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Author Topic: Native JRiver 64bit fp convolution engine for Room Correction (FIR filters)?  (Read 340467 times)

v_erich

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Re: Native JRiver 64bit fp convolution engine for Room Correction (FIR filters)?
« Reply #300 on: February 15, 2012, 12:23:16 pm »

Hallo Uli,

and where is the channel mapping to my fireface?
I have a 2.1 system full active (2x TMT+HT, 1x Sub).

My actual config is:
Code: [Select]
192000 2 6 0
0 0
0 0 0 0 0 0
h:\acourate_Daten\aktiv\gut\Cor1S192.wav
0
0.0
4.0
h:\acourate_Daten\aktiv\gut\Cor1S192.wav
1
1.0
5.0
h:\acourate_Daten\aktiv\gut\Cor2S192.wav
0
0.0
2.0
h:\acourate_Daten\aktiv\gut\Cor2S192.wav
1
1.0
3.0
h:\acourate_Daten\aktiv\gut\Cor3S192.wav
0
0.0
0.0
h:\acourate_Daten\aktiv\gut\Cor3S192.wav
1
1.0
1.0

Thanks,
Erich
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v_erich

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Re: Native JRiver 64bit fp convolution engine for Room Correction (FIR filters)?
« Reply #301 on: February 15, 2012, 12:25:17 pm »

@Matt:
http://www.acourate.com/XOWhitePaper.pdf
Hope this helps ;-)

Thanks,
Erich
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AudioVero

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Re: Native JRiver 64bit fp convolution engine for Room Correction (FIR filters)?
« Reply #302 on: February 15, 2012, 12:46:30 pm »

Hallo Uli,

and where is the channel mapping to my fireface?
I have a 2.1 system full active (2x TMT+HT, 1x Sub).

My actual config is:
Code: [Select]
192000 2 6 0
0 0
0 0 0 0 0 0
h:\acourate_Daten\aktiv\gut\Cor1S192.wav
0
0.0
4.0
h:\acourate_Daten\aktiv\gut\Cor1S192.wav
1
1.0
5.0
h:\acourate_Daten\aktiv\gut\Cor2S192.wav
0
0.0
2.0
h:\acourate_Daten\aktiv\gut\Cor2S192.wav
1
1.0
3.0
h:\acourate_Daten\aktiv\gut\Cor3S192.wav
0
0.0
0.0
h:\acourate_Daten\aktiv\gut\Cor3S192.wav
1
1.0
1.0

You are routing the sub to channels 4/5, the TMT to channels 2/3 and the tweeters to channes 0/1, right?
As you have only one sub you may route both stereo channels to output channel 4 (if the sub is connected physically at this output)

@ Matt: BTW I wonder what happens if the sum of left and right input channels exceed 0 dB. Matt, can you tell us?

Anyway you use the putput channels in consecutive order. If you use the ASIO interface then you can select the starting channel in the MC ASIO settings by the parameter Channel offset (select even numbers).
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Matt

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Re: Native JRiver 64bit fp convolution engine for Room Correction (FIR filters)?
« Reply #303 on: February 15, 2012, 12:51:38 pm »

@ Matt: BTW I wonder what happens if the sum of left and right input channels exceed 0 dB. Matt, can you tell us?

The normalize volume option in the convolution engine targets -6 dB.  This is done to provide headroom and lower the chance of clipping.

However, it's still mathematically possible (but hopefully rare with real world signals) for clipping to occur (exceeding 0 dB in AudioVero's terms).

When this happens, clip protection will engage.  This will turn down the level gradually until no clipping occurs.  It is possible to instead flat-line overflows by selecting this option in the lower left of DSP Studio, but the default 'Clip Protection' method is recommended.

If you use Internal Volume and often listen below 100%, it will provide more headroom which makes clipping less common:
http://wiki.jriver.com/index.php/Volume
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v_erich

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Re: Native JRiver 64bit fp convolution engine for Room Correction (FIR filters)?
« Reply #304 on: February 15, 2012, 03:30:09 pm »

@Uli:
Yes, I "connect" the 2 Woofer channels in the Totalmix software of rme together.
This has historical reasons, because before I use acourate I used a lot of VST plugins to have a full active setup with eq and runtime corrections.
There I connected the channels in the mixer software together, therefore it's stil the same.

So I will try to change the order and use it in MC17.
How is the output channel setting then? 5.1?

Thanks,
Erich
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BradC

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Re: Native JRiver 64bit fp convolution engine for Room Correction (FIR filters)?
« Reply #305 on: February 15, 2012, 06:37:18 pm »

The issue is that the Microsoft DVD Navigator (the thing that reads DVDs on Windows) will not provide the audio more than a little ahead of the video.  This doesn't work well if there's a large audio latency.  People that set the primary buffer size in Options > Audio to a large size run into this same problem.

One solution would be to do DVD title play, which plays the raw MPEG of the main title.  This is what we do when streaming a DVD to a DLNA box.  This solution could work locally as well, but would disable trailers, menus, etc.

Another solution would be to find or write another DVD navigator.  However, it doesn't seem like anyone has made much progress on this, possibly because of the DRM that can be baked into DVD.

Is there a setting in MC to enable DVD title play only?
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Matt

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Re: Native JRiver 64bit fp convolution engine for Room Correction (FIR filters)?
« Reply #306 on: February 15, 2012, 07:24:46 pm »

Is there a setting in MC to enable DVD title play only?

Not for local playback.  It's just something we've discussed adding, so I don't have any guess as to when or if we might do it.

Please feel free to start a thread on the topic and gauge the general interest.  When lots of people want things, they're more likely to happen.

