatsu*, If you read this, how many taps do you typically use per channel with the APL1?


----------



## oabeieo (Feb 22, 2015)

thehatedguy said:


> You sure you can run rephase on that? I don't think you can.


Yes, I just got mine it's a shark and a 2x4 advanced in one. 

Rephase is quite simple to use and can import right from REW. 

It's pretty dope. It makes a GINORMOUS diffrance where the problem are I was having. I need to take time to really get into it but it really is a wet dream for a avid tuner.


----------



## oabeieo (Feb 22, 2015)

You basicly import your fir file from rephase into the fir bank, what sucks is its not real time, well I've only played with it about 20min so far, still tons to learn. But as far as taps go, who care about bass in a car, I shouldn't say that, but the real problems are in the midbass and midrange.

It does compensate for delay time coming out of the unit from its fir. So using the mini on part of the system and your dsp on the other part of the system works with no ill effects. 

It has 2v out so there's no noise issues and can take 4v in, I have mine on 2v in setting , has less noise( go figure) I think the 4v setting is for a CD player or something that is constantly putting out the max signal . 

I just ordered a 2nd one so I can go for on all channels, 
My minidc runs this and the mini2x4 at same time no problem.


----------



## Theslaking (Oct 8, 2013)

brumledb said:


> One thing that is a bit cumbersome is have to use the minidc isolator. The problem is it doesn't come in a housing so have to fashion one somehow. I also need to find out if I could use one minidc for both minidsp's or if each DSP would need its own minidc.
> 
> 
> Sent from my iPhone using Tapatalk


You can. I run a left and right in my work truck off one minidc.


----------



## oabeieo (Feb 22, 2015)

You think when it conpensiate for delay is would slow down everything to the point of how far off a delay is from amplitude differences however it speed up the delays part so the delayed wavelength is sped up to the real time input. So crossovers have be done with fir to get that benefit and so does convolution. For phase eq if you like the crossover delay but just need a fix in one area after/before crossover because of path leinght diffrance. 

getting the basic helped me understand how to approach the controls. Can add delay as well to other parts.

I like what Hanatsu is doing in his thread. He lays it out pretty good. 

I'm pretty glad I got this vs the apl1 even tho it's probably better. The price is good and I'm learning and tuning and fining the problems out on my own. I still want a apl1 tho. 
I was super close to getting one. Just the cost was tough for something that can easily make things worce


----------



## Hanatsu (Nov 9, 2010)

brumledb said:


> Just comparing it to the APL1, the APL1 has 4096 taps per channel whereas the 2x4HD only has 4096 total for a combined 4 channels. But the taps can be allocated however you want with a few stipulations. The max taps you can use on any one channel is 2048.
> 
> *Hanatsu*, If you read this, how many taps do you typically use per channel with the APL1?



4096taps each channel. This can't be compared with the APL system in this manner. rePhase is primarily for low frequency corrections.


Sent from my iPhone 6 using Tapatalk.


----------



## 14642 (May 19, 2008)

Pretty cool. Considering the reflective environment in a car and the lack of annoying effects of phase errors above 1k, this ought to be pretty effective in improving phase differences between channels in a stereo system with an offseat listener. My suggestion would be to build an FIR filter that includes phase EQ only below 1k. In REW, stitch together a spatially averaged measurement above 1k and a single measurement below. In the text file for the resulting measurement, you may need to manually insert phase information above 1k. Just set that to 0 degrees or find a way to generate a minimum phase curve for the spatially averaged curve before you stitch them together.

If you do this and DON'T implement any left and right channel delay in the system, it may be a simple matter to have a two seat stereo car without a center speaker and with symmetrical imaging in both seat. Pay careful attention to phase at and around 250Hz.


----------



## Hanatsu (Nov 9, 2010)

I ordered one. This HD version features compression and delay up to 80ms. I'll be using 2 normal 2x4 units and one 2x4 hd. Will be interesting to see what it can do


Sent from my iPhone 6 using Tapatalk.


----------



## oabeieo (Feb 22, 2015)

Andy Wehmeyer said:


> Pretty cool. Considering the reflective environment in a car and the lack of annoying effects of phase errors above 1k, this ought to be pretty effective in improving phase differences between channels in a stereo system with an offseat listener. My suggestion would be to build an FIR filter that includes phase EQ only below 1k. In REW, stitch together a spatially averaged measurement above 1k and a single measurement below. In the text file for the resulting measurement, you may need to manually insert phase information above 1k. Just set that to 0 degrees or find a way to generate a minimum phase curve for the spatially averaged curve before you stitch them together.
> 
> If you do this and DON'T implement any left and right channel delay in the system, it may be a simple matter to have a two seat stereo car without a center speaker and with symmetrical imaging in both seat. Pay careful attention to phase at and around 250Hz.


Andy, I've literally been trying to figure out a good place to start. I have 1fir bank in use because I don't know what the **** I'm doing. 

This is invaluable, I can't wait to get home to try it. 

It's on the line of what I was thinking to use it for. I've got pink noise sounding the same tonaly to my ear better with it , but I know I can make it much better.

Thank you, 
Andy 

And yes 250hz is a big problem area I have discovered it does act funny .


----------



## mitchyz250f (May 14, 2005)

Andy Wehmeyer said:


> Pretty cool. Considering the reflective environment in a car and the lack of annoying effects of phase errors above 1k, this ought to be pretty effective in improving phase differences between channels in a stereo system with an off seat listener. My suggestion would be to build an FIR filter that includes phase EQ only below 1k. In REW, stitch together a spatially averaged measurement above 1k and a single measurement below. In the text file for the resulting measurement, you may need to manually insert phase information above 1k. Just set that to 0 degrees or find a way to generate a minimum phase curve for the spatially averaged curve before you stitch them together.
> 
> If you do this and DON'T implement any left and right channel delay in the system, it may be a simple matter to have a two seat stereo car without a center speaker and with symmetrical imaging in both seat. Pay careful attention to phase at and around 250Hz.



Sorry coming late to the minidsp thing. How does it eliminate the need for a center channel. Can someone dumb this down to a 5th grade level for me. 

Make that third grade.


----------



## oabeieo (Feb 22, 2015)

mitchyz250f said:


> Can someone dumb this down to a 5th grade level for me.
> 
> Make that third grade.


I think he's sayin 
Use two measurements per side, divide the bandwidth from 250hz to 1k

Than have REW allocate taps for each bandwidth into rephase . 

So that it corrects problem areas diffrent above 1k than it does under 1k because of wavelength. The distance between left and right speakers and where you sit in the car will need to be seperatly accounted for to get correct offsets for single seat. 

Goal, a even phase RELITIVE responce from side to side. 

( have you ever listened to pink noise and could tell even tho you eq left and right to have same FR it has a diffrent pitch or tonality ? That would be the phase diffrances , most decks you can rev the phase and MAYBE fix one frequency, but makes other frequency worce. We're trying to balance those differences which phase) 

I think.  not 100% but I'm starting to kinda understand


----------



## oabeieo (Feb 22, 2015)

It eliminates the need because it will have close to perfect phase in one seat . And will "decompress" the TA conundrum


----------



## thehatedguy (May 4, 2007)

Nice. I might have to get one myself if I ever do a 2 channel system.


----------



## Hanatsu (Nov 9, 2010)

You need to be careful how you measure if you gonna use phase corrections up to 1kHz. Single point measurements is inadequate that high. It varies way too much above the modal region ~200-250Hz. Would highly recommend multiple measurements / spatial averaging. I haven't looked in too much how the phase response is averaged by the built in FR averaging function in RoomEQ to comment on that just yet.


----------



## oabeieo (Feb 22, 2015)

It's tough to wrap my brain around all this. Please keep the tips coming. 

Bitte schön!


----------



## mitchyz250f (May 14, 2005)

thehatedguy said:


> Nice. I might have to get one myself if I ever do a 2 channel system.


I thought you could do 5.1 with minidsp


----------



## oabeieo (Feb 22, 2015)

Does anyone know of a good program that makes biquads for minidsp? The one they give is very.... Oh boy... 

I just found out that I can make crossovers with multiple shapes on the roll off . I really want to do that with my audax. 

It would be very cool if there's a 3rd party generator somewhere that is more like software instead of dos. Or whatever the protocol is.


----------



## oabeieo (Feb 22, 2015)

I'm wrong . It does delay the output signal 

Something about being casual. I don't think the decks delay is enough to sync. 

Need two 2x4s


----------



## Hanatsu (Nov 9, 2010)

oabeieo said:


> Does anyone know of a good program that makes biquads for minidsp? The one they give is very.... Oh boy...
> 
> I just found out that I can make crossovers with multiple shapes on the roll off . I really want to do that with my audax.
> 
> It would be very cool if there's a 3rd party generator somewhere that is more like software instead of dos. Or whatever the protocol is.



Easiest is to use their spreadsheet tbh. There are online calculators as well as one inside the dsp software...


Sent from my iPhone 6 using Tapatalk.


----------



## 14642 (May 19, 2008)

To use phase EQ for making both seats sound similar without delay isn't about correcting the phase for each channel so the phase of the channel is flat and 0 degrees, it's about compensating for the RELATIVE phase between the two channels in regions where the two channels approach -180 degrees because that condition breaks the phantom center. It requires a coherent sum. 

We use delay because it's straightforward, but it destroys one seat in the interest of optimizing the other. We say, "in order for there to be a correct playback, the sound of the speakers has to arrive at the same time." Technically, that's correct. Practically, it's unnecessary. 

The biggest problem in creating a phantom center in both seats is that, because of the difference in distance, the speakers are 180 degrees at a frequency where half the wavelength corresponds to the difference in distance and at odd multiples of that frequency. We apply delay to compensate for the distance. That fixes one seat but DOUBLES the apparent distance in the other seat. 

The problem is distance, but what we hear is the relative phase between the channels. What if we just change the phase of one of the speakers at the frequency that corresponds to that difference in phase? In cars and for door mounted speakers, that frequency is about 250 Hz. That also happens to be in the middle of the vocal range in many recordings and the vocalist is often mixed to the center. 

So, fix that phase problem, and the vocalist is back in the center. DO we need to address phase at the first odd multiple? Probably not. We begin not to hear phase very easily at about 1k. By 3k, we don't hear it very well and above 6k, we don't hear it at all. Add the reflective environment of the car and it's such a mess up there that there's no use in attempting any correction. 

Below is a spreadsheet i built to make viewing this simple. The graphs are an indication of the phase DIFFERENCES between L, R and C channels in my car. The phase differences between left and center and right and center at low frequencies are similar. Changing the phase of the center at low frequencies will put all three back in phase. IF you look closely above that at the 300 Hz range, the left and right are out of phase over a wide range and the left and center are also out of phase over a smaller range. Changing the phase of the left channel at 300Hz will improve that problem.

I built this because, despite the upmixer and the center channel, I was having some trouble with near side bias in the midbass/midrange.

Applying two all pass filters, one on the center and one on the left midbass cleared this up. Even if I turn off the center channel, the image still appears in the center in both seats. I also tried a filter on the center at about 1700 Hz, but the effect was inaudible.

Tuning cars isn't always about applying the "correct" fix. It's often about applying one that improves performance under some set of constraints. My constraint is that the passenger's seat can't suck. That changes the appropriate choice of tools considerably. 

So, there aren't any aftermarket DSPs with this capability, but you can do this with an FIR.


----------



## HulkSmash (May 22, 2011)

oabeieo said:


> I'm wrong . It does delay the output signal
> 
> Something about being casual. I don't think the decks delay is enough to sync.
> 
> Need two 2x4s


You're running a 3-way with horns in your car, aren't you? Just reading along, so if you succeed, I have some hope when I finish installing mine hah.


----------



## astrochex (Aug 7, 2009)

Where is this mystery spreadsheet of which you speak? Thanks.


Andy Wehmeyer said:


> .
> .
> .
> *Below is a spreadsheet i built to make viewing this simple.* The graphs are an indication of the phase DIFFERENCES between L, R and C channels in my car. The phase differences between left and center and right and center at low frequencies are similar. Changing the phase of the center at low frequencies will put all three back in phase. IF you look closely above that at the 300 Hz range, the left and right are out of phase over a wide range and the left and center are also out of phase over a smaller range. Changing the phase of the left channel at 300Hz will improve that problem.
> ...


----------



## Hanatsu (Nov 9, 2010)

astrochex said:


> Where is this mystery spreadsheet of which you speak? Thanks.


http://api.viglink.com/api/click?format=go&jsonp=vglnk_14582466060438&key=8f04fe2f499a5c84eb3c7ba7b154cca2&libId=ilwqd4eo01010zyd000DAl131ico0&loc=http%3A%2F%2Fwww.diymobileaudio.com%2Fforum%2Ftechnical-advanced-car-audio-discussion%2F259201-advanced-filters-allpass-lt-etc-minidsp.html&v=1&out=https%3A%2F%2Fwww.minidsp.com%2Fimages%2Ffbfiles%2Ffiles%2FAll_digital_coefs_v1-20101026.zip&ref=http%3A%2F%2Fwww.diymobileaudio.com%2Fforum%2Fsearch.php%3Fsearchid%3D18119890&title=Advanced%20filters%20(Allpass%2FLT%20etc)%20with%20MiniDSP%20-%20Car%20Audio%20%7C%20DiyMobileAudio.com%20%7C%20Car%20Stereo%20Forum&txt=https%3A%2F%2Fwww.minidsp.com%2Fimages%2Ffbfil...1-20101026.zip


----------



## brumledb (Feb 2, 2015)

So what is the major difference(s) between using this vs APL1, from an FIR point of view? I know the APL1 has the "auto-eq" type function and more taps per channel. Are there any other (important) differences between the two?

Are these two units apples to apples, or apples to oranges?


----------



## Hanatsu (Nov 9, 2010)

Andy Wehmeyer said:


> So, there aren't any aftermarket DSPs with this capability, but you can do this with an FIR.


