# Measuring the results of convolution



## 3ll3d00d (Jun 6, 2006)

I have moved to a jriver+acourate setup and am having trouble measuring the convolved response. It seems like the latency of the filter is large enough to confuse the REW sweep. I am using a focusrite saffire device and hence saffire mixcontrol as the mixer app. 

My measurement setup is as follows

Set REW output to DAW7 
Set DAW7 -> Loopback1 in mixcontrol
Open Jriver Asio Line In for 1 channel offset 14 
Set DAW1 -> Loopback2 in mixcontrol
Set REW input to Loopback2

This works as expected with convolution off but fails with convolution on.

Does anyone have any idea as to how to get this to work?

Thanks
Matt


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## mojave (Dec 30, 2006)

There are two ways:

1. Use the JRiver WASAPI loopback. You set another audio device as your default audio device in Windows. For example, set your motherboard's optical output as the default. Set REW to use the default audio device. In JRiver go to Tools > Open Live > WASAPI loopback. Do a sweep in REW and it will play through JRiver and its DSP.

2. Use the RTA feature of REW with JRiver. You will get the exact same frequency response as method #1, but will not get an impulse response. I have a guide I wrote a couple years ago on JRiver's forum here. This method eliminates an issues with latency caused by the Acourate filter.

Both methods eliminate using mixcontroller and will work much more smoothly. You have to have the Focusrite set as your audio device in JRiver.


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## Mitchco (Apr 12, 2011)

I see mojave beat me to it 

Couple more support links:

http://yabb.jriver.com/interact/index.php?topic=77800.msg528386#msg528386

http://yabb.jriver.com/interact/index.php?topic=70242.msg517062#msg517062

Cheers!


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## 3ll3d00d (Jun 6, 2006)

Thanks for the references. 

Just to be clear, there are no glitches with the asio line in setup that I can hear. It works v smoothly, it just seems the latency is far greater than REW expects. I have had WASAPI loopback working as well but haven't tried it in REW with convolution active, won't the same latency apply in this case though? It's not clear to me how this helps if it is a case of REW getting confused by the excessive delay.

The reason I liked the asio line in method is because it gives me a timing reference as well as being simpler to setup. This is not critical obviously, it just seemed a more elegant method (though I prefer working to elegant!)


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## Mitchco (Apr 12, 2011)

I tried this morning and did not have any issues. I don't use a timing reference.

REW output->JRiver ASIO line input->Convolution engine (with 3-way XO, time alignment and room correction FIR filter)-->Lynx Hilo DAC->6 amps->speakers->measurement mic->mic preamp->Lynx Hilo ADC->REW input.

Attached are a few charts of the 1/12 octave frequency response, impulse, and step response measured at the listening position some 10 feet away using REW's default window size. Very similar to what Acourate predicts, but there will be some differences due to different windowing specs.

JRiver's Convolution engine should take into consideration the FIR filter latency so that REW is in sync. Did you try with no timing reference?

Regards, Mitch


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## 3ll3d00d (Jun 6, 2006)

Thanks for that. Your post made a facepalm lightbulb go off in my head  I suppose the arrival of the timing reference way earlier than the actual measurement makes the timing reference thoroughly meaningless. I will repeat without timing next time I measure. It would be interesting to see what happens if I push the timing reference through jriver (as the R channel say so it is convolved too). It might be nonsense I guess, I don't know how jriver calculates the latency incurred by a filter.


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## Mitchco (Apr 12, 2011)

Cool. There is a long Convolver thread at JRiver that explains how the delay/latency is taken care of: http://yabb.jriver.com/interact/index.php?topic=68828.0 I think somewhere on page 8... I also watch movies with JRiver using Convolution with no lip sync issues.

There is a bit of math here in section 2.1.4 that allows one to calculate the delay of a linear phase FIR filter: http://www.dspguru.com/dsp/faqs/fir/properties

Will be interesting to see your measurements.

Cheers, Mitch


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## 3ll3d00d (Jun 6, 2006)

I tried without timing reference and it still failed to measure properly, I then realised I had jriver set to upsample to 96kHz and removing this option seems to have done the trick as I can now measure without any issues.

I was curious to see how this compared on a chart against my old audyssey based setup so my graphs include the previous results from audyssey for comparison. The dip at 90Hz is because I have the XO wrong (I think the sub needs to cross to the mains at 110Hz) & I'm not correcting again until the new driver for my sub turns up.

Certainly an improvement to be seen anyway, blue is audyssey, gold is acourate.


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## Mitchco (Apr 12, 2011)

Looks good! Especially in the time domain. What are your thoughts with respect to the differences you hear between Audyssey and Acourate, if you want to share...


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## 3ll3d00d (Jun 6, 2006)

Mitchco said:


> Looks good! Especially in the time domain. What are your thoughts with respect to the differences you hear between Audyssey and Acourate, if you want to share...


Warning: I may get breathlessly excited during writing this post 

It is just incredibly good. The clarity is remarkable with such crystal clear separation between the various instruments. Ultimately decent recordings are just so much more musical than before and by that I mean enjoyable to listen to. I don't know exactly which changes contribute to that quality but I like anyway. 

The only downside is that some recordings are exposed badly in comparison, quite annoying really!


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## Mitchco (Apr 12, 2011)

3ll3d00d said:


> Warning: I may get breathlessly excited during writing this post
> 
> It is just incredibly good. The clarity is remarkable with such crystal clear separation between the various instruments. Ultimately decent recordings are just so much more musical than before and by that I mean enjoyable to listen to. I don't know exactly which changes contribute to that quality but I like anyway.
> 
> The only downside is that some recordings are exposed badly in comparison, quite annoying really!


 Sure, I get it. Your comments are similar to mine and others, like in the 3rd paragraph here: http://digitalroomcorrection.hk/http___www.digitalroomcorrection.hk_/Welcome.html

Why this is so, I believe is the difference between minimum phase and linear phase (FIR) correction filters. Here is some info: http://en.wikipedia.org/wiki/Linear_filter It's a bit heavy duty on the math, but one salient point with respect to IIR versus FIR filters: "Another advantage of FIR filters is that their impulse response can be made symmetric, which implies a response in the frequency domain that has zero phase at all frequencies (not considering a finite delay), which is absolutely impossible with any IIR filter." 

Another good read on this is Uli's http://files.computeraudiophile.com/2013/1202/XOWhitePaper.pdf

Enjoy the sound and stay well tuned!


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