Go Back   Home > Forums > >
Home Forums Rules Articles diyAudio Store Blogs Gallery Wiki Register Donations FAQ Calendar Search Today's Posts Mark Forums Read

Multi-Way Conventional loudspeakers with crossovers

rePhase, a loudspeaker phase linearization, EQ and FIR filtering tool
rePhase, a loudspeaker phase linearization, EQ and FIR filtering tool
rePhase, a loudspeaker phase linearization, EQ and FIR filtering tool Please consider donating to help us continue to serve you.

Ads on/off / Custom Title / More PMs / More album space / Advanced printing & mass image saving
Reply
 
Thread Tools Search this Thread
Old 11th January 2020, 09:31 PM   #2901
pos is offline pos  Europe
diyAudio Member
 
pos's Avatar
 
Join Date: Feb 2008
Location: Paris
Quote:
Originally Posted by loommidom View Post
Hello guys,


I am a german student and I am currently working on a Multiroom speaker system, which should provide a room correction function. Therefore I need to know a few things about the functionality of rePhase, so it would be great, if some of you could answer me some questions about it.

First of all, I am curious about what happens to the target response, when I increase one particular frequency in rePhase by hand, because when I do so, not just the amplification of the one particular frequency increases, but several surrounding amplifications increases as well. So, first of all i guess this is the case, because rePhase just calculates the expected target response and rePhase expect the amplification of the surrounding frequencies to increase as well. Now, I wonder why this is the case so, why do the surrounding amplifications change?
And second of all, I wonder how rePhase calculates the expected changing of the surrounding amplifications. I would it expect to be a linear interpolation, but I am not sure. Also interesting to know would be how big the area around the increased amplification is, which rePhase also changes and how rePhase determines how big this area is.

Thanks in advance.
Hello loommidom

By "surrounding amplifications change" I presume you mean that when applying an EQ point to a given frequency you see a sort of bell curve around that point, so that surrounding frequency response is also affected.

This is how PEQ work: you set a frequency, a gain, and a Q (quality) factor that will dictate how sharp the correction is (ie how surrounding frequency points will be affected).
You don't want to have discontinuities in the response (and that is impossible in practice anyway), and using this copes well with how response anomalies (that we want to correct) also manifest themselves.

As for how this shape is calculated, this is based on conventions and formulas such as constant Q EQs, proportional Q EQs, etc.
The magnitude and phase frequency response can be calculated from the their biquad implementation for example.

The time response is also worth looking at (impulse), as Q selection will have a direct impact on it.

Hope this helps
__________________
2019-01-16: rePhase 1.4.3
  Reply With Quote
Old 12th January 2020, 09:16 AM   #2902
chebum is offline chebum  Poland
diyAudio Member
 
Join Date: Apr 2018
Location: Warsaw
Quote:
Originally Posted by pos View Post


Yes, you can do FIR correction with zero delay (that is what I am currently running on my speakers).
Hello Thomas,

Can you describe that in greater detail, please? As far as I understand, we need at least N samples of the input signal to make a convolution. Where N is a number of taps of the filter. Even if the filter impulse peak is moved to the beginning of the filter file, we still need to accumulate enough input data to be able to make the first convolution.
  Reply With Quote
Old 12th January 2020, 05:35 PM   #2903
emailtim is offline emailtim  United States
diyAudio Member
 
Join Date: May 2005
Location: USA
Quote:
Originally Posted by pos View Post
Hello emailtim

Interesting use of this feature, thanks for sharing

As for other uses, I know some people have been using multiple banks (all of them even!) for a single correction, trying to refine the correction to the fraction of dB. I cannot say I support this kind of use as over-correction can cause more harm than good (eg correcting measurement artifacts, or position-dependent defects like comb filtering), but if done based on a reliable (set of) measurement(s) and with good analysis then it is probably a valid strategy if one has the time and patience to do so

I tend to only use a limited number of EQ points, but spread them across banks depending on the target of the correction (anechoic response, room adaptation, loudness compensation, driver to driver variation correction, etc.).

One thing that could help in your case (and mine as well) would be to be able to give a different title to each bank, to explicitly explain what it's intend is.
Right now the only way of documenting the use of each bank would be in the global "note" entry in the general tab.
I have also used the extra banks to try a second or third pass of correction. I tend to use "cuts-only" corrections and 17 PEQs doesn't appear to catch all peaks on the first pass.