MiniDSP can pull off an APF through a biquad. Don't know if it's adequate for these types of corrections. Tbh I've never tried this approach, would be interesting to experiment with it.

http://www.diymobileaudio.com/forum/technical-advanced-car-audio-discussion/259201-advanced-filters-allpass-lt-etc-minidsp.html


----------



## Hanatsu (Nov 9, 2010)

brumledb said:


> So what is the major difference between using this vs APL1, from an FIR point of view? I know the APL1 has the "auto-eq" type function, more taps per channel. Are there any other (important) differences between the 2?
> 
> Are these two units apples to apples, or apples to oranges?


Well the APL1 is just the hardware part of the APL system, it doesn't "have" auto EQ, it's the software that creates the IR that got this function. The power lies in the software, how it measures and corrects the response. It derives the corrections from the full sound power response. You can't use the fir file format delivered from the software to this unit, therefore you can't reproduce the results in this manner. Also since there's a limit for total taps used, there will be a limit in the low frequency resolution where it's needed the most, that's kind of an issue... 

The APL1 hardware unit features a digital output as well, this one doesn't. APL1 got a 16 preset physical "remote". Both can use filters created in rePhase though. This MiniDSP HD features complete DSP software, APL1 is an equalizer. That's about it I think.


----------



## 14642 (May 19, 2008)

Hanatsu said:


> http://api.viglink.com/api/click?fo...www.minidsp.com/images/fbfil...1-20101026.zip


That's not my spreadsheet, but it's a good one.


----------



## 14642 (May 19, 2008)

Hanatsu said:


> MiniDSP can pull off an APF through a biquad. Don't know if it's adequate for these types of corrections. Tbh I've never tried this approach, would be interesting to experiment with it.
> 
> http://www.diymobileaudio.com/forum...-advanced-filters-allpass-lt-etc-minidsp.html


Yes, that will work. Just calculate the frequency for the all pass by finding the frequency where 1/2 a wavelength corresponds to the pathlength difference. It doesn't even have to be precise. You just need to move the phase difference away from -180 degrees.


----------



## oabeieo (Feb 22, 2015)

Andy Wehmeyer said:


> To use phase EQ for making both seats sound similar without delay isn't about correcting the phase for each channel so the phase of the channel is flat and 0 degrees, it's about compensating for the RELATIVE phase between the two channels in regions where the two channels approach -180 degrees because that condition breaks the phantom center. It requires a coherent sum.
> 
> We use delay because it's straightforward, but it destroys one seat in the interest of optimizing the other. We say, "in order for there to be a correct playback, the sound of the speakers has to arrive at the same time." Technically, that's correct. Practically, it's unnecessary.
> 
> ...


Can I use REW with a square wave sweep to measure my own system like that?


----------



## Hanatsu (Nov 9, 2010)

Andy Wehmeyer said:


> Yes, that will work. Just calculate the frequency for the all pass by finding the frequency where 1/2 a wavelength corresponds to the pathlength difference. It doesn't even have to be precise. You just need to move the phase difference away from -180 degrees.


Cool, I'll give it a try.


----------



## thehatedguy (May 4, 2007)

Used this for the XO biquad programming?

the Active Crossover Designer web page


----------



## thehatedguy (May 4, 2007)

I'm sure you can...I just don't know how.

Plus I have a MS8 for multichannel stuff.



mitchyz250f said:


> I thought you could do 5.1 with minidsp


----------



## 14642 (May 19, 2008)

oabeieo said:


> Can I use REW with a square wave sweep to measure my own system like that?



I used the sine wave sweep and then viewed the left and right phase responses as an overlay after smoothing to 1/6th octave. I wanted something to help me explain this when I do trainings, so I loaded those measurements into a spreadsheet. The reason the graphs are square is because in my spreadsheet there are only two values for the horizontal axis: 0 and 180. Everything above 130 degrees displays as 180 and everything below displays as 0. I arrived at that limit because it indirectly helps me see the appropriate Q value for the all pass filter.


----------



## 14642 (May 19, 2008)

thehatedguy said:


> Used this for the XO biquad programming?
> 
> the Active Crossover Designer web page



I didn't, my DSP includes a parametric all pass filter.


----------



## thehatedguy (May 4, 2007)

I don't know if that spreadsheet that I linked would figure all pass filters.

Another one for XO filters:

A biquad calculator | EarLevel Engineering


----------



## oabeieo (Feb 22, 2015)

Andy Wehmeyer said:


> I used the sine wave sweep and then viewed the left and right phase responses as an overlay after smoothing to 1/6th octave. I wanted something to help me explain this when I do trainings, so I loaded those measurements into a spreadsheet. The reason the graphs are square is because in my spreadsheet there are only two values for the horizontal axis: 0 and 180. Everything above 130 degrees displays as 180 and everything below displays as 0. I arrived at that limit because it indirectly helps me see the appropriate Q value for the all pass filter.


That's brilliant. , so 130 would be similar to something like a RMS value of the which would derive Q. Obviously this is min phase only
What should I be doing with liner phase, just use plain old eq and not worry about it ? That's what you were talking 1k as a stop right? It seems like I can completely do away with the IIR eq for minimum phase to simplify the process yes? 
Should I find the Lin/min reigons, do eq work in the Lin areas first than do the process you described earlier? but also to add fir to xo and eq in the min region using the coefficients sparingly. Doing it the way you describe almost seems like I can just tune the system the best it can be with IIR on Lin areas than do the process. Is the post 1k range process help glass reflection areas vs. modal areas?

I haven't got to the 1k up process yet. Still chewing on get first process (250-1k) right. 

So far I've just linerized the phase left vs right under 1k to 160 where the midpoint on phase swing at 0 was in first measurement. kept the IIR crossovers/eq for whole system , let rephase linerize under 1k for everything (L vs.R). I really like it so far, need to learn more first. Took me all night to just do that. And now have a FR change (post procedure) I need to deal with but don't know quite how so that's another problem I'll work on after learn more

Thank you Andy!


----------



## oabeieo (Feb 22, 2015)

thehatedguy said:


> I don't know if that spreadsheet that I linked would figure all pass filters.
> 
> Another one for XO filters:
> 
> A biquad calculator | EarLevel Engineering


This one seems easy to do. TY


----------



## oabeieo (Feb 22, 2015)

brumledb said:


> I am wondering about using two of these. One for left channel and one for right channel. I would be tying into an OEM system. So would go--JL Fix 82 (has two analog outputs)-- left channel to one mini and right to one mini-- amps
> 
> Or I could use the toslink out of Fix82--splitter--toslink into both mini's
> 
> ...


i need to find out if 2 of them can be connected that way. 

I think theres a way to connect 2 miniHDs together using the header pins. 

Something tells me they need to be joined somehow because the filters are "casual" I guess this means the fir can output delays to the output signal from the input. Depending on the frequency wavelength fir is working depends on how delayed the outputs are from the inputs.

Having one L and one R , or heck even two at all could lead to some serious problems in syncing the output times.

Haven't figured it out yet. I need a 2nd one so I'll keep you posted unless someone else knows answer.


----------



## brumledb (Feb 2, 2015)

oabeieo said:


> i need to find out if 2 of them can be connected that way.
> 
> I think theres a way to connect 2 miniHDs together using the header pins.
> 
> ...


Yeah I would be interested to know how that works. But I think I am going to go with the APL1 when I get ready to purchase. I would want 8 channels of FIR and I just bought a Helix DSP. I wish the software didn't cost so much but such is life.


----------



## oabeieo (Feb 22, 2015)

brumledb said:


> Yeah I would be interested to know how that works. But I think I am going to go with the APL1 when I get ready to purchase. I would want 8 channels of FIR and I just bought a Helix DSP. I wish the software didn't cost so much but such is life.


Apl1 seems superb. But the mini is cheap and can still fix my one problem area that has been annoying me.


----------



## mitchyz250f (May 14, 2005)

Andy for a two seat system with FIR filters setup properly and with mid in a good location at the front/top of the doors is having a center channel:
a- not needed
b- somewhat helpful
c- very helpful
d- still required


----------



## oabeieo (Feb 22, 2015)

mitchyz250f said:


> Andy for a two seat system with FIR filters setup properly and with mid in a good location at the front/top of the doors is having a center channel:
> a- not needed
> b- somewhat helpful
> c- very helpful
> d- still required


Still want andy to answer this, I just to take a guess and see if I was right when andy says what it is.

My guess would be 

A- not needed - but only where it is minimum phase or frequencies that are at or longer than The distance between you and the Speakers . 

Why,
Because if wave leinght is shorter you would have to correct for more than 360° At that point time delay becomes the right tool to use to off set seating. Than phase correct between TA offsets. Meaning a center channel would still be needed for frequencies that are shorter than the distance between you and the speaker.


----------



## brumledb (Feb 2, 2015)

Well Andy really already answered this.



Andy Wehmeyer said:


> . Even if I turn off the center channel, the image still appears in the center in both seats. I also tried a filter on the center at about 1700 Hz, but the effect was inaudible.


----------



## oabeieo (Feb 22, 2015)

brumledb said:


> Well Andy really already answered this.


Andy also said the vocal is in the 250hz range where the lower frequencies are.
That's fixable in both seats 

Still dosent address the highs . , 

So it probably means low vocals don't need a center and a woman screaming top her lungs might need a Bit of center channel to help pull it to the middle.


----------



## Patrick Bateman (Sep 11, 2006)

Andy Wehmeyer said:


> To use phase EQ for making both seats sound similar without delay isn't about correcting the phase for each channel so the phase of the channel is flat and 0 degrees, it's about compensating for the RELATIVE phase between the two channels in regions where the two channels approach -180 degrees because that condition breaks the phantom center. It requires a coherent sum.
> 
> We use delay because it's straightforward, but it destroys one seat in the interest of optimizing the other. We say, "in order for there to be a correct playback, the sound of the speakers has to arrive at the same time." Technically, that's correct. Practically, it's unnecessary.
> 
> ...


It's 12:05am and it's been a looooong week, but let me see if I understand this idea:

In a typical car stereo, we are much closer to the left speaker than the right. For instance, the pathlength to my left speaker is something like 100cm and the pathlength to the right speaker is about 150cm.

If we fix this using DSP delay, we delay the left speaker by 1.5ms (50cm). *But when we do this, we've actually made the delay problem TWICE as bad for the passenger.* So the soundstage at the passenger seat has gone from bad to worse. (Because it was 'off' by 50cm before, and now it's off by 100cm.)

As far as timing goes, the 'maximum error' will occur when the speakers are out of phase by 180 degrees. For instance, 340hz is 100 centimeters long. That means that *when my left and right speakers are separated by 50cm, they'll be 180 degrees out of phase at 340hz.*

This means that you could probably create a superior delay filter:
A delay filter that adds 1.5ms of delay... At a single frequency.
This would look similar to a bandpass crossover filter. Except there would be no affect on the frequency response. All you would have is a delay. The frequency response would be untouched, but the DELAY would vary with frequency, and the maximum delay would be centered on the frequency where the left and the right speaker are out-of-phase with each other.



It is not clear to me how this would turn this into a "two seat car."

It is definitely less processing; instead of delaying the entire frequency range you're only delaying small fractions of it. But the speaker on the passenger side is still too close.

One thing which IS interesting, is that it appears that this filter would correct the soundstage for the driver. And while it wouldn't make the phase 100% correct for the passenger, it *would* be in-phase, but delayed by one wavelength.

Does that make sense?

What I mean by this is that Andy's filter would fix the soundstage for the driver. But since the filter is only targeted at a single frequency (the frequency where the left and the right are 180 degrees out of phase), it would have minimal impact on the sound of the passenger side speaker.

But it's interesting to realize that the filter which puts the DRIVER side speaker into phase, also puts the PASSENGER side speaker into phase... but delayed by one wavelength.

And realistically, it's probably better to have the sound in-phase and delayed than out-of-phase. When sound is out-of-phase, the soundstage gets huge and hazy.


----------



## Patrick Bateman (Sep 11, 2006)

btw, I think you can implement Andy's filter using a plain ol' mini DSP.

Hanatsu posted some info on how you can create filters in mini DSP which don't change the frequency response but they DO change the phase response.

So you could do some strange stuff, like create a filter that delays the driver side speaker by 1.5ms, but center it at 340hz, and don't change the frequency response at all. This could e especially powerful if you're using passive crossovers. You could basically filter the frequency response with the passive xover, and use the DSP to correct the phase response.

Hmmm, you could use this to fix the phase response of vented boxes couldn't you? Just create an all-pass filter, and set it so that it flips the phase of the speaker at the vented tuning frequency. So you'd get the output of a vented box and the phase response of a sealed box. In a vented box, the cone of the loudspeaker isn't moving at the tuning frequency, so there's no reason you couldn't 'flip the polarity' electronically with DSP. Hmmmm...


----------



## oabeieo (Feb 22, 2015)

*It turns into a two seat car because*

At lower frequencies where vocals that area is "fixable" for both seats.

Because the low frequency (250hz to 600hz) *wavelength * is long enough to correct in both seats.

If a wavelength is 3' long and the is a PLD of 1.5' between sides. Let's just say, the passanger side speaker is - 180° to the passanger and - 90° for the driver at a frequency that is 3' long. You change its phase so the speaker output a responce the makes the diffrance for driver and passanger to be in the middle of the wavelength. That way they are only a A few degrees diffrent instead of completely out of phase. 

Now it's just a phase for all frequencies where wavelength is long enough for driver and passenger to be A mode that supports positive waveform.

It only works at longer frequencies and as frequency climbs the inability to equalize between both seats gets more and more impossible.


----------



## sicride (Oct 26, 2014)

Compared to you guys I am quite simple, but I understand it much like Patrick. It sounds like using this technique you could get everything into correct "timing" at all frequencies for the driver, and correct for phase in the passenger seat but slightly delayed. So it is a matter of compromise. This technique will improve sound for all passengers, somewhat at the expense of the drivers sound where things are sometimes going to be "good enough".

I think this is all a matter of opinion and desire as to which technique is best. I respect and admire those who strive for a two seat setup, but I honestly almost never have someone in the car with me who gives enough of a crap about sound to even notice these changes if I tried to point them out. Or at least doesn't pay enough attention to realize they appreciate it. So it sounds like there is a compromise available where one could use time alignment to correct sound for a listening position, then at certain frequencies adjust phase for an improvement for an alternate listening position in ways that doesn't hurt the original position much?

