You should verify your settings, the impulses looks flabbyFiltered wave shapes
You should verify your settings, the impulses looks flabbyFiltered wave shapes
Found the tutorial but in French is that right is there one in English?
I could and was about to and then I saw thisNot bad, could you filter it at 60Hz or 100Hz ?
You should verify your settings, the impulses looks flabby![]()
Yes, if you read the linked paper from JJ on acoustics and psychoacoustics, it is the first arrival timbre that our ears care about most.
I did my averaging left to right rather than in a line towards the speaker as my speaker changes response more with distance than other's do and also I was aiming to get a balanced correction across my listening area.Hi fluid, cool. Yes, I did similar using the beamforming technique and moved the mic in 10 cm increments moving forward and back from the LP for 10 measurements in total. Does do a great job of suppressing the room and have similar ETC's like yours. Still considering if I like the sound versus a single measure. Some say that the multiple measurements also helps average the low frequency response and others say that for some DSP correction software, the averaging is built-in to the psychoacoustic filter...
Nothing wrong with that but it does require a reasonable amount of room treatment which not everybody is willing or able to use.I I'm a fond believer in the EBU (European brodcaster union) spec. Early reflections inside 15 ms matters and should be under -15dB in ETC curves.
For myself I am not in any way trying to promote line arrays or run down multi way systems. Both have positives and negatives that are fairly well documented. You are the one that has been asking questions about line arrays and phrasing it in a way that sounds negative. It is hard not to respond to those kinds of posts. I have both a line array and a couple of good examples of a multiway loudspeaker so in that sense I am in a good position to compare. I like both for different reasons.Not to michba:
I will try staying at topic but it is not easy with all the blurbs here lately.
Why insulting other world class system owners beacause you have a world class system yourself?
.....I will try staying at topic but it is not easy with all the blurbs here lately.
Why insulting other world class system owners beacause you have a world class system yourself?
.....Any other volunteers showing their IR results at the listening position?.....
...Is it blurbs share prescription and results on domain meant to help each other using Rephase either as primary or secondary helping tool and how can that at all insult other owners.
I I'm a fond believer in the EBU (European brodcaster union) spec. Early reflections inside 15 ms matters and should be under -15dB in ETC curves.
Not to michba:
I will try staying at topic but it is not easy with all the blurbs here lately.
Why insulting other world class system owners beacause you have a world class system yourself?
I did my averaging left to right rather than in a line towards the speaker as my speaker changes response more with distance than other's do and also I was aiming to get a balanced correction across my listening area.
I think with the averaging it depends on the strength of the filter that is created as to whether it sounds different or not vs single position. My average frequency response looks very similar to the single central position but the ETC is quite different. My template has longer windows at top and bottom which are too much for a base measurement that has a lot of room sound in it.
It would interesting if you could import your correction filter impulse into REW to see how "strong" it is as that might give a clue as to why you found less difference.
I also don't use the psychoacoustic target stage in DRC which could also be part of it.
Because the low frequency response is dominated by room modal behaviour the average was most similar in that range, the REW room simulation was a very good predictor of the best positions for the speakers from a bass perspective.
None of that has anything to do with rephase but the processing and measurement of my speakers does which is why I have posted it here.
Well showing off perfect curves and state or indicate that they are needed to look good to sound good is insulting, at least to me that don't hear phase distortion inside reasonable limits.
So phase distortion will make phase and time curves look bad but it can sound good.
And non averaged off axis measurements are not shown. I believe they matters. That is why we use LR filters. And it is why sharp filters can be dangerous.
And that IS relevant to Rephase use and end result analysis
Measuring closer to the speaker and/or using measurement gating will help you get a more usable phase reading.Hi,
Everyone I am new to using the rePhase software for the first time. I will outline what I have and generated and many questions to ask but limit to asking what I need to know first.
Speakers: 2 way open baffle(similar to LX Mini, cross at 750Hz)
Equipment: MinidspHD 2*4 way
Software:REW via Umik and Rephase
I generated a x-over at 750Hz Using LR 2nd order, then I used the REW to equalise the output to relatively flat. I downloaded to Rephase software and was I unable(due lack of knowledge of rePhase) to make the phase flat. Attached is my freq response and phase outputs, I would appreciate for advice on how to make the phase as flat as possible.
Questions:
1. the Rephase sequence/steps I should be considering to equalise the phase for any output response?
2. What curve smoothing should I use normally?
Yes, sorry about that.Rephase is a powerful software and maybe due to lack of documentation(text and video) I cannot really get a full understanding on how to to use it effectively.
Dropbox - REW_rePhase_tuto.pdf
This is in English, also Mitch linked to another walkthrough on the Computer Audiophile site a few posts back.
I think you have completely misunderstood the point and intention of showing some of the graphs and perhaps because of that not really read the text that goes with them. There is no need to have a perfect looking impulse to get good sound. There are plenty of good sounding speakers that have awful impulse and step responses it is by no means critical to good sound.Well showing off perfect curves and state or indicate that they are needed to look good to sound good is insulting, at least to me that don't hear phase distortion inside reasonable limits.
So phase distortion will make phase and time curves look bad but it can sound good.
I don't have any to share unless I go to the trouble of convolving the same filter with each of my individual measurements. When I move across my couch from left to right I don't hear much difference in sound so for myself I am happy that it works well enough in my application.And non averaged off axis measurements are not shown. I believe they matters.
It is a little hard to see when measurements are of DRC prosessed or non prosessed arrays. Or is some of the measurements just rephase prosessed?
It is very easy to be critical for the sake of it, it is much harder to give positive criticism that is actually helpful.Sorry for sounding negative. It's meant to sound critical.
(I have learnt most of the critical voices in here, enoughas it is.)
Not to michba:
I will try staying at topic but it is not easy with all the blurbs here lately.
Why insulting other world class system owners beacause you have a world class system yourself?
Hi everyone,
Another method for separating the characteristics of the signal produced by the loudspeakers from the noise added by the room is to use some kind of averaging technique.
Here is an academic view on the subject: Enhancement of loudspeaker impulse response measurement using beamforming methods
I am personally using a different technique, which consists in making 8 measurements at the edges of a parallelepiped (stretched cube), which is containing the listening area. I sometimes add a ninth point which is the center of this parallelepiped and is the sweet-spot. In my case, the parallelipiped is 0.8m wide, 0.4m deep and 0.2m high.
Using either the (A+B)/2 impulse algebra function of REW in case of 8 measures, or simply the A+B function, I add all the impulse responses, after having carefully centered each of the impulses in the time domain to make sure not to have phase errors when adding them.
This way, the noise of the room is considerably decreased and the remaining is the signal averaged at the listening position.
Here are some illustrations. In orange color, the measurement at the sweet-spot in 1/48th octave smoothing, without any Frequency Dependent Window. In blue color, with the exact same parameters, the sum of the 9 measurements, corrected for the amplitude.
You can observe that the phase signal for instance is much cleaner with this technique... On the ETC curve, you will notice that the early reflections of the sweet-spot measurement have been wiped out by the summation/averaging process. This is the reason why the signal is much cleaner.
This is the way I intend to try to do it, as well.
Would adding frequency-dependant-windowing on top of this be of any use, you think?