Room EQ Wizard (free measurement and parametric EQ setup software) - Page 14 - AVS Forum
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post #391 of 855 Old 01-16-2006, 01:16 PM
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Thank You for the program.

I had a difficult time getting sound from my soundcard. I eventually gave up and ran the spdif out from the soundcard to my receiver and everything worked great.

I input the programs filters manually and added Sonnie Parkers recommendations for additional filters at 1.0, 1.25, 1.63, and 2.0 khz.

I then ran another sweep and the program said that there are no peaks.

I have not checked the work with another source or a manual check but I am satisfied.

I have had the BFD sitting around, unused for months. I feel better now.

All animals are created equal, but some animals are more equal than others.
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post #392 of 855 Old 01-16-2006, 01:27 PM - Thread Starter
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Randy, you're welcome

John

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post #393 of 855 Old 01-16-2006, 03:39 PM - Thread Starter
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V3.28 is now available for download, changes are detailed in the Change History

This version adds a "House Curve" option to load a desired target shape for the filters to be optimised against e.g. to allow a small rise in the response at LF for a subjectively better response. It is also now possible to change the channel names (Left, Right, etc) by right clicking on the channel select tabs in the Filters pane. The new names are remembered for the next startup, as are the speaker type and bass management cutoff settings. The settings last used for the Set Extents dialog are also remembered. Fractional octave smoothing has been made much faster, 1/2 octave has been added as a smoothing step and a keyboard shortcut, ctrl+3, has been added to provide a quick way to apply 1/3rd octave smoothing to the current channel. Filter BW is now shown in Hz alongside the Q/BW figure. Various other minor changes have been made, and the "would not close" bug and broken stepped sine measurement have been fixed.

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post #394 of 855 Old 01-16-2006, 04:27 PM
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John,

I am learning more about the convolver approach. Can I use your software to generate a filter file and use that for Foobar2000's convolver? Are there instructions in the help docs for generating such a 2 channel filter?

Thank you
Darren
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post #395 of 855 Old 01-17-2006, 01:17 AM - Thread Starter
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Hi Darren,

I don't know what the foobar2000 convolver plug-in requires. The File -> Export -> Channel Filter Impulse Response as WAV option generates the impulse response of the filters for the current channel in WAV format, written as 16-bit mono PCM signed data, with the impulse peak normalised to digital full scale. If something different is required it could be generated with a WAV editor of some sort in the short term, longer term I could add an option to the app if someone can tell me what is needed.

John

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post #396 of 855 Old 01-17-2006, 05:33 AM
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My convolver reads and uses mono wavs fine (although it would be even better if they were floats).

It will also take stereo wavs and apply them channel by channel to stereo input (which is mainly useful for reverb effects, but can also give a good room correction sound, despite the theoretical incorrectness). I imagine that this is what foobar2000 does, but have not checked.

John
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post #397 of 855 Old 01-17-2006, 07:16 AM - Thread Starter
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Quote:
Originally Posted by jrpavel View Post

My convolver reads and uses mono wavs fine (although it would be even better if they were floats).

Floats would be a little awkward, but I'll add a selector to choose 16, 24 or 32-bit signed PCM.

John

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post #398 of 855 Old 01-17-2006, 08:46 AM
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Thanks John!

REW now has 2 features (basically ) I asked for, and 2 more I was going to ask for. Keep up the good work!

CTRL+3 really speeds up taking multiple measurements for mains placement, and the house curve is something I wanted to ask for, and it is great to have!
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post #399 of 855 Old 01-17-2006, 08:54 AM
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jrpavel,

Does this mean all we need for basic room correction in Windows is REW and your convolver plug-in? How do we get REW's mono file to convolve with a stereo source? Do we have to use REW to capture each channels impulse response separately, then combine them into a stereo file in another program (i.e. Audacity)?
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post #400 of 855 Old 01-17-2006, 11:21 AM
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Yes.

Others with more experience of room correction may want to chip in, but there are several approaches of increasing complexity.