If that is so, I honestly don't think it is worth my time and effort, but a more considerate person may think otherwise.


----------



## oabeieo (Feb 22, 2015)

sicride said:


> Compared to you guys I am quite simple, but I understand it much like Patrick. It sounds like using this technique you could get everything into correct "timing" at all frequencies for the driver, and correct for phase in the passenger seat but slightly delayed. So it is a matter of compromise. This technique will improve sound for all passengers, somewhat at the expense of the drivers sound where things are sometimes going to be "good enough".
> 
> I think this is all a matter of opinion and desire as to which technique is best. I respect and admire those who strive for a two seat setup, but I honestly almost never have someone in the car with me who gives enough of a crap about sound to even notice these changes if I tried to point them out. Or at least doesn't pay enough attention to realize they appreciate it. So it sounds like there is a compromise available where one could use time alignment to correct sound for a listening position, then at certain frequencies adjust phase for an improvement for an alternate listening position in ways that doesn't hurt the original position much?
> 
> If that is so, I honestly don't think it is worth my time and effort, but a more considerate person may think otherwise.


Well we were talking about what andy said originally about being able to get both seats incorrect phase. 

Yes you can optimize it for one seat and to do it in a way that is better than time alignment alone . How about fixing the phase between your mid range and your midbass ever thought about doing that and that's basically what I'm going after in part of my system. I'm not aligning mine completely in the way that Andy described , I think he was just making a point that it can be done .

But yes if you really set it up for one seat things will really start to get good .

Time alignment only aligns one frequency we always choose the vocal area because we want the vocal centered, that compresses the left side at all other frequencies so imagine having your time alignment set to have a center Image , but having the tonality of no time alignment . Tell me that's the golden goose


----------



## oabeieo (Feb 22, 2015)

Patrick Bateman said:


> btw, I think you can implement Andy's filter using a plain ol' mini DSP.
> 
> Hanatsu posted some info on how you can create filters in mini DSP which don't change the frequency response but they DO change the phase response.
> 
> ...


 Yes exactly imagine that vented box having no group delay between the front of the speaker and the port .

It would require some very long taps, so the output on the DSP would be delayed substantially which would require some syncing if you didn't have the same fir bank on other channels


----------



## redit (Jan 14, 2012)

oabeieo said:


> Yes exactly imagine that vented box having no group delay between the front of the speaker and the port .


Wouldn't the phase of the port change as soon as you change the phase of the cone?


----------



## 14642 (May 19, 2008)

mitchyz250f said:


> Andy for a two seat system with FIR filters setup properly and with mid in a good location at the front/top of the doors is having a center channel:
> a- not needed
> b- somewhat helpful
> c- very helpful
> d- still required



I think i'd still chose somewhat or very helpful. The majority of the imaging in the vocal range can be fixed with the single all pass filter. The rest will still benefit from the center and appropriate steering.


----------



## 14642 (May 19, 2008)

Patrick Bateman said:


> It's 12:05am and it's been a looooong week, but let me see if I understand this idea:
> 
> In a typical car stereo, we are much closer to the left speaker than the right. For instance, the pathlength to my left speaker is something like 100cm and the pathlength to the right speaker is about 150cm.
> 
> ...


Right. It's a simple all pass filter. Is it a perfect fix? No. Does it dramatically improve the performance in both seats? Yup. Is this new? No. Harman owns a patent on a very specific way to do this. Is there a workaround? Yes. Listen to the Bose system in a Fiat 500. Will it win IASCA? No. Does it image in the driver's seat? Yeah. Does it image in the passenger's seat? Yeah. Is it a knockout in either seat? No.


----------



## 14642 (May 19, 2008)

My suggestion in thinking about these kinds of tools and the improvements they can provide is to focus on improving things that you hear that suck rather than on some idea of flat phase across the spectrum. 

For every wiggle in an acoustic measurement of frequency response, phase changes direction. In the graph I've posted here, there are five frequency response measurements and a spatial average of those five. The spatial average is black. You can clearly see three regions. The low frequency region below 220 Hz is where the 5 mic placements all measure the same thing (the mic was rotated in a 7" circle around the driver's head position). The next region from 220 Hz to 1 kHz is one in which there is big variance between the measurements but the Q of the peaks and dips is mostly pretty low. Then, at about 1k there's another change to a region where the Q of te peaks and dips is really high and there's a big variance between measurements. 

If you think about how we hear phase, you'll see that these regions conveniently correspond roughly to our ability to hear phase and use it to determine the apparent location of a sound in the lateral plane. IN terms of imaging and phase, below 1k and above 80 Hz, phase matters a lot. That's where I suggest focusing on phase EQ to improve imaging. Around the sub to midbass crossover (80 Hz or so) getting the phase right between the sub and fronts really helps to provide a sense of bass in the front of the car. At frequencies above 1k, phase is a mess but we don't hear it easily. 

So, if you care about the performance in both seats, STOP thinking about this in terms of arrival time--the distances from the speakers to the listener are VERY short--certainly short enough for our brains to assimilate them into a single arrival--and start thinking about them in terms of RELATIVE PHASE. In a car, a few carefully placed all pass filters are a much better solution than delay IF YOU CARE ABOUT BOTH SEATS.

If you only care about one seat, then...as you were...


----------



## oabeieo (Feb 22, 2015)

redit said:


> Wouldn't the phase of the port change as soon as you change the phase of the cone?


No, when the port is at resonance , all of the sound comes from the port and very very little comes from the front of the speaker if you change the speakers phase so that the sound from the port moves ahead in phase that change will be distributed to the system. 

So the phase change would be at the Tuning frequency of the port. And not above that where the speaker plays the dominant info


----------



## sicride (Oct 26, 2014)

We are so very lucky to have you on this forum Andy. You're a pleasure and incredibly helpful.

I guess my confusion comes from adjusting phase to be in phase for one seat, isn't it automatically not in phase for the passenger seat? Or are we saying that we have to make a compromise and have both be somewhat out of phase but better than they were? I can understand that at some frequencies you can have it be in phase for one seat and 360* out of phase for the passenger which will essentially sound like it is in phase.

I sure hope your processor comes along soon. I know there are a ton of people anxious about it's capabilities. We can only dream that you will implement all pass filters or FIR taps accessible to the end user, so we can learn more by doing. From what I understand some of this may have been in play with the MS-8 but access to tweaking it was not.


----------



## redit (Jan 14, 2012)

oabeieo said:


> No, when the port is at resonance , all of the sound comes from the port and very very little comes from the front of the speaker if you change the speakers phase so that the sound from the port moves ahead in phase that change will be distributed to the system.
> 
> So the phase change would be at the Tuning frequency of the port. And not above that where the speaker plays the dominant info


Ahhh, copy that, thanks oabeieo.


----------



## oabeieo (Feb 22, 2015)

*THANKYOU ANDY FOR YOUR TIME*

I've think I got it now! It's sick . 

I'll try post measurements soon.


----------



## oabeieo (Feb 22, 2015)

Another way to look at what andy was saying about both seats.


*LETS PLAY WOULD YOU RATHER*       
( we're talking now about door mounted midbass)

1. Would you rather have the left speaker 180° and the right speaker at 0° If your the driver seat. And the passanger seat have it mirrored where left speaker is at 0° and right speaker is at 180° 
*OR*
2. Would you rather have both speaker at 90° For both seats.

That should have triggered the concept.
At frequencies where the wavelength is long enough to shift the phase so both speakers are closer to each other in relitive polarity .

Yes as frequency rises it will be harder and harder to maintain both sides being shifted to our hypothetical 90° . One would be (let's say for argument) 130° while the other might be 200°. A 60° flip is better than a 180flip. You won't notice 60° As much as you'll notice 180° .And as frequency rises the less you are able to compensate simply because the wavelength is too short and the responce becomes unmanageable.

Please correct me if I'm wrong- but this makes sense and it's the way I understand it and the way it measures in my car with miniHD and rephase......so far


----------



## Hanatsu (Nov 9, 2010)

Patrick Bateman said:


> Hmmm, you could use this to fix the phase response of vented boxes couldn't you? Just create an all-pass filter, and set it so that it flips the phase of the speaker at the vented tuning frequency. So you'd get the output of a vented box and the phase response of a sealed box. In a vented box, the cone of the loudspeaker isn't moving at the tuning frequency, so there's no reason you couldn't 'flip the polarity' electronically with DSP. Hmmmm...


http://www.dafx14.fau.de/papers/dafx14_stephan_herzog_low_frequency_group_delay.pdf


----------



## 14642 (May 19, 2008)

You want them in phase with each other. Relative phase is what matters, not the absolute phase.


----------



## thehatedguy (May 4, 2007)

For the guys playing with the filters, are you using even or odd order all pass filters? Just curious. My old Rane RPM88 had the option for both...wish I still had that guy right about now.


----------



## Patrick Bateman (Sep 11, 2006)

Andy Wehmeyer said:


> You want them in phase with each other. Relative phase is what matters, not the absolute phase.


You've heard the Beolab 90s right?
If you want to see what can be done when you go crazy with FIR filters, that's the one.
Best loudspeaker I've ever heard.


----------



## mitchyz250f (May 14, 2005)

Andy Wehmeyer said:


> For every wiggle in an acoustic measurement of frequency response, phase changes direction. In the graph I've posted here, there are five frequency response measurements and a spatial average of those five. The spatial average is black. You can clearly see three regions. The low frequency region below 220 Hz is where the 5 mic placements all measure the same thing (the mic was rotated in a 7" circle around the driver's head position). The next region from 220 Hz to 1 kHz is one in which there is big variance between the measurements but the Q of the peaks and dips is mostly pretty low. Then, at about 1k there's another change to a region where the Q of te peaks and dips is really high and there's a big variance between measurements.


Andy I think I found this in a Harman AES paper somewhere. Where is your 5th measurement?


----------



## 14642 (May 19, 2008)

mitchyz250f said:


> Andy I think I found this in a Harman AES paper somewhere. Where is your 5th measurement?


I just rotate the mic manually in a circle and take a measurement in each position.


----------



## Patrick Bateman (Sep 11, 2006)

Andy Wehmeyer said:


> I just rotate the mic manually in a circle and take a measurement in each position.


Gary has a neat setup with an array of mics.

If I had more time I'd setup some type of jig so that I could switch my mics remotely. (When I measure my car, I use remote access software to control the computer in my garage remotely. That way I can measure the car using a desktop computer, but *run* the mesurements from a laptop. This saves me the hassle of plugging in microphones and preamps to the laptop.)

But at the moment, I'm still moving the mic the old fashioned way, by hand.


----------



## oabeieo (Feb 22, 2015)

Andy Wehmeyer said:


> I just rotate the mic manually in a circle and take a measurement in each position.


For one seat tuning , (I could give a crap about what the passanger listens to)

Should I average the measurements for left and right the same way or should I average around my left ear area for the left speaker and my right ear area for the right speaker? Or should I do my whole head circumference for both ?

Thanks , 

Also, is there anything in REW that I should do to prepare the data before I send it into rephase? (Smoothing...or does it import the same)

Thanks A gain


----------



## Hanatsu (Nov 9, 2010)

Up to 1kHz I'd say 1/6 oct, should be good enough for single point measurements.

I don't understand how you would do this with multiple measurements. RoomEQ removes phase data when averaging. I don't think moving the mic during a sweep is a good idea. You can choose multiple sweep in the measure tab, haven't tested how the software averages the phase response or if it does it at all doing it that way...


----------



## oabeieo (Feb 22, 2015)

Hanatsu said:


> Up to 1kHz I'd say 1/6 oct, should be good enough for single point measurements.
> 
> I don't understand how you would do this with multiple measurements. RoomEQ removes phase data when averaging. I don't think moving the mic during a sweep is a good idea. You can choose multiple sweep in the measure tab, haven't tested how the software averages the phase response or if it does it at all doing it that way...


Okay that solves it. I haven't been able to average phase . Just FR . 
Been trying to figure that out. So it's not supposed to at all than yeah? 
Thanks Dood


----------



## 14642 (May 19, 2008)

Hanatsu said:


> Up to 1kHz I'd say 1/6 oct, should be good enough for single point measurements.
> 
> I don't understand how you would do this with multiple measurements. RoomEQ removes phase data when averaging. I don't think moving the mic during a sweep is a good idea. You can choose multiple sweep in the measure tab, haven't tested how the software averages the phase response or if it does it at all doing it that way...



A spatial average doesn't include phase. That's why I suggest a single point measurement for fixing stuff below 1k. Above 1k, phase isn't really a big deal and a spatial average up there is a better representation of what we hear than a spectral average.


----------



## oabeieo (Feb 22, 2015)

So 1k works great for me because subsequently my horns are crossed at 1k so it works good for me to manage all dsp channels .

My question I'm dying to know is how should I manage the 2 sets of speakers that play under 1k

So the midrange plays to 500 and the midbass plays to 80-500 

In regards to the phase relationship not from LtoR but from midrange to midbass. 

Should I try to fix the phase issues at crossover first between the two drivers on each side than do the overall side to side?

At crossover because the drivers are not very close together there is some phase issues. The midrange is in a pod and the pod has ill effects on min phase as well. 

Or should I just import the measurements the same in all channels because the measurement caught all together anyway.

When I take measurements I turn off crossovers on both midbass and midrange as directed to get the responce.

Should I be doing the midrange seperate from the midbass? 

Thank you


----------



## thehatedguy (May 4, 2007)

Just curious...if I wanted to play around with this in a multichannel system using the MS8, I would need to place the processor after the MS8 on the pair of channels doing the fronts and use the crossovers in the minidsp for the fronts?

That is assuming either active 2 way fronts or passive between the midranges/tweeter and active between them and the midbass...full active would require another minidsp and more amp channels.


----------



## oabeieo (Feb 22, 2015)

thehatedguy said:


> Just curious...if I wanted to play around with this in a multichannel system using the MS8, I would need to place the processor after the MS8 on the pair of channels doing the fronts and use the crossovers in the minidsp for the fronts?
> 
> That is assuming either active 2 way fronts or passive between the midranges/tweeter and active between them and the midbass...full active would require another minidsp and more amp channels.


That would be a andy question,

But I've installed several ms8s as I'm a dealer for JBL .