Thanks.
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Matt Ashland, JRiver Media Center

hulkss

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Re: Native JRiver 64bit fp convolution engine for Room Correction (FIR filters)?
« Reply #307 on: February 15, 2012, 08:36:52 pm »

DVD playback: Is there a solution to get dvds to play without large stutter when the convoled filters have a large delay?

I am doing exactly that (playing optical blu-ray and dvd with large convolver delay) with no problems. I just use the advanced video settings to set a delay for lip sync, usually -300 ms or so.
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hulkss

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Re: Native JRiver 64bit fp convolution engine for Room Correction (FIR filters)?
« Reply #308 on: February 15, 2012, 08:43:13 pm »

The normalize volume option in the convolution engine targets -6 dB.

I believe Matt is sending pink noise to do the normalizing. If the various input channels are correlated, two channels combined will sum to +6db and the normalization will set the gain down -12 dB to compensate and achieve -6dB. Is that how it works?
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hulkss

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Re: Native JRiver 64bit fp convolution engine for Room Correction (FIR filters)?
« Reply #309 on: February 15, 2012, 08:47:34 pm »

Hi, when is it possible to use active filters in MC17?

I am using active filters now with the JRiver convolver (I call it JRevolver). I have three, three-way loudspeakers with active XO and three subwoofers with active XO. Sixteen output channels in all.
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v_erich

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Re: Native JRiver 64bit fp convolution engine for Room Correction (FIR filters)?
« Reply #310 on: February 17, 2012, 03:01:46 pm »

I have done it and changed to full active convolving within MC17 directly.
Sound is like before, maybe a bit more relaxed, but better handling of buffers so no clicks from time to time like before with external VSTHost.

Thanks for this great piece of software.

Erich
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kotani

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Thank you so much for this feature!

Unfortunately, I can clearly hear the difference when using 192khz filters as opposed to 44.1khz filters when I am listening to 44.1khz content. I highly prefer the 44.1khz filters in that case.

I am currently manually switching between the different sample rates depending on my music content, which is a bit tedious.
Thank you for all the hard work, but I would also greatly appreciate automatic sample rate filter switching based on the generated .cfg file.

Thank you very much...
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Gerbrand

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First of all: thanks a lot for this new feature. I am now experiencing the clearest bass I have ever heard in my system!

Nevertheless, I still have some issues. Allthough the convolution engines states it is working at 4x real time, I still experience the occasional stutters and/or slowing down of the audio. I am playing most of my content off a NAS or physical disk and this happens in both cases.

Is there anything that can be done to remedy this?

Thanks,

Gerbrand
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Matt

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I still experience the occasional stutters and/or slowing down of the audio.

Try increasing buffering in Options > Audio > Output mode settings...
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Matt Ashland, JRiver Media Center

frogger

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Hello

Thanks so much for your hard work to do this great piece of software. I`am using JR Convolver with 2 channel
convolving for my main speaker.  2 subwoowers are in work...

The only think that missing is the support of different filters for different samplerates.
Is there a chance for this in the future ? 

Thanks...
 
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Matt

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The only think that missing is the support of different filters for different samplerates.
Is there a chance for this in the future ?

Welcome.

I can't say when it will happen, but supporting different filters for different sample rates is on our list.
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Matt Ashland, JRiver Media Center

Gerbrand

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Try increasing buffering in Options > Audio > Output mode settings...

In the end I could solve the problem by no longer using ASIO. With WASAPI event style I do not experience any issues. This is a pity though, because I would prefer ASIO.

Gerbrand
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ccclapp

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Hi All

Sorry to be dumb, as I'm sure you've covered this, but I cant find it...

I'm trying to use REW DRC filters in MC.  

1)  Do I need ConvolutionVST or not?
2)  When exporting filters from REW should I select "Stereo" and 32 bit if exporting a full range correction filter for a 2-ch setup ?
3)  What is the feeling of REW vs others, eg audiolense?  I much prefer REW's interface and its more full measuring/reporting capability.  Are its correction filters as good as good as those from other product's?
4) for A/B comparison, Iv'e disabled "Normalize Filter Volume"  Is that a mistake (otherwise volume differences make A/B fairly meaningless. 
5) When listening to Generator->Pink PN in REW via loopback in MC, with convolution enabled it stutters, but doesn't with convolution unchecked.  Is there a MC setting to help that?  My PC is plenty powerful.

Thanks

Thanks

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zydeco

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Is an atom based set-up suitable for 6- or 8-channel convolution at 44kHz? (No need for video as machine is dedicated to music). I'm current running an old MOBO with Intel Q6600 (4 Cores, 2.4GHz) which works well but the MOBO is on it's last legs with support for a new solid-state h/d not solid. The thought was to move towards a full silent computer - as it sits near the listening position - and hence the interest in Atom (something like Intel ATOM Dual-Core D525 processor). Has anyone got experience with this type of processor running MC convolution?

Zydeco
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ccclapp

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If using REW and MC is it possible to do corrections on a per speaker basis?  JRiver seems only to take a single convolution file, but I would like to do corrections at the individual speaker level?

In other words: I would like to measure and correct each speaker in my 2.2 setup (4 speakers). After I have each speaker adjusted, using the 4 correction filters, I would like to measure the combined (corrected) response in REW and make an additional filter for further (combined speaker) adjustments.

Can this be done using REW and JRiver MC?  Or, can this only be done with other programs such as Audiolense?

Thanks


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zydeco

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Is an atom based set-up suitable for 6- or 8-channel convolution at 44kHz? (No need for video as machine is dedicated to music). I'm current running an old MOBO with Intel Q6600 (4 Cores, 2.4GHz) which works well but the MOBO is on it's last legs with support for a new solid-state h/d not solid. The thought was to move towards a full silent computer - as it sits near the listening position - and hence the interest in Atom (something like Intel ATOM Dual-Core D525 processor). Has anyone got experience with this type of processor running MC convolution?

