rePhase, a loudspeaker phase linearization, EQ and FIR filtering tool - Page 45 - diyAudio
Go Back   Home > Forums > Loudspeakers > Multi-Way

Multi-Way Conventional loudspeakers with crossovers

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 8th July 2013, 02:02 PM   #441
more10 is offline more10  Sweden
diyAudio Member
 
Join Date: Feb 2011
Location: Solna
Thanks pos.

I have made a small program to convert the text file to a c array, you can postpone that improvement for some time.

The FIR accelerator is in time domain, so the window thing doesn´t really apply I believe. Or does the window parameter affect the generated impulse?
  Reply With Quote
Old 8th July 2013, 02:28 PM   #442
pos is offline pos  Europe
diyAudio Member
 
pos's Avatar
 
Join Date: Feb 2008
Location: Paris
Quote:
Or does the window parameter affect the generated impulse?
Yes it does.
In any case, you should try different settings (number of taps, sampling frequency, corner frequency, windowing algorithm, centering, ...) and see if what you obtain for the result curves (in red): what you see is what you get.
__________________
No loudspeaker system even approaches real life so there is plenty of room for interpretation - Greg Timbers
  Reply With Quote
Old 5th August 2013, 10:59 AM   #443
diyAudio Member
 
Join Date: Feb 2009
Location: UK
A few months ago, the use of rePhase for correcting the phase effects of bass reflex alignments was discussed, concluding that it would be possible to make a bass reflex speaker behave like a sealed speaker (I think I even stipulated phase and transient response in a question) by reversing the measured phase shift near the port resonance. Is this really true?

If a bass reflex port augments the movement of the cone with an in-phase resonance delayed by one waveform period (referred to as 'time smearing' by Wikipedia's bass reflex article), it looks as though whatever we do to undo the delay in the port's output must also lead to premature output from the cone. Theoretically we can deconvolve the driver's signal with whatever it has been convolved with, but only if the system is minimum phase. Is this actually an example of a non-minimum phase system, so it is not possible to remove the time smearing effect of the bass reflex port?

Last edited by CopperTop; 5th August 2013 at 11:01 AM.
  Reply With Quote
Old 5th August 2013, 12:18 PM   #444
pos is offline pos  Europe
diyAudio Member
 
pos's Avatar
 
Join Date: Feb 2008
Location: Paris
I had the same concern and both John K and kgrlee said it was a minimum phase phenomenon:
rePhase, a loudspeaker phase linearization, EQ and FIR filtering tool
__________________
No loudspeaker system even approaches real life so there is plenty of room for interpretation - Greg Timbers
  Reply With Quote
Old 12th August 2013, 06:34 PM   #445
diyAudio Member
 
Join Date: Feb 2013
I got a question:
checking the phase plot of the generated IR i see that with many different windowing functions the phase in the stopband is jumping back and forth between 0 and 180°.

I think this could be related to Gibbs phenomenon: the undershoots of amplitude ripples are reported as positive in the pseudo magnitude graph but with inverted polarity....is that right? (just guessing)

why some windowing functions doesn't have this behavior? for example triangular, or barlett-hann are flat phase even in the stopband.

Complex window is so strange... can't figure out the phase behavior in the stopband...

thanks to anyone that will make me scratching my head

Click the image to open in full size.

Click the image to open in full size.

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 12th August 2013, 07:16 PM   #446
diyAudio Member
 
Join Date: Feb 2013
*i meant "stop scratching" in the previous post...
...furthermore if anyone has problems in expanding the thumbnails...here are the full sized

Click the image to open in full size.
Click the image to open in full size.
Click the image to open in full size.
Click the image to open in full size.
Click the image to open in full size.

Last edited by luigichelli; 12th August 2013 at 07:21 PM.
  Reply With Quote
Old 14th August 2013, 05:42 PM   #447
pos is offline pos  Europe
diyAudio Member
 
pos's Avatar
 
Join Date: Feb 2008
Location: Paris
Hi luigichelli

Thanks for these interesting illustrations

As a side note, it should be noted that only 384 taps are used, and the target slope is quite steep: this side of the "reject low" shape is far sharper than the advertised 48dB/oct. A shallower target curve, LR 48 or 96dB/oct for example, would give lower ripples for the same number of taps.
To compare ripples with different windowing algorithms you should probably stick to LR shapes as those are more common and predictible (the "reject low" hipass is almost brickwall half an octave under its cutoff).
Also if you really want to compare windowing algorithms you should defeat the iterative optimization stage (optimization: none) as it will iteratively change the amplitude target target to make the result curve match the initial target better and better, but will sacrifice phase to do so, and also tend to rise the ripple floor (optimization concentrates of the > -40dB SPL range, but this will be settable in the next version).
With the current version you will obtain lower ripples when setting optimization to "none".

Regarding the "complex" windowing, it was meant for phase-only corrections before the "float" centering option was introduced, to avoid amplitude ripples in the HF (but they still appear in the impulse...).
It splits the audio band in two and apply different windowing on the LF and HF, and also different windowing on the left side and the right side of the impulse...
It should only be used when the "int" centering method has to be used (and generally it should not), and preferably for phase corrections only...
__________________
No loudspeaker system even approaches real life so there is plenty of room for interpretation - Greg Timbers
  Reply With Quote
Old 17th August 2013, 10:41 AM   #448
pos is offline pos  Europe
diyAudio Member
 
pos's Avatar
 
Join Date: Feb 2008
Location: Paris
Hi all

Just a quick update to bring to your notice a nice tutorial recently put together by the miniDSP team:
The rePhase FIR tool | MiniDSP

By the way, following this advice, could a moderator please change the title of this thread as follow:
rePhase, a loudspeaker phase linearization, EQ and FIR filtering tool

Thanks!
(yes, I know this thread should not be in the multiway section, but please leave it here, as it is the only section I really follow and have my habits with )
__________________
No loudspeaker system even approaches real life so there is plenty of room for interpretation - Greg Timbers
  Reply With Quote
Old 17th August 2013, 10:49 PM   #449
kessito is offline kessito  Netherlands
diyAudio Member
 
Join Date: Sep 2010
Location: Amsterdam
Dear pos,
Thanks for making this Great tool available!
I have one request, it would be very Nice to have the option to export the filters also als minimum phase (doable via hilbert transform or cepstrum method) . This way we could listen to the differences.

Thanks again,
Kees
  Reply With Quote
Old 18th August 2013, 06:53 PM   #450
diyAudio Member
 
Join Date: Jan 2006
Location: grenoble
Hi,

to export minimum phase impulse,choose the inv time in the general tab.

Click the image to open in full size.
  Reply With Quote

Reply


Hide 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
Trackbacks are Off
Pingbacks are Off
Refbacks are Off


Similar Threads
Thread Thread Starter Forum Replies Last Post
FIR linear phase plugin for MiniDSP? diyjb01 miniDSP 13 7th February 2014 02:24 AM
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 09:04 PM.


vBulletin Optimisation provided by vB Optimise (Pro) - vBulletin Mods & Addons Copyright © 2014 DragonByte Technologies Ltd.
Copyright ©1999-2014 diyAudio

Content Relevant URLs by vBSEO 3.3.2