Phase Match Your Speakers with AllPass Filters
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- Опубликовано: 3 авг 2024
- Taking the time to properly correct for the phase issues provides a truly worthwhile sonic benefit and an optimized listening experience. Phase correction is complex, and I simply don't have all the answers, but in this tutorial, I’ll pass along original techniques validated through vigorous testing and personal listening experience - video resolution: 2,160 x 1440 (60fps).
Timeline:
00:00:00 Introduction
00:02:30 Tutorial Steps
00:03:17 Measuring speaker phase response accurately
00:05:04 Calibration microphone phase correction
00:09:10 Generating excess phase response
00:10:48 Correcting speaker crossover and box and port phase shifts with rePhase
00:16:03 Correcting peak energy time with all pass filters
00:24:25 Checking filter for pre echo
00:29:33 Aligning speaker phases with each other
Free tool REW Early Access Build used in the tutorial:
www.avnirvana.com/threads/v5-...
Free tool rePhase v1.4.3 used in the tutorial:
rephase.org/
#roon "room EQ wizard" "filter design" "phase inversion" #audiophile #impulseresponse #convolution #filters #rew #roomcorrection #calibration #fouriertransform #fourier #equalization #equalizer #hometheater #cinemasound #soundsystem #dirac #dolbyatmos #speakertest #impulse #phase #hifi #hifiaudio #bass #bassboost #graphicequalizer #hometheater #impulse #iir #fir #lowpassfilter #phase #rephase #sota #viral #audio #rew “speaker placement” “speaker matching” “speaker impedance” “double bass array” #subwoofer “subwoofer placement” “phase correction” “crossover” "speaker port measurement" #systemdesign #systemtuning #livesound #liveaudio #sounddesign #audioengineering #soundoptimization #liveaudio #concertsound #audiomath #systemtuning #proaudio #soundengineering #soundtech #sounddesigner 
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I just did this to rephase my 4k speaker while connected to subwoofer as well, the result is amazing !!!
And now get a DSP to do proper crossover and time alignment from the mains to the sub. The difference is insane!
Excellent tip about making a minimum phase version of your calibration file. I've been trying to match the measured phase response of my EQ'd drivers to the minimum phase version of the target shape. They should be the same, but they weren't quite, especially the tweeter. Now they are, and everything makes sense. Awesome!
Glad it was helpful!
That AI voice introduction was interesting and effective, but its nice to hear the human. I slow this down to 0.75 speed in RUclips, makes it so much easier to follow.
Thanks for your valuable comments.
Thank you for the demonstration. Makes me want to tinker with phase filters as well.
It's worth the effort if done right.
with gratitude
I am blown away,
Could you do a new, up to date, audyssey tutorial with all the new techniques that you are showing please?
I will do very soon and also with Multeq-X!
Awesome!!
Appreciate you. Thanks!
Many thanks!
Your accent sounds like the accent of someone who knows what they are doing. Don't you ever make fun of it again. Just focus on getting the message across. (No need for that dumb voice in the beginning). Thank you for doing these
Your welcome!
Thank you for this explanation! You are really cool for what you know and share, and your voice is much better, i was happy when it appeared.
Is there a way to know the crossover frequency without the techincal data?
If you can look inside for the values of the capacitors, resistors and inductors, there're online tools to determine the order and filter frequency of the crossovers but it's quite a lot of work.
Thanks!
Welcome!
Thanks for taking the time to share your experience with us. I have learned a lot watching your videos. However, I think that in this video, you made a statement that is not completely true. You said that if the tone of the pink noise changes when moving left to right, it means that the speakers are not completely in phase. I believe that the tone will change even with a theoretically perfect speaker in a perfect room since moving your head left or right means that the sound coming from one speaker will arrive sooner. This continuous change in time produces changing peaks and dips in the frequency response. One can easily replicate this effect by playing the same track duplicated in a DAW and slightly delaying one of them. The phaser and flanger effects are obtained using various implementations of this effect. If I'm wrong, I would really like to understand why. Thanks!