Zydeco

I've done a quick test with a friend's Intel D510M Atom/MOBO and it seems to do 3-way x/o with 96kHz filters / 131072 taps with less than 20% CPU usage. I'm not interested in video and the initial delay is irrelevant). Does this sound right? If so, then it seems as if the ATOM processors can handle multi-way cross-over convolution.

Zydeco
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jacqlan111

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If using REW and MC is it possible to do corrections on a per speaker basis?  JRiver seems only to take a single convolution file, but I would like to do corrections at the individual speaker level?

In other words: I would like to measure and correct each speaker in my 2.2 setup (4 speakers). After I have each speaker adjusted, using the 4 correction filters, I would like to measure the combined (corrected) response in REW and make an additional filter for further (combined speaker) adjustments.

Can this be done using REW and JRiver MC?  Or, can this only be done with other programs such as Audiolense?
Thanks
You may take what you want here, just suggestions.  It depends of what you are trying to do.
First, why don't you do all the filters and X/O right in MC and use REW only for measurements?  For each speaker measurements, you can plug only the one you want to test, and so on.  This is what I did with success. 

MC EQ's are working fine and you can even monitor frequency response in real time in REW while you constructing the MC's filters. 
All wave additions and substractions should be included,  direct waves, reflections and from the other speaker your measuring.  3 sources
Check the acoustic XO's at your listening place, should be in phase from symmetric HP distance and work on the best XO fit from LFreq to HFreq speakers arrays left and right. 

jacques
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ccclapp

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First, why don't you do all the filters and X/O right in MC and use REW only for measurements?...

MC EQ's are working fine and you can even monitor frequency response in real time in REW while you constructing the MC's filters....
jacques

Jacques. Are you saying to use mc EQ not convolution?  I suppose I could use EQ to get the frequency response of each speaker in basic balance and then try to run REW through mc in loopback mode ( with the EQ filters applied) to do convolution on the combined all-channel sweep.

With a 2.2 setup ( 2 mains + 2 subs) I want to BOTH work the x/o's between subs and mains AND do room correction.  Certainly room correction works best with convolution rather than simple EQ.

What software are others using to make the filters for x/o and or DRC?
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jacqlan111

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Jacques. Are you saying to use mc EQ not convolution?  I suppose I could use EQ to get the frequency response of each speaker in basic balance and then try to run REW through mc in loopback mode ( with the EQ filters applied) to do convolution on the combined all-channel sweep.

With a 2.2 setup ( 2 mains + 2 subs) I want to BOTH work the x/o's between subs and mains AND do room correction.  Certainly room correction works best with convolution rather than simple EQ.

What software are others using to make the filters for x/o and or DRC?
Yes, I do not use convolution anymore.  I did not like REW's automatic target resolutions, it is more accurate to do it manually through MC's EQ on.  I used a pink noise wave file and REW's RTA graph and MC's parametric EQ side by side.  MC's EQing is directly responding on the REW'S RTA graph for monitoring results.  You should take all precautions on where to apply corrections and where not, using the"excess group delay graph".  MC's low pass and high pass filters are great for XO construction.  
It is always better to use sweep measurements if you can without stuttering to see the phase and delay responses!!  I can't anymore without stuttering:'(.  Working only with frequency responses is hazardous.
The best for you would be to follow Bob McCarthy's instructions
http://www.google.ca/imgres?q=bob+mccarthy+phase-alignment&um=1&hl=fr&sa=N&rlz=1R2ADRA_frCA388&biw=1024&bih=642&tbm=isch&tbnid=mLRdz7KA5BuS4M:&imgrefurl=http://bobmccarthy.wordpress.com/2010/03/11/phase-alignment-of-spectral-crossovers/&docid=hH4RNrTc1uP2dM&imgurl=http://bobmccarthy.files.wordpress.com/2010/03/fig2-35a-spectral-crossover-1khz.png&w=1827&h=1030&ei=Ke6XT9S8L5KY8gTVtaSVBg&zoom=1&iact=rc&dur=3&sig=114531235164415110848&page=1&tbnh=94&tbnw=167&start=0&ndsp=15&ved=1t:429,r:1,s:0,i:67&tx=87&ty=54
http://bobmccarthy.wordpress.com/2010/02/08/phase-alignment-of-subs-why-i-dont-use-the-impulse-response/
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jacqlan111

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Caleb
I am almost sure that you are aware of all those tricky filtering manipulations that I've mention above.   My reply was just to add some information on the subject. ;).
I have fixed XOs before without MC, using mediamonkey 2 channel outputs and doing 8 channel processing in the external Motu 828 and overall correction using the AIXcoustic EQ.  This way I was able to use REW sweeps without any problems.  But now that I am able to fix XOs and EQ from MC alone like I was hoping to do,  I still have this loopback stuttering issue :'(.
I have tried the exasound e18 multichannel DAC along with MC for XO + EQ correction which worked perfectly as long sweeps measurements where not involved. But I do need sweep measurements for accurate processing using MC loopback!!!. 
I'm still waiting for my new RME HDSP AES32 card to arrived, hoping to find a solution for my new upcoming setup.
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ccclapp

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Hi Jaques.  Thanks for the info.  I appreciate it.  I will look closely at your recommendations...
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ccclapp

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MC 64bit Convolution: Trying to Bring it All Together
« Reply #326 on: April 25, 2012, 11:50:31 pm »

I've fully re-read this entire thread today and have a better understanding.  For the benefit of others and myself, given the length of this thread, I am trying to summarize the concepts and procedures discussed in the preceding 7 pages, both through my (layman's) explanation below and by means of my questions and hoped for answers that follow:

-- I see that to use multi-driver/multi-ch filters in MC (meaning needing more than one filter) one accesses the desired filters by creating a config file following the specs for the Convolver config file format:
http://convolver.sourceforge.net/config.html
http://convolver.sourceforge.net/configegs.html

In that text config file one puts the path to all the desired correction filter files to be used in MC's convolution DSP.  In addition, one can add additional delays and other criteria.  This answers my prior question above as to how to use individual driver/speaker corrections given that MC only accepts a single file.

