Take it easy and do not shoot the messenger!
I often fooled myself to believe that my system seemed to sound "BETTER" when it only sounded "DIFFERENT".
My mind was influenced by some obscure rational arguments that drove me to the conclusion that the change was for positively better. But my ears were never quite sure...
Imho, the good thing with Rephase is that it does not come with an automated system for inversion and processing of the room curve, and lets the user choose what corrections he thinks necessary.
I'm not sure how much easier I could have been
But that was a much more reasonable way to say what you think.
Take it easy and do not shoot the messenger!
I often fooled myself to believe that my system seemed to sound "BETTER" when it only sounded "DIFFERENT".
My mind was influenced by some obscure rational arguments that drove me to the conclusion that the change was for positively better. But my ears were never quite sure...
Imho, the good thing with Rephase is that it does not come with an automated system for inversion and processing of the room curve, and lets the user choose what corrections he thinks necessary.
A better way to learn would have been if you spend more time on it to figure out why it did sound different.
There are many ways to get a flat looking measurements. Not all of them are going to sound equally good.
But a speaker will never be more accurate than its measurement suggests.
This isn't voodoo after all, it's pure science. The art is in the music.
But a speaker will never be more accurate than its measurement suggests.
This isn't voodoo after all, it's pure science. The art is in the music.
There is indeed a lot of art in making a loudspeaker play music in a civilized way, and as some audio guru said, a loudspeaker is also in a maneer of speaking, a musical instrument...
http://www.moultonlabs.com/more/loudspeaker_as_musical_instrument/
I prefer to leave pure science to pure scientists... I'm not.
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Plenty, anytime, anywhere... But i am an only empiricist amateur, and engineering is neither my job, nor my religion...
Plenty, anytime, anywhere... But i am an only empiricist amateur, and engineering is neither my job, nor my religion...
would you mind giving specific examples of what you measure for in the magnitude and time domains, and what ideal goals you strive for in those domains?
thanks
I must confess that in the time domain i no longer strive for (almost) anything, though my main interest in experimenting with DSP was driven by ideal goals related to "transient perfect response", "linear phase", "Quasi Optimum Filtering", "precise time alignment", or "Excess Phase Equalization"... But practically don't care anymore about impulse shapes and stuff like that...
In magnitude terms, reasonable smoothness is my humble quest: reasonable freefield smoothness, reasonable on axis smoothness, reasonable off axis smoothness, reasonable power response smoothness, reasonable room curve smoothness, overall nothingisperfect smoothness...
There is indeed a lot of art in making a loudspeaker play music in a civilized way, and as some audio guru said, a loudspeaker is also in a maneer of speaking, a musical instrument...
Moulton Laboratories :: The loudspeaker as musical instrument
I kindly disagree with that point of view. How would it be possible to see a loudspeaker that way if it's prime job is to mimic every instrument ever recorded. The last thing I'd want is for the speaker to add it's own signature (which usually is how instruments work) to every song played on it. It should be neutral, not add it's own timbre.
Same goes for amplifiers etc. Though I'm not that strict as I do like some tonal balance tweaks myself at times, but that need not be part of the job of the loudspeaker (nor the room). Easily done by DSP if one likes that sort of tweak. Just a touch of warmth goes a long way for me. Personal bias of which I'm perfectly aware. Years of listening to "warm sounding" speakers probably had something to do with that. Leaving it up to DSP tweaks means you can actually turn it off too.
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I would also say so, but... all loudspeakers have their own signature, timbre, etc...
Btw you can Eq peaks or deeps, above or under a given smooth/flat target response, but... how can you get rid of resonances buried 10 or 20 dbs below average level?
What is timbre and neutrality all about?
By looking at your measurements? Taking care of reflections will lower the resonances you speak of. If the speaker doesn't *** to much of it's own of course.
Here's the midrange of my speakers in an early waterfall plot:
Each horizontal line represents 2 dB so I'm actually showing a -25 dB scale... as usual measured at the listening position. I love a clean clear midrange.
I'd avoid trying to EQ dips. Unless you know it's not a null. Avoiding them would be better. Clean up that response. Don't hammer it into shape using DSP.
What is timbre and neutrality all about?
That's what we see in that IR, well we don't get to see all of it until we start dissecting that IR into all the pieces hidden in there.
Neutral should mean adding the least of it's own to the source. If I think of musical instruments I think of a guitar body etc. The body of my speakers should actually be as silent as possible.
The above waterfall plot was an older one, here's a more recent plot taken 2 years later:
Again looking at a 25 dB drop at the listening position. Could we see yours?
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By looking at your measurements? Taking care of reflections will lower the resonances you speak of. If the speaker doesn't *** to much of it's own of course.
Here's the midrange of my speakers in an early waterfall plot:
Each horizontal line represents 2 dB so I'm actually showing a -25 dB scale... as usual measured at the listening position. I love a clean clear midrange.
I'd avoid trying to EQ dips. Unless you know it's not a null. Avoiding them would be better. Clean up that response. Don't hammer it into shape using DSP.