I first tried to AutoEQ below and above Schroeder separately to allocate up to 17 PEQs for each AutoEQ (34/pass) but this method had the problem of not properly correcting across Schroeder. The AutoEQ appears to need an "anchor point" on the other side of Schroeder. That is when I decided to try the duplicate file with deletion method which has the "anchor point" on both sides of Schroeder. I can still do a second pass if desired.

My even/odd bank number scheme is my form of organization or documentation to keep track of what each bank is for.

Last edited by emailtim; 12th January 2020 at 05:49 PM.
  Reply With Quote
Old 13th January 2020, 11:45 AM   #2904
loommidom is offline loommidom
diyAudio Member
 
Join Date: Jan 2020
Quote:
Originally Posted by pos View Post
Hello loommidom

By "surrounding amplifications change" I presume you mean that when applying an EQ point to a given frequency you see a sort of bell curve around that point, so that surrounding frequency response is also affected.

This is how PEQ work: you set a frequency, a gain, and a Q (quality) factor that will dictate how sharp the correction is (ie how surrounding frequency points will be affected).
You don't want to have discontinuities in the response (and that is impossible in practice anyway), and using this copes well with how response anomalies (that we want to correct) also manifest themselves.

As for how this shape is calculated, this is based on conventions and formulas such as constant Q EQs, proportional Q EQs, etc.
The magnitude and phase frequency response can be calculated from the their biquad implementation for example.

The time response is also worth looking at (impulse), as Q selection will have a direct impact on it.

Hope this helps

Yeah, this helps a lot and i guess all my questions are answered. thank you very much pos!
  Reply With Quote
Old 13th January 2020, 10:42 PM   #2905
pos is offline pos  Europe
diyAudio Member
 
pos's Avatar
 
Join Date: Feb 2008
Location: Paris
Quote:
Originally Posted by chebum View Post
As far as I understand, we need at least N samples of the input signal to make a convolution. Where N is a number of taps of the filter. Even if the filter impulse peak is moved to the beginning of the filter file, we still need to accumulate enough input data to be able to make the first convolution.
For a real-time direct convolution you simply need to calculate the next sample, and the accumulation effect on the N coming ones that you will later be using. The convolution process itself does not imply any delay.
__________________
2019-01-16: rePhase 1.4.3
  Reply With Quote
Old 17th January 2020, 04:15 AM   #2906
gyfhgyfh is offline gyfhgyfh
diyAudio Member
 
Join Date: Feb 2015
How can I mix two or more "ir wave" files into one wav ?
  Reply With Quote
Old 17th January 2020, 09:49 PM   #2907
pos is offline pos  Europe
diyAudio Member
 
pos's Avatar
 
Join Date: Feb 2008
Location: Paris
You mean adding one correction over the other?
You have to either convolve one with the other (using sox or similar tools) or if both FIR where generated with rephase simply try to integrate settings from both corrections into a single one.
__________________
2019-01-16: rePhase 1.4.3
  Reply With Quote
Old 17th January 2020, 09:53 PM   #2908
pos is offline pos  Europe
diyAudio Member
 
pos's Avatar
 
Join Date: Feb 2008
Location: Paris
I don't know if there are any E-mu SP1200 aficionados around here, but I have been playing with the javascript webaudio API lately, trying to emulate its sound:

SP12k - SAMPLING PERCUSSION

It is a web application, running in chrome desktop, firefox desktop, and android devices (although latency might be quite high there depending on the device and android version).
ios and safari are not supported for now, but it works with chrome on macos.

I am pretty satisfied as far as sound is concerned (close to indistinguishable from the original on the sound comparisons I have tried), but I still have a lot of work ahead of me with functionalities (sequencer, sample editor, mixer, ... filters, etc.).

rePhase, a loudspeaker phase linearization, EQ and FIR filtering tool-sp12k-0-1-3-jpg
Attached Images
File Type: jpg sp12k 0.1.3.jpg (431.5 KB, 370 views)
__________________
2019-01-16: rePhase 1.4.3

Last edited by pos; 17th January 2020 at 10:04 PM.
  Reply With Quote
Old 19th January 2020, 08:57 PM   #2909
krunok is offline krunok
diyAudio Member
 
Join Date: Apr 2018
Quote:
Originally Posted by mitchba View Post
If one models the phase response of an "ideal" speaker, it is similar to the blue curve in the chart below. Of course, it depends on the low frequency cutoff and roll off slope for any particular speaker system, but if you play with different low frequency box alignments, the phase shape is more or less the same.