I also understand that in addition to (or in place of) the above one can use the Room Correction and/or the PEQ sections of MC DSP studio for similar purposes, e.g. delays, channel mapping, x/o's, etc, etc

After following all of the links and/or references in this thread and reading everything on all referenced sites (WOW!), I understand that in general, X/O and DRC corrections are best handled via convolution filters, rather than the more simple PEQ and/or MC Room Correction, because the former will better handle time domain, specifically delays and phase shifts, in addition to frequency adjustments handled by the latter. So even when speaking of a seemingly simple single-speaker X/O (vs an even more complex room correction), being able to include delays and phase shifts (and other components) moves one out of minimum-phase adjustments and into linear-phase adjustments yielding textbook results, potentially perfect in both frequency and time.  This is the fundamental difference between electronic (passive) X/O and computer (active) X/O.  Thus, if using the PEQ and/or Room Correction components of MC DSP studio and NOT the Convolution component, one is, at best, matching passive X/O and never achieving active X/O potential.  On the other hand, two important consequences of adding delay and phase (linear-phase adjustments) are enhanced:  1) correction artifacts, i.e. "pre-ringing" (hearing parts of the correction sooner in time than the direct signal) and, 2) even greater single-listening-position sensitivity (all other locations will likely be even worse).  I'm sure I've at least partially misstated this, but hopefully not fundamentally so.

Understanding the above (and welcoming any corrections others will offer), I have the following questions:

1)  Do we manually create the config text file, or will programs generate this for us (likely in need of editing).  E.G. I understand that DRCDesigner may create the config file for DRC generated correction filters.

2)  Recognizing that REW may not be as robust as other products like Acourate and Audiolense, if desired can our config file point to REW correction files, or are they somehow not compatible with the config file usage?

3)  If one desired, can a config file point to filters made by more than one program (e.g. if one chose to do per speaker corrections in REW and then combined post-adjusted room correction in another program, assuming all filters are .wav files)

4) Recognizing the limitations described above, if one wanted to use MC PEQ and/or Room Correction in addition to convolution, what should be the order of each DSP component, i.e. should PEQ come before Convolution?  Why?

5)  @ TheLion describes he uses Accurate for filters, but then uses Audiolense to calculate delays, which he then manually adds to the config file, because Accurate does not calculate delays (as well or at all?).  Why not do it all in Audiolense?

6) @ TheLion describes he uses hardware DSP on his subs and SW DSP on everything else.  Why not do it all in SW?  To do otherwise implies a HW solution is superior to 64bit convolution in its present state-of-the-art.  Is that your feeling?  Then why not do it all in HW?

7) @ TheLion describes he Audiolense does not correctly calculate the delay for his subs.  How do you know what the correct delay SHOULD BE? Is it only based on relative distance, or other factors?  Why do you think the SW calculation is wrong?

8 )  Being sensitive that Uli has contributed greatly to this thread and thus MC's execution of convolution, what are the pluses/minuses of Acourate vs Audiolense vs DRC, REW, etc

9)  As Uli pointed out, because @ TheLion is summing both the frequencies below 80hz from all 7 separate speaker channels and the LFE channel below 160 hz, all to his subs , and because each of those 7 speakers/channels are at different distances and thus have different delays (and are different than the LFE), the delay  (and likely the phase) from the summed signals to the subs must be all messed up (7 different delays/phase x 2 signal components coming to 2 subs, it must be completely smeared), because at present MC does not support additional parameters for summed delays.  Given your precision, why isn't this a problem for you?    

10) In concept, with a multi-driver/multi channel setup such as @ hulkssor or @ TheLion, what is the workflow to create X/O's, balance each speaker and apply room correction?

I prefer to do XO and DRC incrementally.  By this I mean I would like to proceed asfollows:

--A)  Set speaker locations while running RTA w Pink PNoise, triangle formulas, laser measurements and other conventional methods;
--B)  Take near-field frequency response measurements ofeach driver with band limited frequency sweeps to determine sweet range of eachdriver and successive driver to set X/O points and slopes;
--C)   Measuring mid-field and in exact path betweenspeaker and listening position and at exact listening height, run DRC SW tocreate linear-phase filter to set delay and phase adjustments for the driverson individual speaker basis, based on that speaker's relative acoustic centerfor its individual drivers;
--D)  Load these per speaker correction filters into DRC software or JRiver MC vialoopback feature http://yabb.jriver.com/interact/index.php?topic=70242.0and then use DRC SW for combined response (single sweep to allchannels/speakers) adjustment room correction based on desired "housecurve".
--E)  Create required Convolver config text filepointing to the above generated filters and load into JRiver MC convolution
--F)   Load these combined correction filters into DRC software or JRiver MC via loopbackfeature with convolution applied, re-measure from listening positionand adjust existing filters, as desired/needed.

Does anyone know in DRC SW one can I load incremental and/orfinal adjustment filters and perform additional measurements/corrections "ontop of" existing filters?  


Sorry for the long post, but as stated above Im hoping to bring together much of what has been discussed and to reconciled some inconsistencies from the preceding 7 pages of this thread.

I look forward to your wisdom, answers and corrections.