Assume that you are using stereo, then:

You can measure the impulse response of your left and right speakers separately, and apply the results to the respective channels. The first example on http://convolver.sourceforge.net/configegs.html shows the convolver config file that allows you to do that. Alternatively, you can combine the mono impulse responses into a single stereo WAV file and load that directly instead of using a config file.

That is not, however, the theoretically best way of correcting your system, although it may give good results in practice and has the merit of simplicity. (At least not if you don't listen with a board in front of you that prevents the left channel sound from reaching your right ear ...)

The more correct way, apparently, is to take 4 impulse response measurements (left speaker to left ear, left to right, right to right and right to left) and then apply them using the head-related transfer function (HRTF) network example on http://convolver.sourceforge.net/configegs.html as a model.

I'll leave it to others to comment on / correct this advice, but the long and the short of it is that with convolver you can mix input channels, filter the result, mix/sum the results of different filter paths, and direct them to specified output channels in a completely configurable way. (In fact, I am trying to decide how best to reduce the flexibility, as it makes convolver more complex to set up than is probably strictly necessary.) So, if you want to do other things like split a channel into different frequency ranges and send the results to different output channels, you can do it.
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post #401 of 855 Old 01-17-2006, 11:43 AM
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John,

It is refreshing during these times, that you are able to devote as much of your time as you do to the programming and support of this software. Even to just offer that you would consider possibly adding Foobar2000 convolver support to this software is greatly appreciated.
I personally have listened to all of the software players out there. Foobar2000 just seems to produce the best quality for my system.

You prob already have read this thread regarding DRC:

http://www.avsforum.com/avs-vb/showt...&highlight=drc

There is a guide for using this DRC for creating convolver files. It is fairly complicated and involves several programs. If your software could accomplish this, it could be as simple as taking readings saving a file, import into Foobar and done!

Example Foobar2000 files
http://www.foobar2000.org/impulses.zip

convolver plugin
http://www.foobar2000.org/foo_convolve.zip

Thank you
Darren
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post #402 of 855 Old 01-17-2006, 01:07 PM
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Darren, thanks.

I had not thought of adding plugging convolver into fb2k as I think that there already is one for it.

I've had at the SDK and there is little documentation, so it would be a matter of trying to do it by looking at the sources. It's probably not v difficult but all takes time. A VST interface is also in demand, which is reasonably well documented so that is probably higher on the list.


NB. Convolver will play the impluses, but not generate them
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post #403 of 855 Old 01-17-2006, 05:43 PM
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John and jrpavel,

One thing I am still not clear on - Do we export the channel impulse response as wav, or the channel filters impulse response as wav for use with convolver?

John,

Since it seems using a stereo wav for convolution is easier than 2 mono files, might it be possible to have an option to save 2 channels as a stereo wav file?

jrpavel,

Thanks for the explanation. If I have any more questions specifically about your plug-in, I will head over to your convolver thread.

Darren,

My thoughts exactly! the DRC site makes the whole process sound complicated and using REW to make the nessesary file for convolving would be much less involved.
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post #404 of 855 Old 01-17-2006, 06:22 PM
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jrpavel there is already a convolver plugin for foobar 2k. Although I do not i know yet how well it works.

I am going to try a few experiments this weekend with DRC and/or REQW. I will take to mono readings from REQW and combine them in Cool Edit. We will see what happens!!

trymor,

What do you think? (2 spkrs and sub). Scan left channel with sub. Scan right channel with sub. Combine the two together in cool edit? Something like that. Computers I know. I am a computer tech. However, I currently know nothing about convolving (is that a word?).


Darren
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post #405 of 855 Old 01-18-2006, 01:18 AM - Thread Starter
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Darren, Trymor,

It is the channel filters impulse response you need to export for use with the convolver. I'll look at allowing a pair of responses to be exported as a stereo WAV, for the moment Cool Edit or Audacity or similar should do the trick.

John

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post #406 of 855 Old 01-18-2006, 02:52 AM
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@Randy Mathis:
Quote:


I input the programs filters manually and added Sonnie Parkers recommendations for additional filters at 1.0, 1.25, 1.63, and 2.0 khz.