And I don't know for sure but I think by the way it responds to the setup with headphones vs outcome is this;

It seems the ms8 ignores frequencies I its auto tune if the frequency is blow crossover (HPF) . So that way it dosent try to boost the crap out of the signal of its told there's a crossover . If that's the case you would still want to use the crossovers in ms8 and let the mini just linerize the responce . 

You'll have to take some raw measurements tho of each speaker un crossed over. Don't know how that works with a ms8 inline - maybe some trial and error. 

My gut says hook it up before ms8 but my brain says some manual tuning with phase eq after ms8 could yield a better tune.

That's a really good question tho, dosent ms8 already have some finite impulse artifacts In it already ?


----------



## oabeieo (Feb 22, 2015)

*Just got my 2nd miniHD woot woot !! Time to put my smock on and lab vest and get to work and bury myself in phase and buttons and controls  *


----------



## mitchyz250f (May 14, 2005)

Jason just curios as to why you use the MS8 when you have the knowledge and tuning experience to use dsp's that offer more flexibility. I think the MS8 is great especially for a person like myself with very limited tuning experience. But I am seriously thinking about moving to the minidsp to get a little more flexibility. 

What do you use the MS8?


----------



## oabeieo (Feb 22, 2015)

^ my guess would be be because he likes the logic 7 . It's a very very very very very good up mixer . Best I've used yet for a car.


----------



## thehatedguy (May 4, 2007)

Honestly, for a few reasons.

I really like the Logic 7. Good sound for everyone in the car including my kids in the back seats who are starting to like music.

I haven't manually tuned a car in years...before my daughter was born and she will be 7 this year. I had a long long period where I was rebuilding my Accord and then she came a long and the Accord went bye-bye too because I was not in any circumstances putting a child seat in that car as much as it was cut on. So I am really rusty at tuning...I could get a good base tune, but I was never great at fine tuning even when I was "in" it.

A long with that is the time it takes to tune. I loved tuning and stuff when I was single. But right now at this point in my life, I would rather spend my little free time with my wife and children. I just don't want to spend that time away from them. And with conventional drivers, I think the MS8 does a good job tuning. Yeah, I could get a better tune manually, but I have been pretty happy with the results so far if you follow some basic rules.

I have a MS8. Getting something else would require cash out lay.

Speaking of cash, I may do a 2 channel setup for now just to have some good sound and I could afford a 4 or 5 channel amp easier than the 12 channels I want for the multichannel setup.

So that's where I am at. Time and money...and I like the multichannel stuff.


----------



## mitchyz250f (May 14, 2005)

Have you heard a system tuned with REW?


----------



## thehatedguy (May 4, 2007)

I have not.

TBH it has been a while since I've heard much of anything other than my headphones.


----------



## Patrick Bateman (Sep 11, 2006)

thehatedguy said:


> Honestly, for a few reasons.
> 
> I really like the Logic 7. Good sound for everyone in the car including my kids in the back seats who are starting to like music.
> 
> ...


I can empathize with that! I love car audio because it's such a complex problem. But truthfully, my car is used so little, the battery has actually died while it was parked outside. I think I filled up my gas tank about three times in 2014. I've definitely been tempted to just switch over to home audio, the main reason that I don't is that my family wouldn't tolerate a home stereo that was a mess of wires and plywood, they just want something that works. So my personal car is a constant science experiment, and everything else is off-limits.


----------



## oabeieo (Feb 22, 2015)

The family must come first. No doubt.
It's the primary reason my car has a basic install done to it , I don't have time to do custom , and I run a install bay , go figure. Mama barley gives me time to do the little projects in my car as it is. So I totally understand that 100% . 

My 2nd miniHD dosent work and they won't support it . Kinda suckie , they say I hooked it up wrong. Lol , which I thought was pretty funny. So sadly I must buy a nother one so it will be at least a month till I'm up n running now.


----------



## Hanatsu (Nov 9, 2010)

Patrick Bateman said:


> I love car audio because it's such a complex problem.


Same here


----------



## 1fishman (Dec 22, 2012)

oabeieo said:


> .
> 
> My 2nd miniHD dosent work and they won't support it . Kinda suckie , they say I hooked it up wrong. Lol , which I thought was pretty funny. So sadly I must buy a nother one so it will be at least a month till I'm up n running now.


What exactly is wrong with the second unit? They won't warranty it at all?
I assume you got the kit and not the box if that makes a difference for warranty. https://www.minidsp.com/products/minidsp-in-a-box/minidsp-2x4-hd


----------



## oabeieo (Feb 22, 2015)

1fishman said:


> What exactly is wrong with the second unit? They won't warranty it at all?
> I assume you got the kit and not the box if that makes a difference for warranty. https://www.minidsp.com/products/minidsp-in-a-box/minidsp-2x4-hd


They finally answered my emails. 

They are now sending out a new one , and I have to send this one back .
Their customer service is a little bit slow but it is actually pretty good I'm just used to instant gratification I guess from the United States and being able to call somebody. But yes they are finally taking care of it and because they are taking care of it I will continue to use minidsp.


----------



## oabeieo (Feb 22, 2015)

What's even better is I will have 16,384 taps available.

I'm going to use 4 miniHDs and basically just use two channels on each one that way I can still have my P 99 fully functional . And plenty of taps for the sub


----------



## thehatedguy (May 4, 2007)

Maybe I should check with Eric to see if he could make me some high efficiency 10" 2 ohm coaxs, get one of these, use a passive xo on the coaxes, and power them off of my HSS tuber.

I could probably live with that for a while.


----------



## oabeieo (Feb 22, 2015)

thehatedguy said:


> Maybe I should check with Eric to see if he could make me some high efficiency 10" 2 ohm coaxs, get one of these, use a passive xo on the coaxes, and power them off of my HSS tuber.
> 
> I could probably live with that for a while.


That would be the sickness. They also do USB audio so you wouldn't need a head unit , you could just plug your phone directly into the mini and be over with it .


----------



## Hanatsu (Nov 9, 2010)

I managed to attain almost flat group delay on my vented sub using a normal minidsp. I added allpass filters outside the vent gain area then delayed the entire system by ~20ms. If you want to get creative it can be done with biquad programming (IIR). Pretty nice 


Sent from my iPhone 6 using Tapatalk.


----------



## oabeieo (Feb 22, 2015)

Hanatsu said:


> I managed to attain almost flat group delay on my vented sub using a normal minidsp. I added allpass filters outside the vent gain area then delayed the entire system by ~20ms. If you want to get creative it can be done with biquad programming (IIR). Pretty nice
> 
> 
> Sent from my iPhone 6 using Tapatalk.


 Yeah but you're a lot cooler than us so ha!

That is actually pretty cool but you probably had to cut the response quite a bit in certain areas , then add gain to compensate. No ?


----------



## Hanatsu (Nov 9, 2010)

oabeieo said:


> Yeah but you're a lot cooler than us so ha!
> 
> That is actually pretty cool but you probably had to cut the response quite a bit in certain areas , then add gain to compensate. No ?


Magnitude response is unaffected when you use allpass filters. I had to add delay to midbass, midrange, tweeters by 20ms to compensate though. Had to cascade delay from my P99 to those channels, the MiniDSP only does 7.5ms.


----------



## oabeieo (Feb 22, 2015)

Hanatsu said:


> Magnitude response is unaffected when you use allpass filters. I had to add delay to midbass, midrange, tweeters by 20ms to compensate though. Had to cascade delay from my P99 to those channels, the MiniDSP only does 7.5ms.


Ahhhhh. Okay I see. That's pretty nifty


----------



## Hanatsu (Nov 9, 2010)

oabeieo said:


> Ahhhhh. Okay I see. That's pretty nifty


Just did a quick example. Add a pair of appropriate allpass filters so the delay reaches the red line between 50Hz to the mid crossover point. I managed to get it +/- 5ms in the car earlier. Measurements are stored on my laptop so I can't post it right now :<





That allpass filter managed to fix that crossover null with a better GD plot than doing a polarity swap.


----------



## mitchyz250f (May 14, 2005)

thehatedguy said:


> Just curious...if I wanted to play around with this in a multichannel system using the MS8, I would need to place the processor after the MS8 on the pair of channels doing the fronts and use the crossovers in the minidsp for the fronts?
> 
> That is assuming either active 2 way fronts or passive between the midranges/tweeter and active between them and the midbass...full active would require another minidsp and more amp channels.


Patrick/Andy is the above possible? what about with a center channel?


----------



## Patrick Bateman (Sep 11, 2006)

haha, I just realized both Hanatsu and I have George Orwell quotes in our sig. Seriously, that wasn't on purpose.


----------



## Patrick Bateman (Sep 11, 2006)

mitchyz250f said:


> Patrick/Andy is the above possible? what about with a center channel?


I know next to nothing about MS8


----------



## Hanatsu (Nov 9, 2010)

Patrick Bateman said:


> haha, I just realized both Hanatsu and I have George Orwell quotes in our sig. Seriously, that wasn't on purpose.



Haha xD


Sent from my iPhone 6 using Tapatalk.


----------



## oabeieo (Feb 22, 2015)

Finally got measurements to work. It's pretty good. I kinda wanna keep f*ing with it but I think I've got it nailed down. I mostly wanted that 630-800hz L/R flip fixed and it's fixed .

I think I'm going to explore accorite. Looks promising, anyone know how to obtain a copy?


----------



## subterFUSE (Sep 21, 2009)

Patrick Bateman said:


> Gary has a neat setup with an array of mics.
> 
> If I had more time I'd setup some type of jig so that I could switch my mics remotely. (When I measure my car, I use remote access software to control the computer in my garage remotely. That way I can measure the car using a desktop computer, but *run* the mesurements from a laptop. This saves me the hassle of plugging in microphones and preamps to the laptop.)
> 
> But at the moment, I'm still moving the mic the old fashioned way, by hand.












This is the mic array I put together and I'm using currently with SysTune and Smaart. It does real-time spacial averaging and the center mic can do extra duty for timing and phase.

Actually, Smaart can to phase averaging from multiple mics but I haven't got that far with the software yet.


----------



## oabeieo (Feb 22, 2015)

subterFUSE said:


> This is the mic array I put together and I'm using currently with SysTune and Smaart. It does real-time spacial averaging and the center mic can do extra duty for timing and phase.
> 
> Actually, Smaart can to phase averaging from multiple mics but I haven't got that far with the software yet.


That looks very cool. I have means for multiple mics. I have a mic matrix but never use it. 

But more importantly, can/is smaart the way to go for making and implementation of fir? Rew and rephase are decent . I can at least see what's going on and fix it, but a single platform to handle all of it would seem a much better approach. And I want better controls . Trying to figure the Q of min phase is sorta a *****. I want something that will do it automatically


----------



## Hanatsu (Nov 9, 2010)

RoomEQ can autoEQ and output the minPh version of an IR.


Sent from my iPhone 6 using Tapatalk.


----------



## oabeieo (Feb 22, 2015)

Yeah, and after weeks of reading and getting ready for these units, I just discovered that rephase only linerizes crossovers , it can not linerize an entire sweep. (Stupid) 

However I was able to linerize a crossover with good results.

After all day of fiddling with it over all satisfied.

The minor changes = a big change in sound. 
Center is perfect now. All else about the same, have no TA on midbass and it works pretty well. 

Worth the $

Yes.

Worth the time. 
Sorta.


----------



## Hanatsu (Nov 9, 2010)

What do you mean, can't linearize an entire sweep? 


Sent from my iPhone 6 using Tapatalk.


----------



## oabeieo (Feb 22, 2015)

Hanatsu said:


> What do you mean, can't linearize an entire sweep?
> 
> 
> Sent from my iPhone 6 using Tapatalk.


Well I started to get it when I first installed it a few weeks ago, my unit went bad , sent it back bought another one , than baught two more. 

Today the 2nd one showed up so I decided to hook it up after weeks of reading and trying to understand what rephase can do.

I was under the impression that you can import a measurement into rephase (which you can) but you can't set a target phase responce and convolve to it .

All you can do with imported responce is make liner phase crossovers. You have to import the responce from each speaker with no crossover at all . Than you can make fir crossovers.

I don't know why but I thought you could linerize a entire measurement all at once. (Dumb me) not only would you need billions of taps but I can't even find any software that can do it . :/

Being that I'm going to use 4 of these units I will at least have 2048 taps per channel. Which isn't much. They run out of power FAST and the actual convolution looks nothing like your target . 

So only small tweaks here and there are worth it-and it's worth it . Just takes some serious studying of a measurement to know what changes to manually make. The whole program is manually based. No auto tune anything.


----------



## Hanatsu (Nov 9, 2010)

You need alot of taps yes but I have done phase linearization to my subs using APL VST and my laptop. Phase corrections did indeed show up on acoustic measurements. That was with crossovers and stuff in place. Don't wanna say too much, I haven't done much research myself on rePhase even if I've known about it for a while. A few years back you needed a PC or Dirac to use it so it wasn't of much interest...


Sent from my iPhone 6 using Tapatalk.


----------



## subterFUSE (Sep 21, 2009)

This thing is so cheap, I might just grab one to mess around with it. I've been wanting to learn more about FIR filters for a while. This makes it a low-risk proposition to test out.

Since I already have a Helix Pro, would you put this in front of the Helix or behind it? I know APL1 is supposed to go in front.

I'm running 6 channels out of my Helix currently, although 2 of them are for the subwoofer.

Unfortunately, I'm running the HEC module with the extra Optical Input.... so that means my Helix does not have an Optical out. There's an option for an Optical out, but it's on a different HEC module.


----------



## oabeieo (Feb 22, 2015)

subterFUSE said:


> This thing is so cheap, I might just grab one to mess around with it. I've been wanting to learn more about FIR filters for a while. This makes it a low-risk proposition to test out.
> 
> Since I already have a Helix Pro, would you put this in front of the Helix or behind it? I know APL1 is supposed to go in front.
> 
> ...


I would get the HEC module. Always good for one less conversion. But yes. This thing is cheap and the fir is basic but effective. The helix has APF already in 11.5° increments with the corner frequency at crossover for the midrange , so this unit would help with phase in subs , I don't Think it would be a whole bunch of benefits if you have the helix already but it could be depending on how you use it depending on what your problem areas are now .