Thanks!! and thanks Matt & JR for this feature!!
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jacqlan111

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Re: MC 64bit Convolution: Trying to Bring it All Together
« Reply #327 on: April 26, 2012, 10:22:48 am »

Thus, if using the PEQ and/or Room Correction components of MC DSP studio and NOT the Convolution component, one is, at best, matching passive X/O and never achieving active X/O potential.  On the other hand, two important consequences of adding delay and phase (linear-phase adjustments) are enhanced:  1) correction artifacts, i.e. "pre-ringing" (hearing parts of the correction sooner in time than the direct signal) and, 2) even greater single-listening-position sensitivity (all other locations will likely be even worse).  I'm sure I've at least partially misstated this, but hopefully not fundamentally so.
WOW!!  Caleb and thanks for this synthesis it is very impressive.
I look foreward for answers.   I'm overwhelmed.
I must ask you this trivial question.  Why MC's PEQ could not achieve active X/O potential?  It is digitally processed,  X/O slopes (order) and frequencies position are adjustable, time delays can be managed too.  Phase is also manageable by the way isn't it?  One can move speakers physically too (DIY). 
I must admit that creating those convolution files does not attract me very much.  If I can do without it 8)
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ccclapp

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Re: MC 64bit Convolution: Trying to Bring it All Together
« Reply #328 on: April 26, 2012, 10:50:37 am »

I must ask you this trivial question.  Why MC's PEQ could not achieve active X/O potential?  It is digitally processed,  X/O slopes (order) and frequencies position are adjustable, time delays can be managed too.  Phase is also manageable by the way isn't it?  One can move speakers physically too (DIY). 

Above Erich (@v_erich) posted the following link to a white paper by Uli (Acourate), which gives a good and reasonably understandable (for we mortals) description of X/O's, minimum-phase and linear-phase.  Here it is again:

http://www.acourate.com/XOWhitePaper.pdf

This should answer the questions you posed (assuming I correctly understand what it says).
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ccclapp

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Re: MC 64bit Convolution: Trying to Bring it All Together
« Reply #329 on: April 26, 2012, 11:06:58 am »

10)...I prefer to do XO and DRC incrementally.  By this I mean I would like to proceed asfollows:

--A)  Set speaker locations while running RTA w Pink PNoise, triangle formulas, laser measurements and other conventional methods;
--B)  Take near-field frequency response measurements ofeach driver with band limited frequency sweeps to determine sweet range of eachdriver and successive driver to set X/O points and slopes;
--C)   Measuring mid-field and in exact path betweenspeaker and listening position and at exact listening height, run DRC SW tocreate linear-phase filter to set delay and phase adjustments for the driverson individual speaker basis, based on that speaker's relative acoustic centerfor its individual drivers;
--D)  Load these per speaker correction filters into DRC software or JRiver MC vialoopback feature http://yabb.jriver.com/interact/index.php?topic=70242.0and then use DRC SW for combined response (single sweep to allchannels/speakers) adjustment room correction based on desired "housecurve".
--E)  Create required Convolver config text filepointing to the above generated filters and load into JRiver MC convolution
--F)   Load these combined correction filters into DRC software or JRiver MC via loopbackfeature with convolution applied, re-measure from listening positionand adjust existing filters, as desired/needed.

Does anyone know in DRC SW one can I load incremental and/orfinal adjustment filters and perform additional measurements/corrections "ontop of" existing filters? 


Uli replied to this question on the Acourate forum (in about 30 seconds, very impressive customer support!!), as follows:

"So with Acourate you can:
1. step: create crossover filters
2. step load the filters in the Acourate logsweep recorder and measure a single driver or all drivers
3. step: if you like you can use single driver measurements to linearize the driver. This results in combined crossover/linarization filters
4. step. load the filters in the Acourate logsweep recorder and measure a single driver (e.g. for verification) or all drivers
5. step: compute room correction filters and get filter combined of crossover, driver linearization and room correction
6. step: load the filters in the Acourate logsweep recorder and measure the system for verification (if desired)
7. step: load the final filters into JRiver and enjoy the music
I hope this has answered your questions
--Uli"

It is great to be able to "step" into filters incrementally, vs push one button, hope for the best and never understand what you've done.
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BerntR

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Re: MC 64bit Convolution: Trying to Bring it All Together
« Reply #330 on: April 26, 2012, 05:28:28 pm »

7) @ TheLion describes he Audiolense does not correctly calculate the delay for his subs.  How do you know what the correct delay SHOULD BE? Is it only based on relative distance, or other factors?  Why do you think the SW calculation is wrong?

The reason that the calculated sub delay doesn't work for TheLion is most probably that the delay after correction is not comparable to the delay before correction. At lower frequencies the difference due to a low pass filter for instance, can be substantial. The delay management in Audiolense works well enough to keep the speakers timed within a sample under normal circumstances.
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JimH

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TheLoin should be TheLion.  I've changed it.
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hulkss

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TheLoin should be TheLion.  I've changed it.
Now that's funny  :)
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BradC

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Re: MC 64bit Convolution: Trying to Bring it All Together
« Reply #333 on: April 26, 2012, 10:57:59 pm »

I'll have a go at some answers, see below




1)  Do we manually create the config text file, or will programs generate this for us (likely in need of editing).  E.G. I understand that DRCDesigner may create the config file for DRC generated correction filters.

acourate doesn't but I believe that audiolense will create the file

2)  Recognizing that REW may not be as robust as other products like Acourate and Audiolense, if desired can our config file point to REW correction files, or are they somehow not compatible with the config file usage?