Since I could not find these recommendations in this thread, what are they for ? Also for subwoofer calibration only ?

Thanks, Dieter
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post #407 of 855 Old 01-18-2006, 04:42 AM
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Darren,

Short answer, yes, I think so, but it may depend on how your receiver handles the sub. Is it connected to the sub out of the receiver, or are you running speaker wires to the sub, then to the mains utilizing the subs high-pass crossover and turning off the sub output of the receiver.

Don't know if that really matters, but I now understand jrpavel's statement about the method not being 'politically' correct. When you take separate left and right measurements, there are nulls and peaks that are NOT there when you take a combined measurement, because they get cancelled out. So, if the left and right channels get corrected separately, the combined output will be different than predicted (i.e. different nulls and peaks than when measured with no correction). So does this mean a better result can be obtained using one system measurement (mono signal sent to left, right and sub at the same time) and using the same mono wav for both left and right channels? I don't know, but I think I am going to try it. In my mind, the corrected system response I get in REW should then be what I hear. The thing is, in real music, you have some 'mono' sounds (most low bass and centered singers) and 'stereo' sounds (guitars on the left, ambiance and soundstaging cues for example) so it seems both simple methods have pluses and minuses. That must be why the correct way involves summing and differential correction.

If we are to continue this convolver conversation, perhaps we should take it to an appropriate thread?

GinSonic,

I was wondering the same thing. Supposedly it is not a good idea to use the BFD for anything other than a sub because of the sampling errors and phase shift it creates, making the sound worse.
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post #408 of 855 Old 01-18-2006, 05:27 PM
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Please forgive me if this is already covered (I searched the thread and didn't come up with anything) but how does a program like this and the BFD relate to something like the auto EQ capabilities of Receivers like the Pioneer Elite 74? Same thing, or can I still do more/a better job with this combo?

TIA

ebr

 


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post #409 of 855 Old 01-18-2006, 08:03 PM
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Trymor,

About moving the conversation... It's more actually related to REW and weather John thinks what the best approach for this calibration for his software would be. Also whether he wants to possibly add the ability to create file suitable for convolver from within his software.
Really, IMHO, that jrpavel is prob correct (not exactly sure there is a correct way due to the "fluid" nature of stereo sound) in using averaging. I just don't quite know enough about acoustics and audio streams to make the conversation interesting. I am going to try something with this info, this weekend using REW. We'll see how it goes. I am going to compare the readings taken with R then L then R+L and see how much different they are.

Darren
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post #410 of 855 Old 01-18-2006, 08:23 PM
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JohnPM or others,

What is a quick and dirty way to figure out (using test signals and the REW) if the line output from my Radio Shack digital SPL meter is c-weighted compensated or unweighted (as apparently some are according to other posts on this forum)?
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post #411 of 855 Old 01-19-2006, 01:48 AM - Thread Starter
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Quote:
Originally Posted by ebr View Post

Please forgive me if this is already covered (I searched the thread and didn't come up with anything) but how does a program like this and the BFD relate to something like the auto EQ capabilities of Receivers like the Pioneer Elite 74? Same thing, or can I still do more/a better job with this combo?

Short answer is you should be able to do a much better job at the low end with a BFD and appropriate measurement software, but the process is a lot more involved than pressing a button and sitting back. Auto EQ features would typically provide correction over most of the frequency range which could be done with a BFD or other equaliser but would need a better microphone than an RS meter for frequencies above about 2kHz.

John

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post #412 of 855 Old 01-19-2006, 01:50 AM - Thread Starter
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Quote:
Originally Posted by Darren Wadsworth View Post

About moving the conversation... It's more actually related to REW and weather John thinks what the best approach for this calibration for his software would be. Also whether he wants to possibly add the ability to create file suitable for convolver from within his software.

I've added the ability to generate stereo WAV files with two filter impulse responses in the current dev build so that will be in the next release.