www.linkwitzlab.com/Store/phantom-image.htm
Thanks for another awesome tutorial as usual. One question. When you say the the mic should be placed horizontally, should it face the center of the two speakers (at tweeter height) for all the measurements? Or should they be pointing to the respective speaker that is measured? Also what about the measurements that are made from a position not symmetrically between the two speakers? Where should the mic point in this case? Thanks
That's a tough debate and a question of how precisely you want the measurements to represent MLP and how much you are willing to compromise high frequency measurement accuracy. Assuming the mic cal file was produced with a professional calibration tool then pointing it directly to the speaker will result in the most accurate measurement but you will never be able to have a common left and right speaker measurement for the exact central listening position. I ended up taking one measurement pointing forward at the exact centre for left and right and left+right then rotating the mic stand arm slightly to the left (about where my left ear will be when I sit at the LP) and measure the left speaker and do the same for the right speaker. That works for a single central point very well but is it optimal? I doubt it. The angle is still not directly towards the speakers at ear locations. Fortunately, this really only effects very high frequencies to which for the sake of ultimate clarity, you are very often better off not applying any filters anyway.
@@ocaudiophile thanks for the reply. I’ll try it like the way you described 👍🏼
Hi OCA, I was deeply inspired after watching thev "REW (Room EQ Wizard) Top Tricks: Convolution video." I'd like to try DRC by Rew. However, I've also noticed your new workshop videos and other phase match videos. And due to the mass of information, I'd like to ask which video I should refer to for the most suitable approach. Currently, I have a 2.1 channel system, and my ultimate goal is to generate a calibration file that I can use with Roon for convolution. Thank you!
Inversion is a very quick and quite efficient technique to create convolution filters and use in Roon but it doesn't include and phase correction. I'd suggest to watch workshops I & II since you've already watched this one. Also the latest Trinnov vs ART vs REW VBA video includes a fancy virtual bass array filter.
@@ocaudiophile Thank you for your reply. I'll watch and digest the content of these videos first, and if I have any questions in the future, I'll come to you for guidance.
@ocaudiophile Thanks for sharing your new techniques. Appreciate it!
I think my speakers have the xo and box excess phase adjusted in the crossover as any adjustment from where it is makes the phase less near 0. There Fusion 8 MTM Towers designed by Jeff Bagsby who is a pretty well known speaker designer so that wouldn't surprise me.
The Subsonic filter "2nd order pk + 2db" @45hz really helps my phase response. I'm using a Minidsp 2x4 hd so I need to use biquads. I found a Minimum-Phase Filter with a Compensated 2nd Order all pass with .79q and a normal first order all pass at the frequency you need almost exactly replicates it. Looks like you can just play around with Minimum-phase filters until you get the right combo.
Question:
Both my speakers have excess phase that starts at 0 drops to 180 degrees out of phase and then continues dropping from the top back to zero phase. It's not at the same frequency. One starts dropping at 110hz and finishes at 250hz and the other the other 350hz and 500hz but its basically the same. If i use a 2nd order all pass with a q of 2.7 at 180hz on the 110hz to 250hz speaker it brings the phase to be a straight line at 0 with no drops. Is this something I should do? I have 3 quadratic residue diffusers at 2nd reflections if that could make a difference.
Jacob
Difficult to say without seeing the actual graphs really...Can you share your mdat here? YT will show me Google drive links.
@@ocaudiophileFor sure. Thank you.
drive.google.com/file/d/1-jFARGQVwopAJKz8JzoTN6j1qyBUqmCn/view?usp=drive_link
I wasn't quite right with the frequencies but you should see what I'm talking about.
My room and setup are constantly changing so I'm just making sure I can tune it when it's ready so these aren't final measurements.
You should generally prefer 2nd degree phase filters with Q values of 0.707, 1, 1.414, 2 (square roots of 2). I made a quick phase correction filter for your average EP below:
drive.google.com/file/d/1E7cQ6cl_tjCjoypplGzo92jB-LsMLu5Z/view?usp=sharing
@ocaudiophile Wow, That looks a lot better than what I came up with. I'll play around with it some more and try to figure out what/why you did it. Thanks for taking some time to help me.