Click the image to open in full size.

The red and green traces are my speakers measured at the listening position some 9 to 11 feet away. I used FDW and psychoacoustic smoothing in REW as it closely matches the correction settings used for the DSP FIR filtering.

I have experimented with just linearizing the phase in a 3 way system with a passive XO and must say I did not notice a huge improvement when I switched the filter in and out while listening to music in real time. However, when I time aligned the drivers using linear phase digital XO and then corrected the excess phase, I noticed a considerable improvement in the "depth of field" in listening. Almost like the sound went from 2D to 3D.

I am wondering if this is the same sense that wesayso and others with line arrays are hearing as these types of speakers are inherently time aligned and have no XO to mess with different time arrivals and phase...

Thoughts?
Hi Mitch,

this is probably not the measurement of the last version of your filters but I choose to comment anyway just to draw attention to one common problem that is easilly overlooked.

Your phase response look very neat, except for the 40-80Hz region in which there is more than 100 deg of difference between phase of left and right speaker which will cause a cancellation of response in that frequency region. That cancellation won't of course be visible in each speaker response but is easilly visible if you measure response of both speakers. But as I said, I'm pretty sure that in the meantime you polished phase response of your speakers.

I had the same issue in the same region and was able to correct it via phase EQ with rePhase. Here it is how my graphs are looking. measured from LP (4m from the speakers, FDW of 12 cycles and psychoacoustic smoothing):

Click the image to open in full size.

Click the image to open in full size.

Click the image to open in full size.
  Reply With Quote
Old 20th January 2020, 07:12 AM   #2910
krunok is offline krunok
diyAudio Member
 
Join Date: Apr 2018
Quote:
Originally Posted by fluid View Post
I wonder which is more "correct" then
Both are "correct". Applying FDW is essential with phase correction to filter the refletions. Although the graph will be a little "choppy" for the phase itself you can go solely with it. But if you try to display GD based on that "choppy" phase graph you would see that using some smoothing (like psychoacoustic or 1/6 octave) together with FDW is also beneficial.

Quote:
Originally Posted by fluid View Post
In my system I have found that using a spatially averaged measurement across my couch in a similar way that SwissBear has written about here before is very useful to base a correction on.
Can you please post a link to his post where he described it?
  Reply With Quote

Reply


rePhase, a loudspeaker phase linearization, EQ and FIR filtering toolHide this!Advertise here!
Thread Tools Search this Thread
Search this Thread:

Advanced Search

Posting Rules
You may not post new threads
You may not post replies
You may not post attachments
You may not edit your posts

BB code is On
Smilies are On
[IMG] code is On
HTML code is Off

Forum Jump

Similar Threads
Thread Thread Starter Forum Replies Last Post
FIR linear phase plugin for MiniDSP? diyjb01 miniDSP 17 9th June 2016 02:35 PM
FIR filter design tool for Loudspeaker magnitude equalization ttmusic Software Tools 3 24th May 2013 09:30 PM
FIR Filtering experiences Olombo PC Based 8 10th February 2013 04:45 PM
AVX based FIR VST, crossover / EQ / DRC and delay KOON3876 PC Based 97 26th November 2012 08:18 AM
Phase EQ using FIR filters Grasso Multi-Way 2 2nd July 2003 11:37 PM


New To Site? Need Help?

All times are GMT. The time now is 12:14 PM.


Search Engine Optimisation provided by DragonByte SEO (Pro) - vBulletin Mods & Addons Copyright © 2020 DragonByte Technologies Ltd.
Resources saved on this page: MySQL 14.29%
vBulletin Optimisation provided by vB Optimise (Pro) - vBulletin Mods & Addons Copyright © 2020 DragonByte Technologies Ltd.
Copyright ©1999-2020 diyAudio
Wiki