Unless REW has updates that I'm not aware of, REW does not calculate files for convolution

3)  If one desired, can a config file point to filters made by more than one program (e.g. if one chose to do per speaker corrections in REW and then combined post-adjusted room correction in another program, assuming all filters are .wav files)

yes

4) Recognizing the limitations described above, if one wanted to use MC PEQ and/or Room Correction in addition to convolution, what should be the order of each DSP component, i.e. should PEQ come before Convolution?  Why?

ordering can affect the result see below

5)  @ TheLion describes he uses Accurate for filters, but then uses Audiolense to calculate delays, which he then manually adds to the config file, because Accurate does not calculate delays (as well or at all?).  Why not do it all in Audiolense?


6) @ TheLion describes he uses hardware DSP on his subs and SW DSP on everything else.  Why not do it all in SW?  To do otherwise implies a HW solution is superior to 64bit convolution in its present state-of-the-art.  Is that your feeling?  Then why not do it all in HW?

7) @ TheLion describes he Audiolense does not correctly calculate the delay for his subs.  How do you know what the correct delay SHOULD BE? Is it only based on relative distance, or other factors?  Why do you think the SW calculation is wrong?

delay for subs are different as there are so few (often <1) wavelengths in a room. I have use dthe process where I started with the delay calculated from the distance, then used the RTA in REW and adjusted delays until I got the smoothest result.
Calculation of the delays for the sub in acourate and audiolense can be erroneous as the impulse response does not have a sharp peak.

8 )  Being sensitive that Uli has contributed greatly to this thread and thus MC's execution of convolution, what are the pluses/minuses of Acourate vs Audiolense vs DRC, REW, etc

REW does not do convolution filters (ie time domain)
I have used DRC and acourate, I find acourate much better. Apparently there are different philosophies for the psychoacoustic analysis.
I have not tried the audiolense demo, and it is more user friendly. TheLion reported that he preferred the results from acourate.

9)  As Uli pointed out, because @ TheLion is summing both the frequencies below 80hz from all 7 separate speaker channels and the LFE channel below 160 hz, all to his subs , and because each of those 7 speakers/channels are at different distances and thus have different delays (and are different than the LFE), the delay  (and likely the phase) from the summed signals to the subs must be all messed up (7 different delays/phase x 2 signal components coming to 2 subs, it must be completely smeared), because at present MC does not support additional parameters for summed delays.  Given your precision, why isn't this a problem for you?    

This can be fixed by performing the delays for speaker distance after the bass management. I am not sure where the delays are in the chain in the convolver in MC, but you could instead use the delays in the room correction DSP module, and place it after the convolver



Does anyone know in DRC SW one can I load incremental and/orfinal adjustment filters and perform additional measurements/corrections "ontop of" existing filters?  

-you could run multiple convolutions in series, but I don't think MC allows multiple instances of the DSP filters (could be wrong), so you would need a different convolver such as convolverVST or pristine space.
This would be tricky however, as you would need to do the measurement of the additional filter, with the first one in place already (acourate's logsweep recorder allows this). For changing house curves, this doesn't really make sense.
For separating room correction and active XOs it might make sense, but adds significant computational load (if filters are 64 bit)


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ccclapp

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Re: MC 64bit Convolution: Trying to Bring it All Together
« Reply #334 on: April 26, 2012, 11:28:05 pm »

The reason that the calculated sub delay doesn't work for TheLion is most probably that the delay after correction is not comparable to the delay before correction. At lower frequencies the difference due to a low pass filter for instance, can be substantial. The delay management in Audiolense works well enough to keep the speakers timed within a sample under normal circumstances.

Thanks @BerbtR, I look forward to learning more about the subtle aspects of Audiolense...
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ccclapp

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Re: MC 64bit Convolution: Trying to Bring it All Together
« Reply #335 on: April 26, 2012, 11:59:09 pm »

@BradC

Thanks for your informed reply to my myriad of questions!!  I appreciate it.

I'll have a go at some answers, see below

1)  Do we manually create the config text file, or will programs generate this for us (likely in need of editing).  E.G. I understand that DRCDesigner may create the config file for DRC generated correction filters.

acourate doesn't but I believe that audiolense will create the file


Can/will acourate and audiolense please verify the above?


2)  Recognizing that REW may not be as robust as other products like Acourate and Audiolense, if desired can our config file point to REW correction files, or are they somehow not compatible with the config file usage?

Unless REW has updates that I'm not aware of, REW does not calculate files for convolution

REW does correction filters as .wav files.  These may or may not be as robust as Audiolense / Acurate / DRC, however I do not understand why the REW .wav filters cannot be referenced in a cofig file.  Please explain...thanks

4) Recognizing the limitations described above, if one wanted to use MC PEQ and/or Room Correction in addition to convolution, what should be the order of each DSP component, i.e. should PEQ come before Convolution?  Why?

ordering can affect the result see below

I do not see/follow youre reference "ordering can affect the result see below"  Please be more specific...Thanks

7) @ TheLion describes he Audiolense does not correctly calculate the delay for his subs.  How do you know what the correct delay SHOULD BE? Is it only based on relative distance, or other factors?  Why do you think the SW calculation is wrong?

delay for subs are different as there are so few (often <1) wavelengths in a room. I have use dthe process where I started with the delay calculated from the distance, then used the RTA in REW and adjusted delays until I got the smoothest result.
Calculation of the delays for the sub in acourate and audiolense can be erroneous as the impulse response does not have a sharp peak.

Would Audiolense and Acourate please add their perspective to this...Thanks


9)  As Uli pointed out, because @ TheLion is summing both the frequencies below 80hz from all 7 separate speaker channels and the LFE channel below 160 hz, all to his subs , and because each of those 7 speakers/channels are at different distances and thus have different delays (and are different than the LFE), the delay  (and likely the phase) from the summed signals to the subs must be all messed up (7 different delays/phase x 2 signal components coming to 2 subs, it must be completely smeared), because at present MC does not support additional parameters for summed delays.  Given your precision, why isn't this a problem for you?   

