As far as I understand microphone calibration should be very stable over normal conditions. It can change slightly due to temperature and humidity but shouldn't be by much, if it was then I would look for a different microphone for doing calibrated testing. I just realized the other day that I had been doing something wrong in my testing in that I was using a default setting in my Clio software for the microphone sensitivity, the Earthworks microphone I use is very different than the default setting. This would change general levels and even efficiency results but shouldn't affect the FR graphing. Most calibration microphones are extremely flat in the audio passband, the differences being that the higher cost microphones will have a much wider bandwidth with the highest quality B&K microphones being capable of measuring up to 100Khz. My Earthworks is fairly flat up to 30Khz. What more can you ask for?
example
I measure my system without calibration -> load measurement into generator (rePhase) -> and I want to add a calibration file for microphone into generator (rePhase) -> from there I can see a graph with (mic) calibration -> resulting a graph has microphone measured (without calibration) and calibration can be loaded to compensate the graph, and then can see what needs to be done (equalizing and phase)
I measure my system without calibration -> load measurement into generator (rePhase) -> and I want to add a calibration file for microphone into generator (rePhase) -> from there I can see a graph with (mic) calibration -> resulting a graph has microphone measured (without calibration) and calibration can be loaded to compensate the graph, and then can see what needs to be done (equalizing and phase)
This is not the way it is normally done. Usually you would load the calibration file into the measurement software, and the mic's deviation from flat is compensated for during the measurement process.
Shaun,
Yes I believe you are correct about that. I would worry that if you tried to do this calibration after the fact there could be some multiplication effects while doing the measurements and I don't know how you could simply subtract the calibration curve after the fact and get an accurate response curve. I guess it also depends on the type of microphone you are trying to use to do the measurements. Something like the Earthworks M30 mic I used is very flat in the audio band but if you are using a simple condenser microphone that can be far from true.
Yes I believe you are correct about that. I would worry that if you tried to do this calibration after the fact there could be some multiplication effects while doing the measurements and I don't know how you could simply subtract the calibration curve after the fact and get an accurate response curve. I guess it also depends on the type of microphone you are trying to use to do the measurements. Something like the Earthworks M30 mic I used is very flat in the audio band but if you are using a simple condenser microphone that can be far from true.
I found this looking for something else. Might be useful to someone if they want to add some spaciousness to the room on a couple of surrounds.
"The effects of multi-channel signal decorrelation in audio reproduction"
https://www.google.com/url?sa=t&rct...sg=AFQjCNG_4pWY7Ps1TmO5GNymbrT9IwTHiA&cad=rja
"The effects of multi-channel signal decorrelation in audio reproduction"
https://www.google.com/url?sa=t&rct...sg=AFQjCNG_4pWY7Ps1TmO5GNymbrT9IwTHiA&cad=rja
View on paragraphic eq?
Hello, I'm a newbie on the forum though not to the hobby per se. I have a question whose answer I haven't found:
First, though, rePhase is a marvelous tool. It has given me capability, along with some minidsp boards and plugins, to do things with quantum leap advantage over what came before.
One thing I would like (and cannot see how to do) would be, in the paragraphic EQ, the option to view each bank separately, sum a partial set of them, or simply sum all of them. AFAICT, with 1.0.0 I have only the last option. Even just the first (each bank separately) and the last (all summed) as options for the view window would be a tremendous help, IMO.
Is it there and I'm just missing it somehow?
Hello, I'm a newbie on the forum though not to the hobby per se. I have a question whose answer I haven't found:
First, though, rePhase is a marvelous tool. It has given me capability, along with some minidsp boards and plugins, to do things with quantum leap advantage over what came before.
One thing I would like (and cannot see how to do) would be, in the paragraphic EQ, the option to view each bank separately, sum a partial set of them, or simply sum all of them. AFAICT, with 1.0.0 I have only the last option. Even just the first (each bank separately) and the last (all summed) as options for the view window would be a tremendous help, IMO.
Is it there and I'm just missing it somehow?
Hi leoman
There is no such option.
One think I could add would be a bypass and solo button for each EQ bank. I think that would partially respond to your demand.
Another possibility would be to see the solo curve of the last changed setting (EQ, filter, etc.), in a similar manner as the current frequency "dot" that shows you were you are operating.
What is it you want to see exactly ?
There is no such option.
One think I could add would be a bypass and solo button for each EQ bank. I think that would partially respond to your demand.
Another possibility would be to see the solo curve of the last changed setting (EQ, filter, etc.), in a similar manner as the current frequency "dot" that shows you were you are operating.
What is it you want to see exactly ?
