"Wall-integrated" corner loaded line array with Vifa TC9 drivers

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Thanks guys ... I learned something today!

As mentioned, these software hacks work, but there are often glitches along the way.
I ran into clock sync with the Mac and Aggregate a few times. It's not perfect.

Nothing will beat a simple solution.

I stand by the option of using HDMI to AVR for movies, and a USB stereo DAC for music listening, powered by a cool NUC or similar.
 
I agree, simple is usually the best path forward! Is the USB stereo DAC path because we are assuming that the DAC in the AVR is not as good as the one in the standalone DAC? I tend to agree that's the case academically speaking, but it may not be worth the incremental improvement given I'll need to swap cables to make it happen. Plus, the DAC in the XMC-1 is reasonably good and my XDA-2 DAC isn't exactly world class.

So, perhaps even simpler, I'll just make it super simple and just go HDMI and be done. Of course, this option means I need to keep the AVR, but there's other value in that as well. Plus, I don't need to get as nice a PC now. Pricing out one of the cheaper ones, I can be out the door well under $300, so that's not too bad. I may even be able to get away with an existing Mac laptop if I can be happy with the Mac version of the software! ;)
 
Thanks for all the resources! I think I'll get a NUC and associated RAM/HDD on black friday deal and get the one with TOSLINK and then I can compare using that output for 2 channel and HDMI through the AVR. Then I can make a head to head comparison for my own personal edification. It's a relatively small investment that can be repurposed to one of my kids if I feel the need to upgrade later on. And swapping cables can be planned for in my cabinet build so it's not too onerous. Will be fun!

Now, time to get my subwoofers built!
 
Well I got my Intel NUC, loaded up REW and JRiver, dug out the microphone, and started down the rabbit hole of DSP.

It's going to take a while to get where I want to be and perhaps there will never be an end to the tweaking but for now, a few questions. Hopefully these aren't terribly embarrassing...

1. At a high level, do I have my process right?
- Microphone about 8" from speaker about in the middle of the height
- What smoothing is best? I'm thinking between 1/12 and 1/24
- Any commentary on my target or measurement settings?
- I can load the Filter Impulse Response as .wav file into JRiver... is that the right one?
2. When I play back with convolution after loading, I get a delay... about one second. At first, I thought it was because I had used "acoustic timing reference" but when I tried again with no reference I still had the same problem.
3. When playing through my AVR, does it make sense to use any of the parametric EQ capabilities (Emotiva XMC-1) or just leave it alone with convolution?
4. What's the difference between impulse response and filters impulse response? Is the impulse response just the dirac pulse and resultant ringing? Whereas the filters response is what Jriver uses for convolution?


That's it for now... thanks for your help!
 
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A few things, others will chime in:

- put the mic at your listening position, ear level when seated.
- I usually go for sweeps starting at 20Hz to 20,000Hz.
- I used 1/12 smoothing, but lately, I like to use Psychoacoustic, as it is easier to read.
- target is a preference, so adjust to taste.

- the delay when using convolution is normal. It's the time the JRiver filter needs to apply it in real time. 1 sec is a bit long though. Have a look online on how to optimise audio in a PC. It also depends what kind of interface and drivers you are using. For example, ASIO4ALL is just a band-aid solution and not very good.

- don't add the AVR EQ into the mix. Keep it simple and use the EQ in JRiver only. You can use the AVR to cross to a sub though, if you plan to add one.
 
I agree with perceval, I'd put the mic in the listeners position due to measuring an array of drivers. The geometry of distance from each driver to the microphone changes if you measure at 8" vs your actual listening position.

I do not know what kind of convolution you use. Are you saving the REW filters as a FIR correction? Those are all IIR corrections and you should not need to get any delay.

Another way of getting exactly the same is to put those REW filter settings in one of JRiver's PEQ banks. It might need you to set the REW type of equalizer to something other than "Generic" to get the exact same "Q". That's why there are so many available as pre-set. The RePhase one reasonably corresponds to JRiver's "Q" settings if I recall correctly.