The EP text you uploaded is more jaggedy and chaotic looking. Mine that I produce follows yours exactly but is cleaner looking. Can't figure out what you did differently.
It's the EP of the new "dB+phase average" in REW of left and right speakers after removing their IR delays @@jacobtravis2847
Excuse Dr Gur, to obtain the final filter do we have to multiply XO filter times the impulse found with the compensate stage ( in your example r271pc)? Thank you
That will be the final phase filter, yes. You can multiply as many as you need and combine filters.
Hi OCA, another question. I noticed that I can get my phase response in RePhase fairly flat by using the equalizers in the Paragraphic Phase EQ tab. How good or bad are these parametric equalizers in terms of delay and ringing? Is it advisable to use them at all? Thanks
You can use them comfortably as long as you don't go below 500Hz with higher than Q=1 and +-45 degrees. Pre-echo problems are very easy to get there. For HF most phase correction is a bit uselss as they are inaudible and also hard to measure correctly.
Grazie.
🤩
Dear Serkan, before going on with our discussion (if You are still willing to help), I would like to submit to Your attention another measurement, that represents the "historical" positioning in my room (P0). The measurements we analysed until now are related to a "second choice" positioning, much closer to the front wall (P1, 120cm).
Today I brought the speakers back to P0. Tomorrow I will repeat the measurements with utmost care. Then I would ask for Your help once more (last time I promise).
The fact is that P1 considerably reduces the dip, but it is broader in band.
P0 has a huge dip, but narrower band. After studying Your material, I think something can be done there...
In fact, P0 is farther away from the front wall (1.8m) and sounds much better in terms of space and imaging. Psychoacoustic smoothing seems even favorable to P0 (I noticed today)
Have a nice evening / holidays.
Sincerely
sure thing, share your measurements when done!
Hello Dr Gur, I followed your video "High Fidelity Digital Room Correction with REW & rePhase" and at the end you specify the current video to complete the procedure and improve the phase correction. So instead to start from L0 /R0 as stated here, i have to start from L3/R3 of the previous video?
This video is for advanced phase correction. It's not a direct continuation of a previous tutorial.
You can correct only the frequency response or frequency and phase response. All filters will combine when multiplied (trace arithmetic AxB).
@@ocaudiophile thank you for the answer.So Is not possibile to restore the original procedure of phase correction? I used it few months ago and i was very, very satisfied of the results.Now I changed Place and position of the speakers and i would Like to use It again.
most filters will need to change after new placement. Watch the workshop videos I & II, they cover most aspects of DRC
Hello OCA I have a question. How to apply the smooth phase to your methods correctly? And is it necessary to align all the acusika in the system by phase or only the fronts?
Any number of speakers would benefit from phase correction but you can't do phase correction with Audyssey. If you're running a HTPC based surround sound system, you can apply phase correction for up to 7.1 channels for free but even that will not be able to apply phase correction to Atmos speakers.
Some Minidsp models are capable of low resolution FIR filters and a gentle phase correction can be applied to speakers with them, too.
Hello! In this video, to create an XO filter, you completely changed the measurement settings. Is this only true for XO? Do I need to use the same settings as in the previous videos to create other filters? And do long 4 sec measurements with them?
For only XO correction, shorter measurements can be more precise if you are using USB mic.
Another question. Should one time align the left and right speakers at MLP before going ahead with the phase corrections? Or is that not necessary?
You need to correct excess phase responses and to extract that correctly, you may want to "remove IR delays" from the measurements first.
So, would you recommend this process in place of the phase correction section of your Digital Room & Speaker Correction Workshop video?
Yes, this is the latest technique.
@@ocaudiophile I might misunderstand the impact of these phase corrections, but wouldn't it make the VBA filter kind of useless if I correct the phase afterwards in the frequency region where the VBA filter is also affecting the phase?