Hi pos,
I use two minisharcs, one with 2x2 and one with 4x8 plugin, so I can reserve the limited taps in the 4x8 to do decent crossovers while feeding it with 6144-tap/channel equalized output from the sharc running 2x2. So I tend to do various independent things with the 2x2 sharc using the paragraphic eq. One might be a 'room curve'. One might be cd horn compensation, one a removal of some midbass, etc. The different banks seem a perfect tool for that. But I can't be sure I'm entering them properly because they can overlap and all I see is the summed curve.
Unless I misunderstand, it sounds like your 'solo' button would permit me to see the current bank alone? That would be perfect.
I use two minisharcs, one with 2x2 and one with 4x8 plugin, so I can reserve the limited taps in the 4x8 to do decent crossovers while feeding it with 6144-tap/channel equalized output from the sharc running 2x2. So I tend to do various independent things with the 2x2 sharc using the paragraphic eq. One might be a 'room curve'. One might be cd horn compensation, one a removal of some midbass, etc. The different banks seem a perfect tool for that. But I can't be sure I'm entering them properly because they can overlap and all I see is the summed curve.
Unless I misunderstand, it sounds like your 'solo' button would permit me to see the current bank alone? That would be perfect.
Hi Pos,
First i want to commend you on your excellent skills.
I am not an expert on this subject, so pardon any ignorance.
I am considering Najda for a 4way loudspeaker. It is one of the well integrated solutions available currently but under-powered to do serious FIR, phase/Room corrections etc at 96KHz or higher.
I am planning to use an i7 HTPC as the digital music source feeding Najda through a 2ch USB to I2S board.
I plan to use Najda to do basic IIR crossovers. Can i use rePhase to build some FIR phase/room correction filters and run these on Jriver DSP on the HTPC prior to feeding the Najda filters? Does such a series chained filters approach work to obtain the best possible DSP configuration for the 4way speaker?
Does the 2ch USB I2S interface limit this in anyway?
Any pointers on how to go about setting this up would be very helpful.
First i want to commend you on your excellent skills.
I am not an expert on this subject, so pardon any ignorance.
I am considering Najda for a 4way loudspeaker. It is one of the well integrated solutions available currently but under-powered to do serious FIR, phase/Room corrections etc at 96KHz or higher.
I am planning to use an i7 HTPC as the digital music source feeding Najda through a 2ch USB to I2S board.
I plan to use Najda to do basic IIR crossovers. Can i use rePhase to build some FIR phase/room correction filters and run these on Jriver DSP on the HTPC prior to feeding the Najda filters? Does such a series chained filters approach work to obtain the best possible DSP configuration for the 4way speaker?
Does the 2ch USB I2S interface limit this in anyway?
Any pointers on how to go about setting this up would be very helpful.
Yes you can do that, I've done it in the past. There are so.e good how to quides for using convolution wav files and setting up the file over on the jriver forum. Its a bit convoluted (pardon the pun) but does work quite well.
I have followed your miniSHARC stack adventures on the miniDSP forum with great interest!
The solo button should give you what you are looking for.
I will try to include this in the next release.
jojip, yes indeed you can chain a convolution on the inputs (using a software solution in JRiver of foobar, or an harware solution like the openDRC) and a IIR or FIR crossover.
The difficult part is building the actual crossover, especially with minimum-phase targets as you have to match phase shifts between crossed-over drivers.
The difficulty is even greater with 4 way systems as you absolutely have to take into account phase interactions between crossovers:
Woofer crossover & offset
In comparison building a linear phase crossover from the get go is much easier because you only have to match linear magnitude and phase passband with proper delays and levels...
The difficult part is building the actual crossover, especially with minimum-phase targets as you have to match phase shifts between crossed-over drivers.
The difficulty is even greater with 4 way systems as you absolutely have to take into account phase interactions between crossovers:
Woofer crossover & offset
In comparison building a linear phase crossover from the get go is much easier because you only have to match linear magnitude and phase passband with proper delays and levels...
Hi Thierry,
A question just for fun/semantic ?
Why some people claims linear phase Linkwitz-Riley is not a Linkwitz-Riley definition/attribut ?
Not sure what you mean.
LR filter in the "linear-phase filters" tab in rephase follow Linkwitz-Riley magnitude response, but with linear phase throughout the pass band and stop band.
Of course these cannot be considered true Linkwitz-Riley filters as these also imply a specific phase response.
Maybe I should make this more explicit as this can indeed be misleading...
(why 9 on smiley....i do not know).I'm not sure Siegfried would have a problem with a "linear-phase" Linkwitz-Riley filter. The all-pass phase distortion of the original analog implementation (40 years ago now) was an inherent side-effect of amplitude filtering used to create the objective. I don't see how using contemporary DSP-processing to yield a constant group-delay invalidates it from being a "true" LR crossover.
Dave.
Dave.