John

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post #413 of 855 Old 01-19-2006, 01:52 AM - Thread Starter
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Quote:
Originally Posted by WarnerL View Post

What is a quick and dirty way to figure out (using test signals and the REW) if the line output from my Radio Shack digital SPL meter is c-weighted compensated or unweighted (as apparently some are according to other posts on this forum)?

Without some other reference measurement you can't. It is more likely that the output does need C weighting compensation than not.

John

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post #414 of 855 Old 01-19-2006, 05:06 AM
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John,

Thanks! I may not get to doing drc till the next release anyway, as I have used the software PEQ plug-in for winamp on the bass region and it sounds great now!

Darren,

I apologize. I know the conversation was about convolution and REW, but I thought it started moving toward drc theory and setup, and I didn't want to plug up Johns thread (which I guess I did with my last somewhat lengthy post).
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post #415 of 855 Old 01-19-2006, 05:37 AM
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Thanks, John. I've done my own room corrections using TrueRTA and a Bijoux analog EQ so I understand how involved that process is. Looks like your program helps it out quite a bit. I just wanted to know what everyone thought of the capabilities of the auto room corrections built in to the latest equipment. I heard that the new Pio Elite receivers had added some low end corrections - but I'm not sure how good they are.

I plan to try and tackle most of the room issues with treatments, but there's always some EQ that needs to be done.

Thanks again for your generous work on this program.

ebr

 


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post #416 of 855 Old 01-19-2006, 10:04 AM
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""I apologize. I know the conversation was about convolution and REW, but I thought it started moving toward drc theory and setup, and I didn't want to plug up Johns thread (which I guess I did with my last somewhat lengthy post).""

No need to apologize. You are prob right about plugging up the thread. Prob going off topic. I enjoyed your lengthy post. It gave me some valuable info to think about. I need to finsh reading the DRC thread. Good information over there.

John,

Thank you for deciding to add that in. As soon as that release is ready, I will be testing that out.

Darren
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post #417 of 855 Old 01-20-2006, 01:00 PM
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I used REQW last night to create convolver files for Foobar2000. It seemed to work great! I did have one dumb question though. I know that I am supposed to export channel filters impulse response as wav. My dumb question is, is this done after I adjust the filter settings within REQW?
Also, for John, why does the program only auto adjust filters at and below 500hz? Is this because of the limitations of the RS SPL? I am using a full range mic. So any major peaks after 500hz I have to look for and adjust manually. This is very time consuming. Is there anyway that you could add the ability to adjust filters for up to 20k hz?

I tried using the DRC method also. I am a computer tech so the proceedures were not that hard for me. However, the filters it created were not correct. The music was garbage.

Thank you
Darren
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post #418 of 855 Old 01-20-2006, 04:53 PM
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Wow, thats great that REW does drc files better than DRC. Yes, export the filters as wav after all your adjustments are done, or you wont be correcting anything.

I was wondering the same thing about the 500Hz cutoff, but I'm not sure if the precison can be good enough considering the automatic filters are far from giving me flat response. I have to add a bunch of filters, and tweak the supposedly optimized ones, but at least it gives me a starting point. So far I have only used the generic EQ setting, and entered the filters manually into the Winamp plug-in.
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post #419 of 855 Old 01-21-2006, 04:22 AM - Thread Starter
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The peak search function is restricted to lower frequencies as it is looking for modal resonances. Above a few hundred Hz the resonances overlap to such an extent that it si no longer useful to try and correct individual resonances, but you can of course apply broader filters to even out the response. Future versions of the Wizard will address other parts of the range, but that is quite a way off as there is much yet to do for the low frequency range. It is worth bearing in mind that room treatments work very well in these ranges.

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post #420 of 855 Old 01-22-2006, 02:21 PM
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John,

Thank you. Very concise.

Is there anyway that a filtered sweep could be played and measured through Room EQ Wizard. I use a PC for all of my music playback. If not, would it be at all possible for you to add the capability of playing a sweep with the filters inplace, and record the results?
Currently the only way I know of to accomplish a "comparison" sweep of before and after equalization is through the use of a hardware EQ like the BFD.

Thank you
Darren
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