VBA usually improves the phase response and phase correction improves some more. Don't forget that original phase response is incorrect almost everywhere. @@_koXx_
Thanks again ! Do you have threads on ASR or another forums to discuss these techniques ? I did phase measure my speakers but didn't had phase inversions on the XOs for my floorstanders but on my bookshelves they were clearly there. Anyway for my floorstanders I did had a couple of inversions that I managed with all-pass filters and they do improved the phase response, who knows why I did not get inversions for my floorstanders but all-pass did improved the system anyway 🤷♀
Crossover and box phase correction will improve every speaker response albeit some active ones. And if you don't need to EQ for a large area ie single person seat, phase inversions also can do miracles to regional delays. ASR is a bit of a mess, but this subject is being discussed just now in the below link:
audiosciencereview.com/forum/index.php?threads/minimum-phase.6065/page-4
Thanks @@ocaudiophile I'll try to measure one floorstander alone with MIC at 50-100cm to see if I catch the XOs frequencies (35, 350, 2700 on my speakers)
EDIT: It seems my MiniDSP Dirac is smoothing the phase on the XOs points. I believed it didn't correct the phase. OTH do you recommend another forum beside ASR mess 😁
For REW related issues & questions AVNirvana, for MultEQ app, etc. Audyssey AVSForums has quite sophisticated discussions.
@@ocaudiophile Thanks. I've been on those forums. My question is more oriented if you maintain a thread for each of your usefull videos in order to follow other and yours expiriences. I remember you did on Roon or MiniDSP forum I think for one of your videos. Usually on a forum thread comes a lot of tips for beginners like me. For example even if I removed IR delay for EP it did had a delay which fool me to chase a phase inversion on high frequency which was only because 30us of delay on the EP IR signal. After a few tries I remembered an older video from you showing the efects of IR delay on high frequency phase 😆
Looking for a little clarification, around 4:30, were the sets of multiple measurements for L0 and L1 each vector averaged before vector averaging L0 and L1? Or some other averaging algorithm each to the set of L0 and L1 measurements? Or the full set of all L0 and L1 measurements vector averaged all at once? (I see some differences between each method)
Vector average all measurements for the same speaker but make sure to cross correlate them to the measurement at the MLP first. At this stage, you should eliminate odd measurements out. Do the same for the right speaker. To obtain the original stereo response you need to create the filters for, you should also cross correlate the VA of left and VA of right speaker to each other (or remove IR delays from them which should do the same thing) and vector average them. Lately, I also generate dB & Phase average of "unsmoothed" VA of left and VA of right to determine the common "auto" target level of left and right speakers. Other averages auto targets are either too low or too high IMO.
@@ocaudiophile Awesome, I got a first try at creating a phase correction filter with allpass filter yesterday, didn't get to test it yet (took a while to figure out how to get CamillaDSP running on my Windows machine). Predicted phase response was massively improved, even found a spot at ~180Hz with a sharp phase shift and was able to correct that with no predicted pre-ringing. Going to finish up this video and your workshop videos today and apply some filters :)
Hey OCA, I have a question and the phase and how positive and negative gains affect the phase. Are there benefits with phase to only using negative numbers when you EQ? Same with the inverted speaker over the target curve. If I make all my numbers negative, are there improvements or more consistency with phase?
EQ filters are minimum phase filters and have only a minimal local effect on the phase response. I've seen boosting filters improve phase and cutting filters to make it worse (although still minimal) but more often cutting filters are harmless to phase if not better. The best phase correction is a general crossover phase delay correction sepcific to that speaker and unfortunately, this cannot be done with frequency EQ filters.
Thanks for the feedback, I appreciate it.@@ocaudiophile
Whats your opinion on boosting vs cutting? Does it sound cleaner to cut or in the end, it's all relative?@@ocaudiophile
Up to 5dB boosting filters up to 200Hz doesn't reduce clarity. Beyon that frequnecy only cuts.@@hdmoviesource
Hi OCA, yet another question. Should we apply FDW while exporting the EP version of the measurement for rePhase? I remember seeing/reading somewhere that using FDW captures phase info better as it takes the room reflections out of the measurements. Thanks again.