This can be fixed by performing the delays for speaker distance after the bass management. I am not sure where the delays are in the chain in the convolver in MC, but you could instead use the delays in the room correction DSP module, and place it after the convolver

Would Audiolense and Acourate please add their perspective to this...Thanks

8 )  Being sensitive that Uli has contributed greatly to this thread and thus MC's execution of convolution, what are the pluses/minuses of Acourate vs Audiolense vs DRC, REW, etc

REW does not do convolution filters (ie time domain)
I have used DRC and acourate, I find acourate much better. Apparently there are different philosophies for the psychoacoustic analysis.
I have not tried the audiolense demo, and it is more user friendly. TheLion reported that he preferred the results from acourate.

Would Audiolense and Acourate please add their perspective to this...Thanks

9)  As Uli pointed out, because @ TheLion is summing both the frequencies below 80hz from all 7 separate speaker channels and the LFE channel below 160 hz, all to his subs , and because each of those 7 speakers/channels are at different distances and thus have different delays (and are different than the LFE), the delay  (and likely the phase) from the summed signals to the subs must be all messed up (7 different delays/phase x 2 signal components coming to 2 subs, it must be completely smeared), because at present MC does not support additional parameters for summed delays.  Given your precision, why isn't this a problem for you?   

This can be fixed by performing the delays for speaker distance after the bass management. I am not sure where the delays are in the chain in the convolver in MC, but you could instead use the delays in the room correction DSP module, and place it after the convolver

Is this the recommended workflow?


BradC, just checking...do you have any identity of interest (affiliation) with either Audiolense or Acourate?  No offence implied, just trying to keep all information clear.

Thanks very much for your thoughtful and knowledgeable reply!!

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BradC

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Re: MC 64bit Convolution: Trying to Bring it All Together
« Reply #336 on: April 27, 2012, 12:56:55 am »



Can/will acourate and audiolense please verify the above?

I believe that audiolense will, but acourate won't generate the convolver config file. It's not that hard though


REW does correction filters as .wav files.  These may or may not be as robust as Audiolense / Acurate / DRC, however I do not understand why the REW .wav filters cannot be referenced in a cofig file.  Please explain...thanks

It seems that you could use REW filters in a convolver. They would however, only be minimum phase correction. YOu would probably not get much (if any) improvement in the impulse response of your system, unlike with linear phase filters

If you mix correction files from different programs in one convolver config file, make sure they are all for the same sampling rate and are of the same length

I do not see/follow youre reference "ordering can affect the result see below"  Please be more specific...Thanks

when using delays, the order of the delay can affect bass management

Would Audiolense and Acourate please add their perspective to this...Thanks


Would Audiolense and Acourate please add their perspective to this...Thanks

Would Audiolense and Acourate please add their perspective to this...Thanks
 
Is this the recommended workflow?


BradC, just checking...do you have any identity of interest (affiliation) with either Audiolense or Acourate?  No offence implied, just trying to keep all information clear.

I have purchased acourate and used DRC in the past.

Thanks very much for your thoughtful and knowledgeable reply!!


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jacqlan111

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I take the chance to move this post here.
Not this thing again ?  Try to see it through newbies eyes.
Many pages where dedicated for this wasapi loopback option.  MC experts (right here) might see it as an easy ride.  It's not that obvious to everyone (me).
Can sweep measurements can be done without using the live://loopback URL option? (for ASIO and win XP)
I am not familiar with the VST convolver softwares mentioned frequently overhere (audiolense, convolver.sourceforge, DRC........).  What is the main differences with the REW measurement capacities and convolution IR WAV files imported in MC?  
Can someone gives a quick idea of their uses and if it is the case, their sweep measurements MC playback possibilities, without using the Wasapi loopback?
I saw that config text files for EQ and X/O DSP multichannel processing can be imported into MC.
I, and surely others are interested to make sweep measurements and being able to do it through MC playback to confirm their DSP filtering.
What is the easy way to do it, even if a VST software purchase is needed?
A short overview would be appreciated.
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ccclapp

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I'm happy to give a summary reply to Jacques's question above, but first, since this is the MC thread, it would be helpful to understand one aspect of MC for convolution, as follows:

At various times with various measurement/convolution software, especially REW, its useful to be able to run live measurements through MC with MC applying convolution, e.g. to compare before/after convolution measurements.  I know this can be done recording the sweep and playing back in MC and using the RTA in REW.  However, that will only give frequency response and not waterfall and other time domain analysis.

My question is:  Will using the MC loopback feature while doing live sweeps & measurements in REW yield reliable "corrected" measurements?  Or, will MC loopback add delay, phase shifts or other distortions that will render the corrected live sweeps measurements unreliable?

If  MC loopback does harm the measurement, is a workaround to do ALL ("before" and "after" correction) measurements via loopback?  That way, both the "before" and "after" results will include and MC loopback delay, phase shift, etc, etc.  Would that not make the comparison meaningful?

Thanks!!
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ccclapp

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...partially answering my own question above, I just now I confirmed I get a good signal from REW through MC in loopback (no stutters, clicks, pops).  MC  Playback Options is setto WASPI Eventstyle with hardware buffer at anything from 10-100 ms to my firewire Mytek 8x192 DAC.  If I use ASIO instead of WASPI I get stutter.  I assume I should use the least possible buffer, or does it matter?

Correct me if I'm wrong, but I also assume it doesn't matter what sound device should be set to default for REW to play on (to be looped back to MC).   