But on the helix if you going AN (analog) I don't think it would matter which side , I would go before the helix to get the be idiot of the higher voltage of the helix, although it has a 2v out that I measured at 2.7v 1k1w


----------



## Babs (Jul 6, 2007)

subterFUSE said:


> This is the mic array I put together and I'm using currently with SysTune and Smaart. It does real-time spacial averaging and the center mic can do extra duty for timing and phase.
> 
> Actually, Smaart can to phase averaging from multiple mics but I haven't got that far with the software yet.


I want one!!!! Except what is that, like $700 worth of mic's there? You're my hero John! LOL


----------



## Babs (Jul 6, 2007)

subterFUSE said:


> This thing is so cheap, I might just grab one to mess around with it. I've been wanting to learn more about FIR filters for a while. This makes it a low-risk proposition to test out.
> 
> Since I already have a Helix Pro, would you put this in front of the Helix or behind it? I know APL1 is supposed to go in front.
> 
> ...


Maybe an optical switch of some kind in front of the Pro, so to utilize just one toslink input then run out with the HEC.


----------



## subterFUSE (Sep 21, 2009)

Well.... we also don't know whether the Helix optical out is fixed or variable.

How would we control volume if the Helix out is fixed, and the miniDSP is placed after the Helix?


----------



## Elgrosso (Jun 15, 2013)

subterFUSE said:


> This is the mic array I put together and I'm using currently with SysTune and Smaart. It does real-time spacial averaging and the center mic can do extra duty for timing and phase.
> 
> Actually, Smaart can to phase averaging from multiple mics but I haven't got that far with the software yet.


Very cool!
I have two mics, so maybe I could build something like that!
Not for accuracy but at least to save time when I just want the big picture.
What do you use to plug them all on the laptop? Is there a specific multi mic option in Smaart or REW?


----------



## subterFUSE (Sep 21, 2009)

Elgrosso said:


> Very cool!
> 
> I have two mics, so maybe I could build something like that!
> 
> ...




Tascam 1608 sound card. It has 8 mic inputs.


----------



## Elgrosso (Jun 15, 2013)

Thanks!
I just read that REW can't handle multiple mics anyway


----------



## oabeieo (Feb 22, 2015)

subterFUSE said:


> Well.... we also don't know whether the Helix optical out is fixed or variable.
> 
> How would we control volume if the Helix out is fixed, and the miniDSP is placed after the Helix?


It's a variable long as you have the director


----------



## oabeieo (Feb 22, 2015)

Hanatsu said:


> You need alot of taps yes but I have done phase linearization to my subs using APL VST and my laptop. Phase corrections did indeed show up on acoustic measurements. That was with crossovers and stuff in place. Don't wanna say too much, I haven't done much research myself on rePhase even if I've known about it for a while. A few years back you needed a PC or Dirac to use it so it wasn't of much interest...
> 
> 
> Sent from my iPhone 6 using Tapatalk.


Does APL software have ability to export a .bin (IEEE794)?

Wondering if there's anyway I can use that with the miniHD


----------



## Velozity (Jul 6, 2007)

Andy Wehmeyer said:


> Yes, that will work. Just calculate the frequency for the all pass by finding the frequency where 1/2 a wavelength corresponds to the pathlength difference. It doesn't even have to be precise. You just need to move the phase difference away from -180 degrees.



As mentioned here, I want to try the all pass filter function using biquads on a minidsp. Based on my PLDs, I've calculated that the APF frequency for the midranges should be 356Hz and for the midbasses should be 308Hz. The LR4 crossover center frequency I happen to be using right now between midbass and midrange is 302Hz. How do I determine what Q to use in my all pass filters? Is it determined from the phase response graph? Thanks.


----------



## oabeieo (Feb 22, 2015)

Velozity said:


> As mentioned here, I want to try the all pass filter function using biquads on a minidsp. Based on my PLDs, I've calculated that the APF frequency for the midranges should be 356Hz and for the midbasses should be 308Hz. The LR4 crossover center frequency I happen to be using right now between midbass and midrange is 302Hz. How do I determine what Q to use in my all pass filters? Is it determined from the phase response graph? Thanks.


Determining Q I have found to be a bizzznitch. 
The phase plots on rew are SO choppy.

I'll post screen shots of what I did with rephase and maybe will help. 

I'm having aboslutely amazing results with rephase and the miniHD . 

Give me a few to post up what I did .


----------



## oabeieo (Feb 22, 2015)

I had amazing results so far with this adjustment. It sounds like I am in a arena the stage is so wide its infinite sounding. Car has completely disappeared.
Not sure if its correct. But dam it sounds good. And center is pinpoint better now and is center. I have eliminated delay on the 10' midbass
the 6" mid-range is the only thing i am using FIR on. 

Here are some screen shots. Being that re-phase is *manual *I decided to just scratch what I have been trying to do and just look at my measurements.

Here is measurements I took the peaks and divided the frequency by 2 and added it to the lowest peak to determine the center for finding Q


*than* i looked at each measurement and realized if I cut 45deg out of each side and match the uncertain Q i just tried. I only used 45deg instead of the entire 180deg it shows on measurement because I remember Andy posting some square-waves on a measurement he did. He did it IIRC to show how to window the phase so that you can see what would be the equivalent of a RMS for the phase spikes. Being they wrap its hard to see where it is really going.



I used re-phase with the Q i found. Than sorta a eyeball guess along with it in the re-phase plot graph as well. Made sure they looked OK. I can import a measurement into re-phase and try to align them that way but re-phase has a crappy screen resolution in its window so it wasn't worth it. It was easier to just do it this way. I used negative phase change where it was appropriate. 


The left side I only Adjusted for the *SAME* problem at the 324HZ area in the same degree of phase. Being both sides needed the same adjustment. Knowing the lower in frequency whatever change you made to one side will affect the other side unless they are both shifted together. The left side I also added some negative shift at around some 700+hz for that diffrance. Knowing before I had FIR I could reverse the polarity on that speaker and hear the center come in perfect but the lower part was a mess. So I kinda already knew where to target. 




Did just the right side at the single 324hz shift.

Exported it as a moderate and only used 600 something taps per side.

I imported in the the mini


DONE...super easy. I will be doing more. But my problem areas I have been trying to fix with dash-pods that house 6.5'' mid-ranges. the 324hz shift im 99% sure i fixed the ringing and problem created by the dash pods. The pods are basically small enclosures that ring at about the same frequency which affects the overall phase. There is a feature in re-phase that will fix box resonances I have yet to learn. But so far just these two things I have figured out with measurements and this little self do-it program. I have incredible results. Defiantly took my dash horn/dash pod system to a completely different level. Again not sure if I even did it right but it sure sounds like a completely different system.

I also had to do impulse responses of each driver to determine how much delay I had to add to the horns which are not on FIR yet. The FIR did move things a bit in time about 8ms or so. But it was easy to overcome with delay.

Being this is the only software I have that i halfway know how to use I might as well master it for fun. Its a kinda half to now sorta thing. I'm in it this deep so cant quit now.


----------



## oabeieo (Feb 22, 2015)

I *REALLY * wish pos made a manual. :/ 
It really would come in nice about now. 

Does anyone know of a link that has re-phase step by steps for all its features?

I saw a 50 pager on diy but dosent go into specifics.


----------



## Hanatsu (Nov 9, 2010)

oabeieo said:


> Does APL software have ability to export a .bin (IEEE794)?
> 
> 
> 
> Wondering if there's anyway I can use that with the miniHD



It exports a wave and the propritary fir format. You can convert it using roomeq/rephase I guess.


Sent from my iPhone 6 using Tapatalk.


----------



## oabeieo (Feb 22, 2015)

Hanatsu said:


> It exports a wave and the propritary fir format. You can convert it using roomeq/rephase I guess.
> 
> 
> Sent from my iPhone 6 using Tapatalk.


So, if I sent you some REW files, could you convolute it for me?
I can import into rephase and just export it as a .bin and not change anything I suppose. 

I want to do a measurement of the entire left and right channel using sweeps from single mic placement, from 250-1k and than convolve it. And just send the whole file into each bank and let the crossovers in the mini do the seperating to each speaker (MR-MB) . 

Bad idea? Should it be driver specific? I can take measurements of seperatly too if that's would be best. 

Well guess it dosent matter , MR is crossed at 200 anyway 
Would you mind giving it a spin if I can get you the rew files? No rush, of course


----------



## Hanatsu (Nov 9, 2010)

oabeieo said:


> So, if I sent you some REW files, could you convolute it for me?
> I can import into rephase and just export it as a .bin and not change anything I suppose.
> 
> I want to do a measurement of the entire left and right channel using sweeps from single mic placement, from 250-1k and than convolve it. And just send the whole file into each bank and let the crossovers in the mini do the seperating to each speaker (MR-MB) .
> ...



Sure, I'll send my mail to you.


Sent from my iPhone 6 using Tapatalk.


----------



## oabeieo (Feb 22, 2015)

Hanatsu said:


> Sure, I'll send my mail to you.
> 
> 
> Sent from my iPhone 6 using Tapatalk.


Pm replied. Thank you! Your awesome !


----------



## Hanatsu (Nov 9, 2010)

I used rePhase on my subwoofer to my computer system.

Before/After



Impossible to do with a ordinary DSP, especially this MiniHD. You require 20 times the processing power to do these corrections in the lowest frequencies... :<

Is it audible? I'd like to think so, I thought I could do ABX testing with Foobar2000 but can't seem to remember how.


----------



## Hanatsu (Nov 9, 2010)

Anyone seen this DSP?

WAF-Audio

It got 8500 taps freely distributed over all channels, the MiniHD got 2048 max/ch. This is also a 10ch DSP with some interesting features.


----------



## oabeieo (Feb 22, 2015)

Hanatsu said:


> I used rePhase on my subwoofer to my computer system.
> 
> Before/After
> 
> ...


True, but the miniHD has 30ms of available delay, so realistically, fix higher frequencies to the bad spot and delay everything else to it. 

I think that's how they want it to be used. I can't think of any other reason as to why they give so much available delay on board. 

Regardless, your right tho. It's a basic engine , it works miracles on stuff over 50hz and than has enough delay for everything else .


----------



## thehatedguy (May 4, 2007)

I don't think the Nadja ever really took off.

But we could just get a car PC and run Ultimate Equalizer on it...only about 32k taps on every channel.


----------



## Hanatsu (Nov 9, 2010)

Yeah... I'm thinking of simply buying a raspberry pi instead. I'm afraid of the startup time though. More than 30sec is horrible for me.


----------



## oabeieo (Feb 22, 2015)

thehatedguy said:


> I don't think the Nadja ever really took off.
> 
> But we could just get a car PC and run Ultimate Equalizer on it...only about 32k taps on every channel.


 

Geesh . If that dosent do it I don't know what will. Lol


----------



## oabeieo (Feb 22, 2015)

Hanatsu said:


> Yeah... I'm thinking of simply buying a raspberry pi instead. I'm afraid of the startup time though. More than 30sec is horrible for me.


I actually thought I had another broken mini. I didn't know you had to wait 2 full min for it to boot. And when it does it makes sound of a ballon filling with air. 

That must be common than. Love how the manual says nothing about that


----------



## Babs (Jul 6, 2007)

You speak of logic 7. I really hope for something new on that line in the future. Maybe Andy has something in store. 


Sent from iPhone using Tapatalk


----------



## 1fishman (Dec 22, 2012)

oabeieo said:


> I actually thought I had another broken mini. I didn't know you had to wait 2 full min for it to boot. And when it does it makes sound of a ballon filling with air.
> 
> That must be common than. Love how the manual says nothing about that


Wait what? Trying to follow you guys, what takes 2 minutes to start? Your not talking about the Mini hd. :/


----------



## Elgrosso (Jun 15, 2013)

Hanatsu said:


> Anyone seen this DSP?
> 
> WAF-Audio
> 
> It got 8500 taps freely distributed over all channels, the MiniHD got 2048 max/ch. This is also a 10ch DSP with some interesting features.


I looked at it for a while (edit: it does not use sigma studio), but I'm not ready for that UI
Here's a review from a frenchy (youtube translate?):
https://m.youtube.com/watch?v=ov4xqY9zDEg


----------



## SSSnake (Mar 8, 2007)

Jason have you used Ultimate Equalizer? It seems to be an older product and I am not sure that it is still available. I have an email out to the developer...


----------



## thehatedguy (May 4, 2007)

I have not, but the developer spams pretty much every thread about DSP and processing in the PE and DIYA forums. Seems pretty legit though. JohnK of Music and Design likes the product, and he is a pretty smart cookie.

But you are right, it doesn't look like it is getting a lot of support right now.


----------



## thehatedguy (May 4, 2007)

https://dephonica.com/

128k taps per channel.


----------



## Patrick Bateman (Sep 11, 2006)

thehatedguy said:


> I don't think the Nadja ever really took off.
> 
> But we could just get a car PC and run Ultimate Equalizer on it...only about 32k taps on every channel.











Just use a Windows tablet.

Since it's Windows, you can plug in any ol' USB sound card you want.

Add one of those cel phone battery banks and you can leave it on for days.


----------



## 14642 (May 19, 2008)

128k taps? That puts 64000 taps between 10k and 20k at the expense of processing power and latency. Seems excessive to me.


----------



## 14642 (May 19, 2008)

Ultimate EQ looks pretty cool, but the measurement that has to be done will be really problematic in a car. There's really no near-field and far-field. I'm not sure what one would need to use for the "anechoic" data--that's what the near field measurement is supposed to provide. This is a common technique for rooms that are much larger than a car.


----------



## oabeieo (Feb 22, 2015)

Andy Wehmeyer said:


> Ultimate EQ looks pretty cool, but the measurement that has to be done will be really problematic in a car. There's really no near-field and far-field. I'm not sure what one would need to use for the "anechoic" data--that's what the near field measurement is supposed to provide. This is a common technique for rooms that are much larger than a car.


Off subject andy,
But hey . Can you PLEASE, point me in the right direction as to reading material to learn about how to REALLY read phase graphs from REW. I have read the help books twice and it dosent say ****. It just tells you how to take measurements. 