No, don't. Just make sure you remove IR delays from the response before you generate EP version and also include mic cal effects if you have generated a min phase version of your mic cal file. You can export a 1/1 octave band or psychoacoustic smoothed version of your EP along with the unsmoothed if you get too confused in rePhase with the EP result.
Hi @@ocaudiophile Unlike your phase graphs which are clean and quite linear, mine seem to have a lot of rotations, even in the PA smoothed versions. They have at least 3 rotations until 200Hz and then one more just after 500Hz. Is this bad?
remove IR delays before generating excess phase version
@@ocaudiophile I did... still the same...
Then check its 1/1 octave smoothed version to find the main XO filter orders.@@paulsoumya
Hallo OCA,
leider ist auch hier ab 2:56min kein deutscher Untertitel vorhanden!
Bei dem anderen Video, wo ich den gleichen Fehler mal nannte, scheint es auch noch nicht zu funktionieren.
Gilt dieses Vorgehen auch für Subs um die Filter in JRiver zu nutzen?
Ein andere generelle Frage: Kann ich bei einem Yamaha A6a auch an die Messdaten kommen um diese zu bearbeiten?
Ps.: Beantwortest Du auch PN in Foren? Hatte vor einiger Zeit mal was geschrieben ;)
Es tut mir leid, dass ich sie nicht aktualisieren konnte, aber obwohl ich in Deutschland lebe, spreche ich kein Deutsch (dies ist von Google übersetzt) und ich habe DeepL für die Übersetzungsskripte verwendet, die entschieden haben, dass ich mein Kontingent erreicht habe und dafür bezahlen muss weitere Dienstleistungen. Es ist keine einfache Übersetzungsaufgabe, Sie müssen jeden Satz mit den richtigen Zeitstempeln unterteilen. Das Verfahren ist in JRiver anwendbar, es verfügt über eine sehr leistungsstarke Convolutoon-Engine, das Format ist etwas anders, Sie müssen die Wave-Dateien in Konfigurationstexten finden. Weitere Informationen finden Sie hier: yabb.jriver.com/interact/index.php/topic,136877.0.html Ich muss diese PM verpasst haben, welches Forum wusste das?
@@ocaudiophile
Ah, I didn't know that. I thought German was your language too.
I wrote to you on AVNirvana in German, but I see that there were some errors there.
Then I have to work my way through it without subtitles.
I meant, can I also use this procedure for the subwoofers to use in JRiver? Or is a MiniDSP 2x4 HD the better choice here?
The other question related to the data that you pull from the Marantz and then edit. Can I also get this data from a Yamaha A6a? I haven't been able to figure anything out yet.
I think I understood your question now. Unfortunately JRiver or any other program for that matter, cannot apply convolution filters above 7.1 although most recent JRiver beta is capable of playing Atmos music files. Atmos signals need to go by HDMI passthrough and no filters are possible. The only option I know (and it's free) is Cavern which can redirect 2 of the 7.1 channels to heights and can do 5.1.2 but you need to convert each Atmos file first. Similarly, there's only one channel (.1) in the PCs for a subwoofer and for a multiple sub configuration you will need miniDSP or a surround receiver with multiple sub outputs.
The measurement data cannot be hacked from the receiver itself, it's generally encrypted with SHA. We can only hack them from the measurement apps like MultEQ Editor or MultEQ-X. If Yamaha has a calibration app, then it should be easy. I did it for Arcam couple of weeks ago.
@@neutronenflusterer9643
@@ocaudiophile
I see in JRiver's options that in addition to 5.1 and 7.1 I can also select 10, 12, 14, 16, 18, 20, 22, 24 and 32 channels.
There is also a tab with additional channels from 1-16, but I still don't understand what they are for.
I would go to my AVR via HDMI on the PC with JRiver. From there to my current 5.2 system.
Then I have to switch off the EQ on the AVR but I still use the distances and levels?
My AVR can handle 2 subwoofers separately, which is what I currently have. But the treatment is somewhat more limited. So if JRiver can't handle 2 subwoofers, I can't avoid a MiniDSP.