Does anyone know REW well enough to suggest what graph I should inspect to determine of Im getting any additional delay, phase shifts, etc from loopback that harms by before/after comparison?

Also, does anyone know why most have said we can only use REW RTA, not MC loopback, to inspect before vs after results??  It seems to me the way to do it is via MC loopback, unless there are issues that arent apparent.

PS:  Since I posted this elsewhere, Ill report these findings there as well..

Thanks
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BradC

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If your soundcard can do loopback in hardware, you can use a VST host, such as plogue bidule to allow REW to run with the corrections applied.
Even if your hardware can't loopback internally, if you can assign a software channel to any hardware output in the mixer, then you could use a physical loopback cable
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ccclapp

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If your soundcard can do loopback in hardware, you can use a VST host, such as plogue bidule to allow REW to run with the corrections applied.
Even if your hardware can't loopback internally, if you can assign a software channel to any hardware output in the mixer, then you could use a physical loopback cable

Jrmc does this via software. The only question is if its (or for that matter a hw loopback) causes time domain issues that render the resulting measurements of little benefit .
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jacqlan111

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MC does this via software. The only question is if its (or for that matter a hw loopback) causes time domain issues that render the resulting measurements of little benefit .
Hi
here a idea that I submit without having tested it.
If a hardware loopback is possible (like in REW time reference loopback or sound card calibration), would it be possible to use the REW card calibration option and calibrate with MC in the path, without MC's corrections? That could eliminate delays issues and render possible comparison before and after corrections?  
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jacqlan111

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If your soundcard can do loopback in hardware, you can use a VST host, such as plogue bidule to allow REW to run with the corrections applied.
Even if your hardware can't loopback internally, if you can assign a software channel to any hardware output in the mixer, then you could use a physical loopback cable
So you say that a VST host "as plogue bidule " could allow me to run REW sweeps through MC without using the  "live://loopback" url option?
If you do not recall, the "live://loopback" is not working in my win XP setup.   the VST option could work and solve my problem?
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ccclapp

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...would it be possible to use the REW card calibration option and calibrate with MC in the path...? That could eliminate delays issues and render possible comparison before and after corrections?  

Yes, I was thinking that as well:  if we do a calibration using MC loopback, would that adjust for any detrimental time domain impact?
Further, if we take both the "before and "after" measurements in loopback wouldn't that mitigate any variance caused by the loopback setup?  Has anyone tested this?
Again, can anyone describe the REW test to determine is any variance exists from loopback?  Ill do the tests...
Thanks
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BradC

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So you say that a VST host "as plogue bidule " could allow me to run REW sweeps through MC without using the  "live://loopback" url option?
If you do not recall, the "live://loopback" is not working in my win XP setup.   the VST option could work and solve my problem?

Yes, if your hardware can do loopback.
I use an RME fireface 800 that can send the hardware outputs to an input channel internally.
Plogue bidule uses the input channels, applies delay, convolution and bass management via VST plugins and sends the results to software output channels. The software outputs are routed to different hardware outputs
I use this process from applying correction to TMT5 and windows media center.

I have compared the before and after correction using REW and plogue bidule. The 'after' looks as expected (matches target curve)
The only catch is that if you move the microphone between before and after measurements, the results no longer look as good.
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mojave

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Also, does anyone know why most have said we can only use REW RTA, not MC loopback, to inspect before vs after results??  It seems to me the way to do it is via MC loopback, unless there are issues that arent apparent.
I posted how to use the REW RTA and was using it last October, but that was before MC had the loopback feature. I use the loopback for both before and after REW measurements so my signal chain is identical. There is no need to use REW in standalone mode anymore.
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ccclapp

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Thanks

Ill be running tests today to determine if sweeps show and time-domain changes in loopback vs non-loopback.  Ill include REW calibration and hardware 2nd channel mic loopback...

Ill post results so others can evaluate any concerns over effectiveness of this procedure...
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jacqlan111

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Thanks

Ill be running tests today to determine if sweeps show and time-domain changes in loopback vs non-loopback.  Ill include REW calibration and hardware 2nd channel mic loopback...

Ill post results so others can evaluate any concerns over effectiveness of this procedure...
I would guess that you could see changes with the phase graph in rew where you can save the first sweep measurement from REW alone and save the other one passing into MC and overlay the two phase curves.  The overlays of the impulse responses will be nice too.
I did so many delays measurements in REW for aligning physically all my DIY speakers, it worked very good.

If they are identical, you could go foreward with it, (lucky guy who can do MC loopback (*)). If they are not identical, but always have the same delay, then the same error will be made for every measurements, then it does not matter??  Is there any other consideration to be aware of?
RTA frequency measurements has to be used in last when all other delays and phase corrections have been made with sweeps.


(*)I need to know which VST software to go around my problem?  Don't want to hurt anybody, but suggestions are welcome.
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jacqlan111

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Yes, if your hardware can do loopback.
I use an RME fireface 800 that can send the hardware outputs to an input channel internally.
Plogue bidule uses the input channels, applies delay, convolution and bass management via VST plugins and sends the results to software output channels. The software outputs are routed to different hardware outputs
I use this process from applying correction to TMT5 and windows media center.

I have compared the before and after correction using REW and plogue bidule. The 'after' looks as expected (matches target curve)
The only catch is that if you move the microphone between before and after measurements, the results no longer look as good.
OK, I think I could do the same when my RME HDSP card will be installed. 
If the Plogue bidule uses the RME input channels  (which consist in a sweep coming from the REW output setting and has been sent to one RME output channel) and MC playback is directed to an other RME output with corrections made in the VST plugin?
Is it that simple? ;)
I could use the MC EQ on top of the plogue bidule or it as to be the plogue bidule corrections only?

A bit confusing!  I will have to try it in real.
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