I look at the data , and I understand what's kids going on, I can tell if the amplitude is high it shows like hundreds of degrees high and stuff. But I need to know more. Anything useful is much appreciated 

Andy


----------



## mitchyz250f (May 14, 2005)

thehatedguy said:


> Just curious...if I wanted to play around with this in a multichannel system using the MS8, I would need to place the processor after the MS8 on the pair of channels doing the fronts and use the crossovers in the minidsp for the fronts?
> 
> That is assuming either active 2 way fronts or passive between the midranges/tweeter and active between them and the midbass...full active would require another minidsp and more amp channels.


Andy, Would this work?


----------



## thehatedguy (May 4, 2007)

The reason I was thinking that was if you were to do any all pass filters and stuff, I was thinking the way the upmixer works all of the channels would see the effects of what is going on up stream. A 150 hertz all pass filter would/could be seen possibly in all of the channels minus the sub.


----------



## Hanatsu (Nov 9, 2010)

oabeieo said:


> Off subject andy,
> 
> But hey . Can you PLEASE, point me in the right direction as to reading material to learn about how to REALLY read phase graphs from REW. I have read the help books twice and it dosent say ****. It just tells you how to take measurements.
> 
> ...



We have three different phase plots to view. It's the Phase (Actual phase), Minimum Phase (phase calculated from magnitude response assumed to be Minimum phase) and the Excess Phase (highlights difference between actual phase and minimum phase). 

A downwards slope means there is a delay of the signal (negative slope = peak in group delay). 

Wrapped phase response means when the phase hits -180deg it goes back to +180deg. When comparing wrapped ph responses make sure that number of shifts (the sharp transition) is the same. 

360deg phase equals one period. 1 period consist of a time equal to 1/freq. For instance 50Hz is 1/50=20ms. In this example 360deg equals 20ms, 180deg would equal 10ms. If the phase is close to 180deg (null) at a 50Hz crossover between two drivers we can add an APF that got a group delay of 10ms to bring them back in phase.

In a minimum phase system, the phase can be calculated from the magnitude response. If the mag response is flat the phase will be flat. Phase will shift with any magnitude deviations. Therefore, EQ will fix any phase shifts caused by the room if the system is minimum phase. It takes phase shift to fix phase shift. A car is mixed phase, both non-minimum phase and minimum phase regions are present. A way to see what areas are minimum phase is to look at the excess GD.


Sent from my iPhone 6 using Tapatalk.


----------



## oabeieo (Feb 22, 2015)

Hanatsu said:


> We have three different phase plots to view. It's the Phase (Actual phase), Minimum Phase (phase calculated from magnitude response assumed to be Minimum phase) and the Excess Phase (highlights difference between actual phase and minimum phase).
> 
> A downwards slope means there is a delay of the signal (negative slope = peak in group delay).
> 
> ...


Thank you, 
And yes I agree and all of that I pretty much understand already maybe I should've been a little bit more clear I'm known for posts that are missing elements. But I'm trying to do is convert the measurement so that I can find Q also what I want to know is the conversion of distance to phase angle .

A while ago andy posted the spreadsheet that had everything in it it kind a Look Like Square, waves for phase ,when you first look at it and it really showed a lot of information that spreadsheet he has could be really useful or at least just knowing how to make my own


----------



## Hanatsu (Nov 9, 2010)

Found this Q value convertion into octave width bandwidth:

0.7 = 2 octaves
1 = 1 1/3 octaves
1.4 = 1 octave
2.8 = 1/2 octave
4.3 = 1/3 octave
8.6 = 1/6 octave

Something like this?

http://www.sengpielaudio.com/calculator-timedelayphase


Sent from my iPhone 6 using Tapatalk.


----------



## oabeieo (Feb 22, 2015)

Hanatsu said:


> Found this Q value convertion into octave width bandwidth:
> 
> 0.7 = 2 octaves
> 1 = 1 1/3 octaves
> ...


I know how to determine Q freq/bandwidth 
What I'm looking for , example 

How many inches is 1degree of phase at let's say 300hz for example


And what I meant by determine q is the phase graph is so choppy is so uneven how do I find its center , and even if I did find the center one side of the slope is different shape than the other side of the slope so what do you use?


----------



## sqnut (Dec 24, 2009)

oabeieo said:


> I know how to determine Q freq/bandwidth
> What I'm looking for , example
> 
> How many inches is 1degree of phase at let's say 300hz for example
> ...


As already explained T=1/F to the time period of a 300hz wave is (1/300)*1000 ms = 3.33m/s 

Now 360deg = 3.33 m/s, therefore 1 deg 3.33/360 = 0.009 ms.

speed of sound in air = distance/time

13.5 = d/0.009 therefore d = 0.03"

1 degree at 300 hz is approx 0.03"..... I hope I got it right .


----------



## Hanatsu (Nov 9, 2010)

oabeieo said:


> I know how to determine Q freq/bandwidth
> What I'm looking for , example
> 
> How many inches is 1degree of phase at let's say 300hz for example
> ...



Place APF center freq where phase pass 0 deg. 

x2 on sqnuts post ^^


Sent from my iPhone 6 using Tapatalk.


----------



## oabeieo (Feb 22, 2015)

sqnut said:


> As already explained T=1/F to the time period of a 300hz wave is (1/300)*1000 ms = 3.33m/s
> 
> Now 360deg = 3.33 m/s, therefore 1 deg 3.33/360 = 0.009 ms.
> 
> ...



If I could make a smiley face that's 100fret tall I would post it . 

Thank YOU!!!!

Thank you sqnut and Hanatsu!!!!

Okay . I can't wait to go try it now


----------



## oabeieo (Feb 22, 2015)

Hanatsu said:


> Place APF center freq where phase pass 0 deg.
> 
> x2 on sqnuts post ^^
> 
> ...


Ok where it passes 0 . THAT MAKES SENCE ! 

So just ignore the wiggly lines than yeah?


----------



## oabeieo (Feb 22, 2015)

Hanatsu said:


> We have three different phase plots to view. It's the Phase (Actual phase), Minimum Phase (phase calculated from magnitude response assumed to be Minimum phase) and the Excess Phase (highlights difference between actual phase and minimum phase).
> 
> A downwards slope means there is a delay of the signal (negative slope = peak in group delay).
> 
> ...



Okay so , I'm not making all pass filters yet, just doing rephase. I look at rew I've looked at all the diffrent viewing modes. When I choose 1/24th smoothing unwrapped I can see a hill going downwards , (it's pretty steep) a scroll the phase scroll bar on right in rew and go down, it's pretty smooth to about 600hz, than it goes up and down and has sharp spikes and dips. So if I have a spike that's has a Q of about 40 I go into rephase and make a negative spike in the phase eq that matches the positive spike, get the center , set the q, than I generate the file and send it to dsp. I fix 2 spikes per speaker. Than I run out of taps . It sounds good, only doing 600-1.2k range cause that's where it's chaos and not smooth. 

It makes the center sound very defined. But also sounds a bit .......processed if you will.


Also, the minidsp at low volumes inserts some nasty distortion at low levels. The distortion sounds just like a blown speaker. When I take the minidsp off and go back to the normal deck/to-amp rcas distortion is gone.. Wtf . I'm not overloading the input because volume is low. I have unity gain within the mini so nothing's boosted or cut in the in/outs and I'm using no eq in minidsp. 

It sounded better without the minidsp, but the fir has some things I like so far..


..... What to do ....


----------



## 14642 (May 19, 2008)

mitchyz250f said:


> Andy, Would this work?


Yes.


----------



## 14642 (May 19, 2008)

sqnut said:


> As already explained T=1/F to the time period of a 300hz wave is (1/300)*1000 ms = 3.33m/s
> 
> Now 360deg = 3.33 m/s, therefore 1 deg 3.33/360 = 0.009 ms.
> 
> ...


(1132 x 12)/300/360

1132 Ft/second = Speed of sound in feet/sec
12" in a foot, so 1132 x 12 = speed of sound in inches/sec
(1132 x 12)/300 = length of 300Hz wave.
360 degrees in a sine wave
(1132 x 12)/300/360 = .125"

1/8"


----------



## 14642 (May 19, 2008)

Or, if you'd prefer to calculate by using time rather than distance, as SQNUT has done, then it's

1 second/ 300Hz = .003333333 seconds
.003 seconds / 360 degrees = 8.33333333e-6

Then, to find the length in feet, we MULTIPLY 8.33333333e-6 by 1132 feet

And then multiply again by 12 to find the distance in inches. 

In either case, it's 1/8"


----------



## 14642 (May 19, 2008)

oabeieo,

My spreadsheet with what looks like square waves, wasn't a measurement. It's just a "viewer" that makes it easier for me to see what's going on. 

To build the sheet, I imported the wrapped phase plots of each of the speakers into Excel. Then, I subtracted the phase of one speaker from the phase of another speaker for the graphs. Since I was concerned about cancellation between speakers because that destroys a phantom image between them, I wanted to view areas where the phase difference creates cancellation. That happens when the difference is near 180 degrees. Doesn't matter whether the difference is +180 or -180, so I used the absolute value of the phase difference. That graph wasn't easy to read. I wanted something that would help me see this easily and to help me estimate the Q value for an all pass filter, I wanted to see the areas where cancellation begins and ends, I set that value at 130 degrees. 

To build the final graph, I used "IF(AND...)" to display all values between 130 and 180 as 180 and all values below 180 as 0. Then, I offset the graphs by a value of 10 so they wouldn't fall on top of each other to make viewing easier


So, areas where the graph displays 180 are areas in which the image is destroyed and areas at 0 are areas where the image is OK. Then, I estimated Q by frequency/BW which is the center of the horizontal line divided by the distance between the two ends of the horizontal line. I have my doubts about the audibility of the very narrow regions because those change dramatically if the microphone is moved by a very small amount. Also, because we don't hear phase t high frequencies easily. Finally, because between 1k and 3k the wavelengths are close to the distance between our ears and neither phase nor intensity differences are easy for us to interpret in figuring out the apparent origin of a sound.

That gives me an estimate of the center frequency and the Q for an all pass filter. I applied all three of these, but the third one was inaudible (as I sort of expected). The first two fixed my two phantom image problems in both seats. This was using simple IIR all pass filters.


----------



## 14642 (May 19, 2008)

I wouldn't attempt any phase EQ above about 1k. Clearly, the region where this is important for imaging is between about 80 Hz and 800 or so. It's also important to get the phase between the sub and the front speakers to within about 90dB, so there aren't any nulls in the midbass. Depending on the location of the sub, you may find that the opposite side midbass is out of phase with the sub while the near side is in phase. A compromise that shifts the sub about 90 degrees at the null can be helpful in balancing left and right. At 90 degrees, the sum is +3 rather than +6 and that makes it possible to apply a little EQ to fix the smaller resulting dip.


----------



## oabeieo (Feb 22, 2015)

Andy Wehmeyer said:


> oabeieo,
> 
> My spreadsheet with what looks like square waves, wasn't a measurement. It's just a "viewer" that makes it easier for me to see what's going on.
> 
> ...



Andy, 

Thank you! Boy did I misinterpreted the last time you posted that 

it makes perfect sence now. thanks man 

Gosh dang this last two days has been fun and learning. For the last two weeks somehow I was treating two seperate principals as one. Nothing I was saying probably made sence  

And thank you for the formula, I WILL use put that to good use.  I want to make a APF for the right side to move everything from 250-500 to match the wavefront from other side where wavelength is long enough to do it. I needed that formula .


----------



## 14642 (May 19, 2008)

It's a common problem. Just use a second order all pass filter centered at 250 Hz with a Q of about 1.


----------



## oabeieo (Feb 22, 2015)

Andy Wehmeyer said:


> I wouldn't attempt any phase EQ above about 1k. Clearly, the region where this is important for imaging is between about 80 Hz and 800 or so. It's also important to get the phase between the sub and the front speakers to within about 90dB, so there aren't any nulls in the midbass. Depending on the location of the sub, you may find that the opposite side midbass is out of phase with the sub while the near side is in phase. A compromise that shifts the sub about 90 degrees at the null can be helpful in balancing left and right. At 90 degrees, the sum is +3 rather than +6 and that makes it possible to apply a little EQ to fix the smaller resulting dip.


Okay! sweet !  

Also, 

so let's say I have a 10"midbass in a sealed box /or ported box let's say the box is a bit too small and has a 100hz peak. 

Should this be a good spot to use linear phase EQ?


----------



## 14642 (May 19, 2008)

No. The response of the woofer in the box is minimum phase. You can fix that with simple EQ. If, instead of a peak, it was a dip caused by a reflection, then the law of causality applies and it can't be equalized by applying filters BEFORE the speaker. If the dip was caused by interaction with another speaker, then being able to alter the phase independently of magnitude response would be the right fix.


----------



## sqnut (Dec 24, 2009)

Andy Wehmeyer said:


> Or, if you'd prefer to calculate by using time rather than distance, as SQNUT has done, then it's
> 
> 1 second/ 300Hz = .003333333 seconds
> .003 seconds / 360 degrees = 8.33333333e-6
> ...


Thanks yes I see where I was wrong you don't multiply 1/F by 1000 and its 13584 not 13.5 (1132*12=13584).


----------



## oabeieo (Feb 22, 2015)

I'm getting vERY FRUSTRATED WITH MINIDSP 
They have now sent me 2bad units! 

Aarg , I have 2 working and 2 bad ones. What the heck I need to vent. 
Very very upsetting!  

Now I have to wait for there 2week email coraspondance to get new ones in another month.


----------



## Hanatsu (Nov 9, 2010)

oabeieo said:


> I'm getting vERY FRUSTRATED WITH MINIDSP
> They have now sent me 2bad units!
> 
> Aarg , I have 2 working and 2 bad ones. What the heck I need to vent.
> ...


Damn... I've had no issues with the normal 2x4 units for years now. Ordered a C-DSP instead of this. Hope that works better.


----------



## oabeieo (Feb 22, 2015)

Hanatsu said:


> Damn... I've had no issues with the normal 2x4 units for years now. Ordered a C-DSP instead of this. Hope that works better.