That's what we see in that IR, well we don't get to see all of it until we start dissecting that IR into all the pieces hidden in there.
Neutral should mean adding the least of it's own to the source. If I think of musical instruments I think of a guitar body etc. The body of my speakers should actually be as silent as possible.
The above waterfall plot was an older one, here's a more recent plot taken 2 years later:
Again looking at a 25 dB drop at the listening position. Could we see yours?
Wesayso, it's hopeless talking with you. It' just too obvious that your only interest is jump to the nearest available solution, and mine to the closest possible problem around...
Btw, congrats for your outstanding wonderful measuring and sounding system...
Me saying they sound good would say nothing .... I build them and I am biased, no doubt. Over time I've had a few guests that "reviewed" my system, 3 of them members on here. I value their opinion as they didn't have any stake in it. I'm glad they were willing to coma and visit and write something about it.
The funny thing is: I actually can relate what I hear back to the measurements. But it didn't start out that way at all. We need to learn where to look and figure out how that grey mass between our heads process it all. Two ears do way different than one microphone. Two ears and a brain? Oh no!
DSP alone could have never gotten me this far. This is after all at the listening spot, not an up and close measurement.
Lets return to talk about RePhase, the actual topic of this thread. I am a thankful user even though I do also use other products it has helped me in my journey. A satisfied user so to speak. I use it a lot for simulations as well. Whenever I get face to face with POS I sure will buy him a couple of beers .
.... I build them and I am biased, no doubt. Over time I've had a few guests that "reviewed" my system, 3 of them members on here. I value their opinion as they didn't have any stake in it. I'm glad they were willing to coma and visit and write something about it.The funny thing is: I actually can relate what I hear back to the measurements. But it didn't start out that way at all. We need to learn where to look and figure out how that grey mass between our heads process it all. Two ears do way different than one microphone. Two ears and a brain? Oh no!
DSP alone could have never gotten me this far. This is after all at the listening spot, not an up and close measurement.
Lets return to talk about RePhase, the actual topic of this thread
. I am a thankful user even though I do also use other products it has helped me in my journey. A satisfied user so to speak. I use it a lot for simulations as well. Whenever I get face to face with POS I sure will buy him a couple of beers .
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Personally I get exited to see more and more people showing an interest. The collaboration of John (REW) and POS (RePhase) responding to us as users... makes me proud to be a member of this lively community.
No doubt we all have different goals, but there should be room enough for all of us.
No doubt we all have different goals, but there should be room enough for all of us.
JohnPM said:
I was thinking interpolation (cosinus or cubic) or a low-pass filter.
upsampling the time domain (impulse) with 0 followed by a low-pass brickwall (sinc fct) at 48k should give double resolution and new upper limit.
the zeroes will be "smoothed".
the display in the frequency domain still stopped at 24KHz.
But that is exactly the problem with upsampling. Cepstrum uses log of magnitude of frequency response, so frequency response cannot be zero anywhere.
A bit of topic, and only for information for those interested in using FIR/Rephase with windows 10 AND has latency requirements:
https://msdn.microsoft.com/en-us/wi...low-latency-audio?f=255&MSPPError=-2147217396
https://msdn.microsoft.com/en-us/wi...low-latency-audio?f=255&MSPPError=-2147217396
Hello Carl
linear-phase EQ are a no go for sound reproduction.
Minimum-phase EQ (either IIR or FIR, or even analog for that matter) are what you are looking for, as flattening magnitude (I suppose that is what you are looking for when applying EQ) will also flatten phase in the same time.
The only thing you want to correct using non minimum-phase corrections are all-pass filters (more or less textbook ones...) caused by crossovers, and that kind of correction does not require tweaking from one place to another as it is related to the source itself.
So if you are looking for a linear phase system, what you should do is first measure your system in anechoic conditions (that can be done in a normal room, using gating and several measurements at different distances and angles...), EQ that to flat using minimum-phase EQ, and then apply gentle phase corrections (filter linearization tab as well as a tad of phase EQ if you need to) to flatten phase.
Once you have all that in a FIR you will only need minimum-phase EQs to tailor your system to a given venue, and that is perfectly done using IIR EQ devices, and in real time if you so wish
pos,
I am stilling learning to use rePhase. Just want to confirm about the following steps after doing several measurements ( I use REW). I assume I need to click the 'Average The Responses' button under the 'All SPL' section. While the response-averaged measurement doesn't have phase information anymore, is it ok for me to import the response-averaged measurement to rePhase? Or do I miss any important step(s)
Thank you for creating such a great tool!
ppmmcc
Hello ppmmcc
Thank you for your kind words.
Averaged measurements are one tool you can use, among others (or combined with others).
If phase info is missing then you have to concentrate on minimum-phase magnitude EQs only, and then possibly use another (close range and/or gated) measurement for phase linearization.
Thank you for your kind words.
Averaged measurements are one tool you can use, among others (or combined with others).
If phase info is missing then you have to concentrate on minimum-phase magnitude EQs only, and then possibly use another (close range and/or gated) measurement for phase linearization.