There is a Yamaha Media app, but I haven't tested what it can do yet.
I guess (only a guess really) these multiple channels are for studio audio interfaces but they wouldn't be able carry Atmos signal unless you buy a very expensive and limited to professionals Atmos rendering suite.
Hi OCA,
1. Is this method also suitable for subwoofers with MiniDSP 2x4 HD?
Or what approach do you recommend here?
No FIR filters are used in your older sub video.
2. Before this process, as shown in previous videos, the correct distance is first set using impulse, I assume?
3. I always find it difficult to understand why which settings are chosen such as sample rate or window width and where the number of taps come from (131072 at 1000ms width).
Or when is a first or second order all-pass filter chosen?
When do you take no smoothing or Phsy smoothing?
It would be nice if something about it were discussed in your videos!
Ps.: Where is your premiere from VBA+?😉
1. Yes but you need to do a lot fo optimization in rephase to cvreate the same filters with just 2042 taps (MiniDSP limit for 2 channels)
2. Correct
3. Noted ;)
VBA+ video has just been launched:
ruclips.net/video/CurDymVz7aI/видео.html
@@ocaudiophile ok Thanks! Then I'll probably do it the old way first😄
I can still remember that you significantly reduced the taps or FFT in a video through various settings, but I can't remember it exactly.
Does that mean I have 1021 taps available per output and sub?
The MiniDSP 2x4 HD should arrive in the next few hours.
What if i measured the speakers already with applied calibration?
In the listed measurements it doesn't show any calibration file applied, but it does so in the preference.
Would applying the additional phase calibration with zeroed out magnitude correction within the calibration file be the correct way?
In addition, as i adapt all this stuff for CarAudio it's normal to measure with a 90° calibration.
If the measurement in the list doesn't show a calibration file, it simply means it doesn't contain a mic calibration. You should manually browse to the correct file and add it to that measurement.
For atmos/surround HT measurements and I guess also in car measurements, you'll have to use 90deg cal file and vertical mic position.
@@ocaudiophile well… before I made the measurements REW asked me if I want to add a calibration for the mic i just connected and then i added the corresponding calibration before i made the measurements.
After saving and opening on a different computer the measurements look the same, but they don’t have a calibration attached to them.
When i add the calibration the frequency response is totally wrong afterwards compared to the measurement on the original computer.
Therefor i asked if it’s necessary to add the calibration back in when you made the measurement with a proper calibration file.
Ok, if you changed PCs in the meantime then it's more complicated. They might have cal files already embedded. @@Cathul
@@ocaudiophile problem is… it was the original minidsp cal file without phase information. Therefor i thought about generating the minimum phase version, nulling the frequency response part and add it to the measurement to correct the phase in the measurement. Does that sound like a valid idea?
If at all possible i would like to skip redoing all measurements. 😜
The delays caused by the filters and MiniDSP itself will change the phase response quite a bit. I'd remeasure with acoustic timing reference when minidsp is active and then apply phase correction to that and add the filter as FIR to MiniDSP.
So, do you recommend taking measurements using the 0-degree UMIK calibration file instead of the 90-degree one? If I use the 0-degree file, should I put the microphone horizontally pointed at the speaker I'm measuring or straight in the center?
You need to do this only for the central main listening position measurement. You can direct the mic towards the speaker for all other measurements, making sure you don't measure the other speaker with that mic angle. Or you can simply keep it forward and measure both speakers at each point and call it a day. The differences are minimal and only at the very high frequencies where you cannot EQ anyway.
@@ocaudiophile I already have the measurements with the file at 90 degrees. Do you think it's worth redoing them with the 0-degree one? Would the phase be measured more precisely?
@@BarileTixxoFilms You can create the MP version of your 90deg çal file and replace the çal files of measurements with it. That will surely improve results. But if you need even more precision at high frequencies, you might wanna remeasure.