My 6x8 and 2x4 were great,
It's the dam 2x4HD. Shot dood this is 3 out of 4 bad. Using the same wiring and same platform I only have 2 working. The others work, turn on but have no output of 2ch. I have the input matrix setup correct, it shows signal on both inputs . The other one wouldn't even turn on.  and the one they sent back as warranty same ****. This is garbage


----------



## oabeieo (Feb 22, 2015)

After a whole day of tuning and fiddling around, I extra carefully did the math , and did thoughtful processes. I got two APFs in place , a tiny tiny bit of convolution, and some careful impulse measurements,.... The sq is fantastic, system sounds good but it is soooo not what I expected. Big problem. The center is now directly in front of me. This sucks. I can't just add delay because it will ruin all the thought and hard work I did on changing phase. I didn't think about it but I took all my measurements from basicly my nose area. So that was the "center" for both sides. So now everything is aligned to that. I got phase dialed in as much as I could, I made the very best I possibly could with 128 samples. 

It makes sence tho now. I guess a whole day of learning. by making phase better for one spot and equal amplitude on everything the center is directly in front of me. 

Tomorrow I'm going back to way it was. But this is what I learned today. I think that when we hear the phantom center in middle of car because amplitude at all frequencies are very close but phase is not. Alignment of phase and amplitude results in center being right in front of you. This I didn't think of while I was doing it. But it makes sence now. On the home stereo we hear the center when were between the speakers. In a car we hear it between the speakers as well with TA.-only because the phase is diffrent from side to side making a natural shift that points the phantom center to be between the speakers. That natural shift in most frequencies is preferable to me and I only care to "correct" things that are 180 out. AND ALL ELSE LEAVE ALONE. 

16hrs I wasted learning this- was worth the learning experience, but boy I wish I read it somewhere. 128 samples is enough for a car stereo. We don't needs billions and billions of taps . If you like it to sound like a home stereo with center in front of you and with a hard reflection to the left (on the right of you drive in England or Japan) go for it . But I'm going back to the way it was with minimal fir and one APF. 

(Btw that APF did do the job very well)


----------



## Orion525iT (Mar 6, 2011)

oabeieo said:


> I'm getting vERY FRUSTRATED WITH MINIDSP
> They have now sent me 2bad units!
> 
> Aarg , I have 2 working and 2 bad ones. What the heck I need to vent.
> ...


That sucks. My 2x8 and all the associated plug in cards have been flawless.

Since I am running a car PC, I might start to dig into plug-ins for VST and other players instead of a hardware device. I only really care about what happens between 80 and 300hz because of my bandpass midbasses. Maybe for the ported sub too?

Has anyone tried or know of a way to alter the original song tracks to make the adjustments? It would not allow for any flexibility, but once it's done you wouldn't need to worry about it again unless you changed your setup.


----------



## sqnut (Dec 24, 2009)

I've never used FIR but from what I understand of it, if used correctly it shouldn't move the center, just make the sound more focused. I'll let someone with hands on experience comment.


----------



## Hanatsu (Nov 9, 2010)

Orion525iT said:


> That sucks. My 2x8 and all the associated plug in cards have been flawless.
> 
> 
> 
> ...



I actually tried EQing and applying T/A to all songs I loaded on a cd to be able to playback processed music on a basic stock system. Worked quite well actually. Might do a thread about it.


Sent from my iPhone 6 using Tapatalk.


----------



## Babs (Jul 6, 2007)

Might not be the place for this, but I've come to the realization I have a 1-2-3 kind of a regimen for tuning that's not really set in stone but as a newb I'm trying to dial it in. Is it final and a constant in the universe like Pythagorean theorem? No, but it's kind of what I've established from what I've learned so far.. Trick is, the tool.. File saving. 

Listing the process, it makes me think no wonder there's so much to discuss just on that one little noun "tuning":

1. Rough crossovers, rough leveling
2. Initial distance TA
3. Acoustic slopes and top side matching by levels and crossovers
4. Individual EQ from REW Auto EQ and/or by graph, tool of choice.
5. Individual EQ verification by measurement
6. TA finalizing with individually balanced pairs
7. TA and Phase verification
8. EQ Pairs by Measurement
9. EQ Pairs by 1/3 octave to center stage
10. EQ overall and resulting 'curve'

No wonder I'm neurotic. The struggle is real.

I guess point to all this is, I've found from here and discussion with folks, there are some items with tuning that absolutely MUST be done in certain sequence from other items, so it sets your tasks in list. And it's iterative but some things you kinda can't go back to without completely redoing the tasks that have to occur after the previous that was tweaked. And these items above take none of the bleeding edge stuff into account such as all-pass filters, software measured TA, etc etc.. Just plain ole DSP stuff.

And what's even frustrating in the stage I found as a newb is all bets are off in some cases when you go to the next stage. You can have your individual responses just spot on nailed, glorious and one line when they're shown together on a plot, then you play both sides and what the heck just happened to that wonderful curve. Oh yeah.. Car.


----------



## 14642 (May 19, 2008)

oabeieo said:


> Alignment of phase and amplitude results in center being right in front of you.
> 
> 
> 
> (Btw that APF did do the job very well)


Something else must be wrong. Check the levels and the frequency response matching carefully. It should place the center in the center.


----------



## 14642 (May 19, 2008)

Babs said:


> Might not be the place for this, but I've come to the realization I have a 1-2-3 kind of a regimen for tuning that's not really set in stone but as a newb I'm trying to dial it in. Is it final and a constant in the universe like Pythagorean theorem? No, but it's kind of what I've established from what I've learned so far.. Trick is, the tool.. File saving.
> 
> Listing the process, it makes me think no wonder there's so much to discuss just on that one little noun "tuning":
> 
> ...


Set TA first using a tape measure. It affects the sum of the crossovers.


----------



## Babs (Jul 6, 2007)

Andy Wehmeyer said:


> Set TA first using a tape measure. It affects the sum of the crossovers.


Thank you Andy but ah.. TA can be later because I'm setting crossovers just based on individual plots in order to match.. Check left, check right, see both, etc. So since it's driver independent at that point and just individual FR timing isn't even necessary at that step, but I understand what you mean as in TA affects 'everything' between more than one driver. 

Edit: Changed my ordering though because we all know realistically who drives around with a partial tune without TA done at least to distance.


----------



## Orion525iT (Mar 6, 2011)

Hanatsu said:


> I actually tried EQing and applying T/A to all songs I loaded on a cd to be able to playback processed music on a basic stock system. Worked quite well actually. Might do a thread about it.
> 
> 
> Sent from my iPhone 6 using Tapatalk.


Please do when you get the chance. It would be an interesting experiment if nothing else. It would be neat to compare the results to those achieved by other methods. Theoretically, it should work out the same I think as long as the changes don't degrade the original signal in some manner. Basically just re-engineering the songs. Of course that goes to **** if you stream.


----------



## oabeieo (Feb 22, 2015)

Andy Wehmeyer said:


> Something else must be wrong. Check the levels and the frequency response matching carefully. It should place the center in the center.


Oh that is good to know. 

I'm rethink this now. Midbass is on Stinkin point now. And the subs low low notes are much more sharp sounding. It's the darn midrange that's messed up. 
Im going back work on it now. Gotta love blowing both days off tuning 


Thank you!


----------



## Hanatsu (Nov 9, 2010)

OT. Here's an easy way to do it if you got measurement gear and a powerful proper DSP.

1. Measure all drivers individually. No EQ, no T/A, no crossovers (except highpass on Tweets, place it as low as possible).

2. Apply EQ towards a curve in a wide range as possible, 1 octave below/above the intended crossover point.

3. Set crossovers.

4. Overlay all EQed individual responses, set levels to the curve and L/R response.

5. Set delays. After this, measure full system (L/R) to see that drivers sum around crossovers etc. 



Sent from my iPhone 6 using Tapatalk.


----------



## Hanatsu (Nov 9, 2010)

Orion525iT said:


> Please do when you get the chance. It would be an interesting experiment if nothing else. It would be neat to compare the results to those achieved by other methods. Theoretically, it should work out the same I think as long as the changes don't degrade the original signal in some manner. Basically just re-engineering the songs. Of course that goes to **** if you stream.



Yep, I'm going away on a work seminar for few days. I'll go ahead and make a thread about it when I'm home again.


Sent from my iPhone 6 using Tapatalk.


----------



## oabeieo (Feb 22, 2015)

I went out and re did levels. It's all better now. 

John over on the minidsp fourm sent me a config file that brought the dead mini back to life. All channels working now. Woot woot !!!!

So I have 256 samples per channel now, 2046 taps per channel . 


I have everything sounding better than ever before. No arena sound anymore although it was cool it wasn't correct. Everything is correct now. Super happy . 

The only thing I haven't worked on now is the sub. 
I have 2046taps per output, what to do....

This is the wierd part maybe I can get help with. When I choose a linear phase filter in rephase it works although it crossover is much higher than I wanted and it drops the level by about 10db ... Wth.

I tryed every type of windowing, and a 80hz48db xo sounds like a 200hz 12-dbxo.

What would I do ?


----------



## Babs (Jul 6, 2007)

Hanatsu said:


> OT. Here's an easy way to do it if you got measurement gear and a powerful proper DSP.
> 
> 1. Measure all drivers individually. No EQ, no T/A, no crossovers (except highpass on Tweets, place it as low as possible).
> 
> ...



That's certainly another way to skin a cat. 

I sure wish REW would allow you to show the target curve on the main screen. Minor thing but man that'd be cool. I suppose once you measure all together you can compare easy enough in the EQ screen as a single plot.


----------



## Hanatsu (Nov 9, 2010)

Orion525iT said:


> Please do when you get the chance. It would be an interesting experiment if nothing else. It would be neat to compare the results to those achieved by other methods. Theoretically, it should work out the same I think as long as the changes don't degrade the original signal in some manner. Basically just re-engineering the songs. Of course that goes to **** if you stream.


Here you go...


----------



## Hanatsu (Nov 9, 2010)

Babs said:


> That's certainly another way to skin a cat.
> 
> I sure wish REW would allow you to show the target curve on the main screen. Minor thing but man that'd be cool. I suppose once you measure all together you can compare easy enough in the EQ screen as a single plot.


It does allow for that. Just import the curve as text in File/Import Frequency Response.


----------



## oabeieo (Feb 22, 2015)

oabeieo said:


> I went out and re did levels. It's all better now.
> 
> John over on the minidsp fourm sent me a config file that brought the dead mini back to life. All channels working now. Woot woot !!!!
> 
> ...


Okay I got it! Just linerizing the xo made such a amazing difference. That one thing works very darn good. Sub is not a train wreck of arrival times anymore.

But...... All of the sub seems just a tiny bit out of time.......how do I determine that. Even gated impulse is marginal. 

Also Simon, heya how do I get OpenOffice to work right? 

The worksheet opens I can do anything whitin it just fine, when I cut and paste the cells where biquads are out putted it's not the right format. Spacing is wrong and no commas also just dosent quite look right at top? 

I'm supposed to cut n paste cells right? Maybe that's where I'm messing up.

Btw thanks again for all your help dood!  I finally getting a really sick system


----------



## Hanatsu (Nov 9, 2010)

oabeieo said:


> Okay I got it! Just linerizing the xo made such a amazing difference. That one thing works very darn good. Sub is not a train wreck of arrival times anymore.
> 
> But...... All of the sub seems just a tiny bit out of time.......how do I determine that. Even gated impulse is marginal.
> 
> ...


I formatted cells to do dots instead of commas.


----------



## oabeieo (Feb 22, 2015)

Hanatsu said:


> I formatted cells to do dots instead of commas.


Do I highlight the rows or cells? After I change it to dot I than cut n paste like I did before?


You have the "stable" row selected do I do that too?


----------



## Babs (Jul 6, 2007)

Hanatsu said:


> It does allow for that. Just import the curve as text in File/Import Frequency Response.



Sweet!!


----------



## Hanatsu (Nov 9, 2010)

oabeieo said:


> Do I highlight the rows or cells? After I change it to dot I than cut n paste like I did before?
> 
> 
> You have the "stable" row selected do I do that too?



Tou can't just copy paste it though, it must follow the format in the minidsp software as I linked before. The stable row isn't required, I was just sloppy when when I marked them..


Sent from my iPhone 6 using Tapatalk.


----------



## oabeieo (Feb 22, 2015)

Hanatsu said:


> Tou can't just copy paste it though, it must follow the format in the minidsp software as I linked before. The stable row isn't required, I was just sloppy when when I marked them..
> 
> 
> Sent from my iPhone 6 using Tapatalk.


Sorry I'm half retarded , I got it finally. 

That spreadsheet has everything . Will FIR banks recognize these biquads?
The xo banks work fine, just don't wanna mess anything up in regards to my luck with minidsps. Lol  

Im also working on making a spreadsheet andy was talking about. I'll host it when I get it coded right. It works so far just importing back into rew isn't working yet. I'll figure it out tho.


----------



## Hanatsu (Nov 9, 2010)

Biquads are IIR building blocks. That spreadsheet doesn't output anything related to FIR, perhaps you can combine iir/fir filters in the HD?


----------



## oabeieo (Feb 22, 2015)

Just got filter hose .

Now it's on.


----------



## oabeieo (Feb 22, 2015)

Question;
Is the latency of a filter, the literal time offset that can be used to ajust TA controls on dsp to compensate for delay caused by any specific filter? 

Please kindly, give explained.


----------



## Hanatsu (Nov 9, 2010)

oabeieo said:


> Question;
> Is the latency of a filter, the literal time offset that can be used to ajust TA controls on dsp to compensate for delay caused by any specific filter?
> 
> Please kindly, give explained.


Eh don't really understand the question. You mean the DSP would use T/A as compensation for some filter...?

Latency increase with FFT length used in FIR processing.


----------



## oabeieo (Feb 22, 2015)

Hanatsu said:


> Eh don't really understand the question. You mean the DSP would use T/A as compensation for some filter...?
> 
> Latency increase with FFT length used in FIR processing.


 Right in the latency that they talk about in F IR does it trickle down effect the timing of the system? Basically does it add more delay ? And if so is that delay the amount of milliseconds of the latency? Reason I ask is after processing there is definitely a change in timing I've been using impulse responses to realign things but the file shows the latency it would be a heck of a lot easier to just add that number to the TA settings does that make sense? 