Any delay in the FIR filter will result from the way the FIR filter is setup or saved. IIR do not need to have any delay. Whenever you actually change phase (to be different from the EQ cuts and/or boosts) too, then you'd need delay. How much would be dependant on the actual used filter resolution and the sample rate you use in the audio chain. The rephrase thread has a lot of information on that should you want to learn more about it. RePhase offers to change phase independent from frequency.

REW's filters are not changing the phase independent from frequency last time I looked. You can exchange certain things from REW to RePhase though, what route do you use?

Personally I do the tweaking of phase and magnitude with DRC-FIR.

I use whatever smoothing for the task I am after, there isn't a magical one that works for every job. I look at unfiltered and filtered plots to determine my goals. My thread is full of those plots, covering from Psychoacoustic to 1/48. And lots of frequency dependent windows. They all have a different way to look at the same information. REW has many tabs, each of them shows the same measurement in a different way. All of them have a role to play to learn what your system and room are doing.
 
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As wesayso talk about REW makes IIR filtering so theoretical there should not be any delay, haven't tried this scenario before myself but think before export IR for correction then look at IR window for that correction and see if there is unusual long time before attack of real impulse itself and if there is then try with "Controls to see if that could be edited out, because for IIR filtering all correction is positioned after top of impulse itself and why IIR filtering is real time correction.
 
Hmm, ok. I was confused by the JRiver convolution wiki and I was indeed using the IIR .wav file in the convolution tab. And the delay was not a total delay, more like a reverb in that I would hear the same thing a second later. I assume that's because I was doing it all wrong!

I think my next stab will be to remeasure at the listening position. Then input the PEQ settings from REW into Jriver.

We'll see how that goes and then I'll take a stab at DRC-FIR for the phase tweaking. Is it true that DRC-FIR will generate a .wav file that will go into the convolution tab of JRiver as I was *trying* to do with the REW IIR .wav file?

One other question... I assume I leave my AVR in "direct" mode where it does zero processing then? As a reminder, my signal chain is Intel NUC - HDMI output - AVR - monoblock - speakers.
 
So the REW FIR output is not really FIR at all. It's just the IIR parametric EQ configured as a FIR file so that it can be used via the convolution .wav file functionality, correct?

In any case, I guess I did something else wrong as to your point, it should have at least worked to a certain degree, without the weird delay thing I was experiencing.

I'll give it another shot tonight.

If I get DRC-FIR up and running, will that generate a file that includes both IIR and FIR in one convolution file such that I would remove the parametric EQ settings I would have previously input into the parametric EQ section? I'm confused how I would get DRC-FIR to use the Jriver EQ settings so that the FIR convolution file and the IIR PEQ settings would be independent of each other if that makes sense.
 
Ok... quick update.

I got REW to output PEQ files that I could simply import into the XMC-1 with a USB stick. It was super easy actually. And it sounds pretty darn good if I do say so myself. I've been living with the Dirac result which to tell you the truth, had me worried that I messed up the speaker build somehow. Not sure if I did something wrong with the Dirac setup, but man, I was not impressed. After figuring out the basics of REW, getting a measurement done was super fast and easy. I may play with some different house curves but it's already pretty darn good!

Second, I got DRC up and running and thought I got it to generate filters but I can't find them on my PC. As far as I can tell, they should be in the convolution filter folder. But it's empty. And I can't seem to find them anywhere despite the program telling me that it successfully generated the filters. Any ideas?

Also, the DRC tutorial says to install the convolved VST plugin but I'm guessing that's not necessary anymore since the tab exists already in the DSP window? I should just be able to upload the file... once I find it.

Question there though... DRC-FIR generates two files, left and right. How do I use both files in JRiver? I only see one file upload option?
 
Jriver convolution

For JRiver two channel convolution is the easy way in one can just create a stereo wav-file and point to that, when per channel correction is needed one shall point to a config-file that is simply a homebrew txt-file that then again point to the various wav-files. That txt file shall follow a known format as seen here where the one at top probably is good enough for your case Config examples. Because sample rate conversion can add some distortion suggest you keep an eye on create correction files in same rate as engine is running and if engine is not set as locked but is dynamic into "DSP Studio/Output Format" then see in first link below rules is to get engine pick among multiple correction file rates.

More links for subject:
Convolution - JRiverWiki

Config file
 
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