After I made the Xo filter and the BOX on the EP file and any 2nd order all pass filters ..... I can also manually correct through the parametric phase EQ so as to make the phase as flat as possible?? From what I understand, if I work on the EP file there is no danger that it can cause pre ringing because I will always be above the minimum phase.... Have I understood correctly?@@ocaudiophile
@@BarileTixxoFilmsYou can use paragraphic phase equalizers but they can cause pre-echo with high Qs and low frequencies more readily than allpass filters. Attempting to correct for minimum phase doesn't mean there will not be ringing. You can only move the room walls to a certain extent with just phase equalization. The idea is to bring the room as close as possible to min phase with ring free filters.
I was worried for a second, I was like noooo where is the normal , what I presume, eastern european twang I’m used to hearing
😂😂😂
What is your opinion regarding Genelec’s GLM calibration system?
I have no experience but had a quick look. I respect Genelec as a brand and I am sure the correction system is powerful and efficient but I didn't see any technical data of the filter capacity and it doesn't seem to be doing any phase correction other then subwoofer distance calibration. You can always do better with free REW and a cheap calibration microphone if your player source is a computer.
No matter what I do. I cannot time align my left front height and my left rear height channels. I change the distance and it doesn’t move no matter what. All
Other speakers time align with no issues. What could I be doing wrong?
I don't really have a clue. If a speaker is selected as acoustic reference speaker then its impulse peak will always be at time=0 regardless of the distance to mic but that cannot be the case with your two different left side speakers.
@@ocaudiophile I use my front left as the timing reference. All speakers measure and adjust according to distance with it and line up exceptionally close to the reference. However those two speakers which are way closer to the main listening position always have an impulse directly to the left of zero and show about 14ft. However the distance is set to zero in my processor. I can’t take the distance number negative for any speakers. Even if I raise my reference speakers distance 15ft higher and then try to adjust it just stays put. The distance never moves on those two speakers. It’s weird. Unfortunately you can hear a difference in audio pan on those two speaker. I can’t run a manual calibration to fix it. Only Dirac can fix it. Kind of a bummer.
So best is to have actual measurements of each way in the speaker and know the correct crossover slope because of it? In my case i designed my own speaker's. with a similar crossover design to Jeff Bagby's Kairos speaker's. My crossoverpoint is at 1500hz. The tweeter acoustically rolls off from 3khz down with a 6db slope and below 1khz transitions to a 18db per octave rolloff. The woofer off course i made to do the same rolling off 6db per octave and then later with 18db. For this type of crossover i gues it's better to use the phase EQ option of this programm as this is far away from a standard crossover type.
You could still probably imitate it with a combination of 1st and 2nd degree allpass fillters. Use Q of 0.707 or 1.414 for 2nd deg, switch between compensate and normal modes.
@ocaudiophile,
Trying to keep up with you and check my work and see if I need to go back over the steps again.
If you were to Measure your L/R speakers together at the same time/position as the Left/right individual measurements were taken, Would the EP of the real world Left/right combined EP be the same as Vector Averaging the Left/right together?
Mine are almost exactly the same until about a 1,000hz and then my real world left/right measurements deviate radically from the L/R Vector Average EP I made by Vectoring The Left/Right together.
The Left/right individually have close to the same excess phase that drop from 0 EP at a 1000hz to -180 EP at 17,000hz. When I average them together for my L/R Vector Average it looks just like each individually.
The real world measurements EP looks about the same until 5000 hz where its EP stops dropping and levels out at -45 EP and stays there until 24,000hz. It seems like the Left/Right taken together have some positive summing effect that levels there excess phase out that doesn't show when I vector Average the Left/Right Together.
Your Stuff is Next Level. Thanks for doing what you do.
"Remove IR delays" before generating EP, that should sort the differences in the high frequencies...
@@ocaudiophile, That was it. Solved my problem quickly. Thanks!
Prefer normal voice to ai text to voice
Merhaba.
Özelden sizinle görüşmem mümkün mü?
Buraya email yazamamam. Bir google drive link eklenyin, bana gelir sizin email'iniz.