Sorry I'm bombarding everybody with 1 million questions I super appreciate your time


----------



## Hanatsu (Nov 9, 2010)

FIR based processing definitely delay the signal in whole, you know the famous "pre-ringing"... They even speak about it on the MiniDSP site. People have issues with "lip sync" after FIR corrections. If lip sync is an issue then the delay must be significant. Seen people on forums talk about 100ms+ delays on DIRAC setups. FIRmaker's website says 5ms for 250taps @ 48k. Add internal latency and delays from crossovers and we might be seeing ~10ms even with small filters.

Problem is that you must delay all speakers not affected by the IR offset, you might need lots of time delay to fix this.


----------



## oabeieo (Feb 22, 2015)

Hanatsu said:


> FIR based processing definitely delay the signal in whole, you know the famous "pre-ringing"... They even speak about it on the MiniDSP site. People have issues with "lip sync" after FIR corrections. If lip sync is an issue then the delay must be significant. Seen people on forums talk about 100ms+ delays on DIRAC setups. FIRmaker's website says 5ms for 250taps @ 48k. Add internal latency and delays from crossovers and we might be seeing ~10ms even with small filters.
> 
> Problem is that you must delay all speakers not affected by the IR offset, you might need lots of time delay to fix this.


 Oh thank you that clears it up for sure, for that sakes I'm just going to use the latency delay as a solid number for now my impulse measurements dictate that it is correct But you know as well as I that the impulse isn't always exactly perfectly on time yes that pre-ringing really does confuse things before the peak of the IR


----------



## strohw (Jan 27, 2016)

Can you chain these units in a way? I.E. 1 input to the first one and 1 set of outputs to a speaker set and another set of outputs to another MiniHD. Then both sets of outputs from the second unit to speaker sets.


----------



## XR250rdr (Mar 22, 2011)

strohw said:


> Can you chain these units in a way? I.E. 1 input to the first one and 1 set of outputs to a speaker set and another set of outputs to another MiniHD. Then both sets of outputs from the second unit to speaker sets.


I don't see why you couldn't. There will be some added latency to the second unit, but that should be able to be accounted for.

If I had a digital source I would be really interested in trying out a miniSHARC with a miniDAC8.


----------



## oabeieo (Feb 22, 2015)

So there are some small flaws with these units.

1. When an FIR filter (from rephase)is loaded that contains a FIR crossover the unit boots up muted. (Really gay and annoying) filter linerization files work fine.

2. When you use a linear EQ in rephase the frequency is wrong. Rephase shows a certian curve loaded and 2x4HD shows the same curve but a few hundred hertz off of what rephase shows. When changing sample rates it moves the responce curve around slightly but it never matches what rephase shows. Can't tell if it's a rephase problem or a 2x4hd problem. (Again really annoying) 

3. Because of problem #2 I am uncertain if the filter linerization is even correct or if it's a few hindered hertz off as well. 

4. Pre-ringing. Even with a hann window some feed back is heard at low volumes. Makes me think it's a artifact of problem #2 as well.

Trying to figure out how I going to take measurements to see if I can off set the values and compensate. Overall not super pleased with these units. Fir is nice but not at the expense of accuracy between software platforms.


----------



## oabeieo (Feb 22, 2015)

"Neo" on the movie "the matrix" was a fir filter


----------



## Patrick Bateman (Sep 11, 2006)

Andy Wehmeyer said:


> oabeieo,
> 
> My spreadsheet with what looks like square waves, wasn't a measurement. It's just a "viewer" that makes it easier for me to see what's going on.
> 
> ...


Did anyone ever save the images from Andy's post?

I'm thinking about implementing this for my home theater. Basically we have two couches and nobody sits directly in front of the loudspeakers.

So it seams like the innovation that Andy describes for making a 'two seat car' could also be used to make a 'two couch home theater'


----------



## Patrick Bateman (Sep 11, 2006)

I'm too impatient to wait for someone to upload the pics, so I think I'll try and decipher this myself. 

OK, first step, I got a tape measure and I measured the pathlength from my door speakers to my ears. The distance for the left speaker is 1.12M and the distance from the right is 1.47M. A pathlength difference of .35M.

Here's what happens to our two speakers when they play midrange and midbass:

1) 971Hz is 35 centimeters long. That means that when my left speaker is playing 971Hz and my right speaker is playing 971Hz, the left speaker is "ahead" of the right speaker by one full wavelength. This isn't going to cause amplitude problems (because they're in-phase) but it will cause imaging problems (because the left speaker is "ahead" by one wavelength.)

2) 486Hz is 70 centimeters long. That means that when my left speaker is playing 486Hz and my right speaker is playing 486Hz, the left speaker is "ahead" of the right speaker by one half wavelength. This will cause amplitude problems; the left and the right speaker are 180 degrees out-of-phase, they're cancelling each other out. The imaging problems may be less severe, since it's just a fraction of a wavelength.

2) 243Hz is 140 centimeters long. That means that when my left speaker is playing 243Hz and my right speaker is playing 243Hz, the left speaker is "ahead" of the right speaker by one quarter wavelength. This won't cause amplitude or phase problems; basically the waves are so long that the left and the right speaker are behaving like a single unit.


----------



## oabeieo (Feb 22, 2015)

Patrick Bateman said:


> I'm too impatient to wait for someone to upload the pics, so I think I'll try and decipher this myself.
> 
> OK, first step, I got a tape measure and I measured the pathlength from my door speakers to my ears. The distance for the left speaker is 1.12M and the distance from the right is 1.47M. A pathlength difference of .35M.
> 
> ...



This is old thread, but still relevant very much.

I'll upload pics soon 

This thread goes way back before I knew swat and was still figuring this out. Things have come a long way since than 
A lot of noob ish talk if anyone reads this, at least from my points .


----------



## Patrick Bateman (Sep 11, 2006)

Patrick Bateman said:


> I'm too impatient to wait for someone to upload the pics, so I think I'll try and decipher this myself.
> 
> OK, first step, I got a tape measure and I measured the pathlength from my door speakers to my ears. The distance for the left speaker is 1.12M and the distance from the right is 1.47M. A pathlength difference of .35M.
> 
> ...


*

OK, I re-ran the numbers, using the processing that Andy described earlier.









Here is a graph of the frequency response, the phase, and the group delay of the filter that Andy recommended. I worked this out using the spreadsheet that @Hanatsu posted. (Took 30 seconds, super easy)
The processing is what Andy recommended: 

"Just use a second order all pass filter centered at 250 Hz with a Q of about 1. - Andy Wehmeyer" (from post #181)


1) 971Hz is 35 centimeters long. That means that when my left speaker is playing 971Hz and my right speaker is playing 971Hz, the left speaker is "ahead" of the right speaker by one full wavelength. This isn't going to cause amplitude problems (because they're in-phase) but it will cause imaging problems (because the left speaker is "ahead" by one wavelength.)
The processing that Andy recommends delays the left speaker by 0.09ms. Basically no difference.

2) 486Hz is 70 centimeters long. That means that when my left speaker is playing 486Hz and my right speaker is playing 486Hz, the left speaker is "ahead" of the right speaker by one half wavelength. This will cause amplitude problems; the left and the right speaker are 180 degrees out-of-phase, they're cancelling each other out. The imaging problems may be less severe, since it's just a fraction of a wavelength.
The processing that Andy recommends delays the left speaker by 0.49ms That means that before Andy's processing, the left and the right speakers were 1.03ms out-of-phase. After the processing, they're 0.54ms out-of-phase. This means that the two speakers used to be 180 degrees out of phase, but now they're approximately 90 degrees out of phase. This means that Andy's processing will accomplish a couple of things. First, you won't have a null because of the phase difference between the left and the right speaker. Second, the two speakers will be very close in phase, which helps imaging.

3) 243Hz is 140 centimeters long. That means that when my left speaker is playing 243Hz and my right speaker is playing 243Hz, the left speaker is "ahead" of the right speaker by one quarter wavelength. This won't cause amplitude or phase problems; basically the waves are so long that the left and the right speaker are behaving like a single unit. The processing that Andy recommends delays my right speaker by two and a half ms?

Okay, there's a problem here, the filter is tuned to the wrong frequency. It clearly should be tuned higher.

Stay tuned...

*


----------



## Patrick Bateman (Sep 11, 2006)

OKAY, the post right above shows you what happens if you don't do the filtering correctly.

Do NOT set the filter to 250Hz, that number is ONLY good if your pathlength difference happens to be 68cm.

What you WANT to set the filter to is *one half of the pathlength difference between your left and your right speaker.*

For instance, the pathlength difference between my left and right midbass is 35cm. So I set the filter to 486Hz. (972Hz is 35cm long.)

Once I do that, I get the following corrections:

1) At 972Hz:
Before I implemented the filter, my left speaker is "ahead" of the right speaker by one full wavelength. *After implementing the filter my left speaker is "ahead" of the right speaker by three quarters of a wavelength. * So it hasn't "fixed" the one wavelength difference between the left and the right speaker, *but they're still in-phase* so we don't get any amplitude problems. And though the phase isn't perfect, it is indeed better.

2) At 486Hz:
Before I implemented the filter, my left speaker is "ahead" of the right speaker by one half wavelength. This is the real problem frequency, it's what we need to fix, we don't want the left and the right speaker 180 degrees out-of-phase. *After implementing the filter my left speaker is "ahead" of the right speaker by one quarter of a wavelength.* This has made a significant improvement in the phase response of the system; the phase difference between the left and the right speaker is so small it should be inaudible. And there won't be any cancellation between the left and the right speaker. Nice!

3) At 243Hz:
Before I implemented the filter, my left speaker is "ahead" of the right speaker by one quarter wavelength. This is no big issue. *After implementing the filter my left speaker is "ahead" of the right speaker by one quarter of a wavelength.* No real difference here.



In summary: this filter isn't a cure-all, it doesn't fix the phase perfectly. But it's pretty darn nice. It really seems like a more intelligent way to do time delay. This filter fixes the phase issue for the driver *while only manipulating the phase of the system over a very narrow bandwidth.* IE, the driver will perceive a significant improvement in the phase response without totally nuking the passengers soundstage.

For me, the next question is why you wouldn't just run this instead of a time delay? I guess the answer to that question depends on how audible the phase shift at 1khz is. IE, a TIME DELAY will fix the pathlength difference across the entire bandwidth of the driver. But by doing so, it makes things very poor for the passenger. Anyone who's listened to a "one seat car" from the passenger seat knows what I'm talking about.

Some experimentation is required here. But I have a hunch that if you put in the effort to PHYSICALLY equalize the pathlengths, and then you used this filter, I'll bet it would sound superior to a system that uses time delay and "stock" speaker locations. IE, there's no replacement for physically locating the loudspeakers in the proper location, which is why guys like Mark Eldredge rebuild their entire interiors. But once you've got the pathlengths as equal as you can get, I have a hunch that this filter will get you to the next level. Particularly because a lot of what we hear in the car is reflected energy, and this filter doesn't make dramatic changes to the phase response of the system like a conventional delay filter does.


----------



## Patrick Bateman (Sep 11, 2006)

By the way, if Andy is reading this thread, here's a free patent idea:

If you look at the way that your filter works, *you're using an all-pass filter to correct the phase over approximately two octaves.*

Now here's something to consider:

Instead of using the filter to CORRECT the phase, you could also use the filter to INVERT the phase. And combine that with a second woofer, you can make a very nice cardioid.

This has real commercial potential, because cardioids sound fabulous but they're not efficient. But by implementing your filter (in reverse) you can get the fabulous sound of a cardioid, with higher efficiency. And prosound people LOVE efficiency 

More info here: Beer Budget Beam Steering - Page 4 - diyAudio

Here's an example of the commercial potential of the filter:

Let's say you have an outdoor concert and there are multiple stages. With a conventional set of loudspeakers, the speakers will lose directivity control when the wavelengths are larger than the loudspeakers themselves. For instance, a loudspeaker that measures 30" x 30" will lose directivity at 450Hz. This means that all of it's output below 450Hz will be omnipolar, it will be radiating in 360 degrees.

So we take the filter that you invented, but we flip it, so that it intentionally makes the loudspeaker directional. Now we can 'tune' it so that the 'null' maxes out at a specific frequency. For instance, with a prosound speaker that plays down to about 150hz, you might 'tune' the filter to 200Hz. So that the loudspeaker is very directional at 200Hz, but less so at 400Hz. (The reason that we tune it like that is because the cabinet itself will constrain the radiation above 400Hz.) The combination of the cabinet dimensions and the filtering will yield a loudspeaker with controlled directivity over nearly the entire bandwidth, using inexpensive and affordable filtering. For a prosound venue, it means you can have two stages that don't "bleed" into each other without sacrificing all your efficiency.

I know that cardioids are all the rage in Prosound, but I've noticed that they're not widely used. I think this is largely because you have to know what you're doing to get them right, and traditional implementations sacrifice efficiency. This solution fixes both issues, because the cardioid filters would be built right into the loudspeaker, not implemented on site. And it's more efficient. Win-Win.


----------



## oabeieo (Feb 22, 2015)

Patrick Bateman said:


> By the way, if Andy is reading this thread, here's a free patent idea:
> 
> If you look at the way that your filter works, *you're using an all-pass filter to correct the phase over approximately two octaves.*
> 
> ...


So, yes It definitely Could work this way. However in a car the APFs time lag would make that challenging at the least , so an fir would be much easier , if in an fir there's better ways to target a band than the use of an APF . 

I think cardioid would be an excellent option for combfilters in a car, just like prosound a cardioid has to be implemented with exacting accuracy. 

I think prosound hasn't taken off much because only big headliners have the bucks to make a cardioid do its full magic and the amount of money on a 2nd playback system meant to do the exact opposite they spent all the money on the 1st system to get loud and clean. But more than that, the big shows are always in a different location, so implementing a cardioid may not be in order at every venue. 

In a car, the location is always the same so it makes sence to do it. Especially targeting combfilters and reflecting surfaces that are offensive.....


And to comment on your opening line, 
Another way to "invert" the polarity would be to switch the speaker wires 
No APF needed for that :shy:


----------