@@ocaudiophile drive.google.com/drive/folders/1SWWgeDiA9h2gh3O_S8IIa4yOux1IgvUb?usp=sharing
Hi Obsessive, i love you Channel, but i can dis not 😢, i surche for privat help for my system
Is it a stereo or surround system?
@@ocaudiophile i have 5.2.6 Atmos, with Marantz AV10 with MultiEQ X and Dirac Lizenz
Did you watch the Audyssey ART video:
ruclips.net/video/LwORN-tSPjk/видео.html
Hello Dr Gùr,
i'm trying to replicate your last work (excellent as usual) on my system., but i think i need a pair of tips.
in the last part of the video, when you explain how to align speaker phases with each other (you got only one freqency)
unfortunately i've got three frequecies for both speakers 236, 311 and 1800 Hz:
this because i think is was not possible figure out the exact XO parameters;
(i wrote the person who design my 2 way speaker (aliante mod. linea)
that told me they have not a classic scheme (?!), with 2500 Hz and first order filter, so 6dB/oct(!) i guess,
however from the measurement i did i alway see an "S" in the phase response near 2kHz, so i suspect it is mistake.
• is it ok to replicate the same procedure for all 3 frequencies?
• I did not clearly understand why and how counteract the double of the Q used ( clockwise counterclockwise?),
could you please tell me something more?
• can we add to these phase alignement filter with the Convolution with Inversion FIR?
If yes we should modify the settings of the windowing, etc?
Thank you as always for your patience and kindness.
If you share your measurements, I can have a look. Google drive links are not deleted
These are the files i worked on.
To adjust the first rephase stage with the excess phase copy of the average vector
(i named it impulse5, because a test a lot),
as you can see i set to nothing the XO frequency and slope,
put to 41 Hz the frequency of the bass reflex with Hi Q( i measured with mic inside the port),
and set a minimum phase filter mode compensate 0.7 Q at 2000 Hz,
these are the setting gave to me the best results.
the following measures are about i wrote you before.
I hope is clear.
Thank you.
drive.google.com/drive/folders/1Mclefm6g_MpWD_Ajw59nMraYTSt1tQYq?usp=sharing
@@ocaudiophile
These settings seem to work well for your speakers' XO&Box correction (Copy everything below and "Load from clipboard" to rePhase):
rePhase settings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Hello Dr Gur,
unfortunately even with your settings i ended with three frequencies adjusted for L and 2 for R Channel
R 220.1 311.87
L 234.37 310.44 1819.04 and the results are here
drive.google.com/drive/folders/1UTKFlBv6KAUZBBQ9mdIFEb45uklNiW-F?usp=sharing
i tried to add the "normal" version with double value of the "compensate",
but things went worse.
I think that likely will be pre ringing. don't you?
Thank you
@@ocaudiophile
Well, both seem to be creating pre-echo but you cannot knwo for sure without listening.
Grazie.
Dear Serkan, here You can find a folder with today's measurments and some notes in a dedicated text file. There are also a couple of pictures of the set up
drive.google.com/drive/folders/1TilbNjdFLVsAi793aWtSpIc2e8RP9rGG?usp=sharing
Thanks so much. Have a great day.
Sincerely
Are these 801s? The main problem is that they're too big for that room :) I had a similar issue with my Kantas in my previous apartment but at least my room was rectangular shaped. You have no side wall at one side from what I see in the photos.
PS You haven't swept 0-24kHz and the resonant frequency is probably below 15Hz which I cannot even see! FYI those speakers are so well protected, they cannot be damaged even if you plug them directly to AC :)
@@ocaudiophile yes. the measurements look the same with my former 803d2, which were really smaller, and the 802d2, that I had before these...
same huge dip, same peaks.
different sound to the ears of course... I can share measurements from that time if you want, and you will see the same problem
@@ocaudiophile I added an mdat from 2017 with 803D2's in the same folder!!
the room is 7.5 by 6, not that small. plus the long corridor behind... due to asymmetries, there have always been problems in that region, where the signals arrive at the mic out of phase by 180 degrees... as you will read in the notex.txt .
Kindest regards