[bcwang]

I think the easiest thing to do is place one speaker in the center of the room, optimize that speaker with any ad hoc room treatments you can muster, and the rotate your head and body. That way, you'll get the best in room performance of the speaker. Even better, if you have a DSP unit with A to D analogue inputs (such as the miniDSP 2X4 HD, or the SHD) you can run something like Dirac Live on that speaker, after doing room treatments, and get the triple benefit of getting the effect of a speaker playing in a bigger room (b/c its farther away from room boundaries than it would otherwise be), ad hoc room treatment (b/c you may not like what they do to your room decor if used on an everyday basis), and DSP (b/c the miniDSP can accept the Realiser signal on an analog basis, digitize it and them DSP it before doing its own D-A conversion).

That's what I plan to do with my in house KEF LS 50/SB2000 system when I get a Realiser.

The question then would be how to keep the same output so you don’t have to keep disconnecting and reconnecting cables for each speaker.

 

[phoenixdogfan]

The question then would be how to keep the same output so you don’t have to keep disconnecting and reconnecting cables for each speaker.

I guess I'm missing something regarding the necessity to "change outputs" for the test signal. There has to be a way (even if it involves creating a distinct PRIR for each speaker angle) to use the same analog output. After all, what you are looking to capture is the IATD (Inter Aural Timing Difference) occasioned by having all these speakers in different positions in your listening room. Head rotation should do this as long as the angle correct angle can be ascertained (with Headtracking assist?) and you can contort yourself into that angle. The signal output by the A16 should be identical regardless of which analog output it comes out of, it's the way you hear it as a result of the differing comb filterings your head and body introduce depending on the angle you are hearing it from.

 

[Richter Di]

44.1khz works fine. I can’t use Tidal Masters as some albums end up in 88.2khz. But in Hifi mode they are all 44.1 or 48 kHz and those both work on the a16.

But this is via USB, right?

 

[Sordel]

I think the easiest thing to do is place one speaker in the center of the room, optimize that speaker with any ad hoc room treatments you can muster, and the rotate your head and body. That way, you'll get the best in room performance of the speaker.

That certainly works ... if you live in a round room. The illusion of listening to speakers works because the reverberation bounces precisely map onto the room you're sitting in. The paradox is that while you can indeed listen to better speakers in a better room through the A16, that benefit will always come at a cost in terms of the credibility of the illusion. So while you're measuring a system you are actually measuring the room and getting the speaker into a credible point in space is important. If you're doing the “revolving chair” method it make sense to do it near-field to avoid any room bounces, in which case you are effectively creating an idealised headphone array rather than a speaker environment.

 

[GeorgeA]

About how close it should be to the speaker? Might 60 - 80 cm distance be considered as near field?

 

[audiohobbit]

I guess I'm missing something regarding the necessity to "change outputs" for the test signal. There has to be a way (even if it involves creating a distinct PRIR for each speaker angle) to use the same analog output.

Please see the manual on p. 37ff and 62ff on creating a PRIR sound room.
You have to assign a speaker label (L, R, C,...) to a channel number. The test signal for that speaker is then output through that channel.
So if you do it in one PRIR, as Got the Shakes described, you have to reconnect the amp to the new output channel for every speaker. So you star by measuring the L spesker with the amp connected to analog out channel 1 on the A16, sit in the chair, measure your look angles, then press BACK, then you have to reconnect the amp to channel 2 (and I think also go to the speaker select page and unselect the L speaker and select the R speaker, Got the Shakes may correct me here), then sit back again on your chair, rotate 30 degrees and do your measurements for all your look angles (however you do that when rotated away from the center looking position) for the R speaker, then press back, reconnect to channel 3, and so on, up to 16 times...

If you do it with one PRIR per speaker, if that's possible, then you maybe only define a new PRIR sound room for every measurement where you assign every spesker label to chsnnel 1, if one can assign every speaker label to every channel! Since you can only define two PRIR sound rooms in parallel you have to go to that menu many times to reassign spesker label to channel 1 to create a new sound room for every speaker.

Theoretically you can do the latter method w.o. leaving the chair. But if you're rotated facing away from the Realiser you'll have difficulties looking on the display, and this is necessary for changing options because the web interface isn't implemented yet.

When creating the listening room later on, as far as I understood only the spesker lsbels are important, not the channel number with which it was recorded.

 

[audiohobbit]

If you're doing the “revolving chair” method it make sense to do it near-field to avoid any room bounces, in which case you are effectively creating an idealised headphone array rather than a speaker environment.

As long as you don't do this in a totally anechoic room, I'll bet that you can recognize that the speakers are very near while listening to that PRIR via Realiser!
I'd claim that even in an anechoic room you'd recognize the proximity of the sound source.

 

[arnaud]

As long as you don't do this in a totally anechoic room, I'll bet that you can recognize that the speakers are very near while listening to that PRIR via Realiser!
I'd claim that even in an anechoic room you'd recognize the proximity of the sound source.

But what if, for multi-channels soundtrack from movies at least, this was less distracting than an unrealistic reverb taken in much larger room than the listening space?

In a perfect world, to create an illusion of an object anywhere in space, these large channel count rendering systems like atmos and other auro systems should in principle prefer that the rendering is just the sum of a bunch of direct fields from many different headings, rather than some mess of a reverb combined to each speakers direct field.

arnaud

 

[Dixter]

Like really? I can not connect my CD player? That makes no sense for me.

the CD player should not be any problem... for music testing I plugged a fiio X5 into the A16 and it plays that fine..

 

[audiohobbit]

When my unit arrives I'll plug my Oppo 103 via HDMI to the A16 and will test CDs that way. I could check the SPDIF as well, if necessary.

 

[bcwang]

But this is via USB, right?

I was testing spdif. Haven’t tried usb yet since I haven’t installed the driver.

 

[bcwang]

I guess I'm missing something regarding the necessity to "change outputs" for the test signal. There has to be a way (even if it involves creating a distinct PRIR for each speaker angle) to use the same analog output. After all, what you are looking to capture is the IATD (Inter Aural Timing Difference) occasioned by having all these speakers in different positions in your listening room. Head rotation should do this as long as the angle correct angle can be ascertained (with Headtracking assist?) and you can contort yourself into that angle. The signal output by the A16 should be identical regardless of which analog output it comes out of, it's the way you hear it as a result of the differing comb filterings your head and body introduce depending on the angle you are hearing it from.

I guess there is a way to use the same output if you reconfigure the sound room between each measurement. That’s quite a few extra steps for each measurement but I guess it works.

What I ended up doing was doing just setting up one sound room with all the speakers. Then I did two separate measurements each time using the left and right speakers where they sat and moved to be directly in front of the one that would be active. Then I only had to move the cable on the back of the a16 to the next pair and choose the active speaker in the calibration menu. It made it decently quick to make a calibration but I stopped at only doing two overheads, could not figure out where the rest actually need to be and didn’t want them to interfere with each other if I got it wrong.

By the way, can someone confirm that when it does the look angles it always says center and then left and then right? I actually can’t hear the command with the mic in my ear and not sure if for some speakers it asks for a different rotation order or actually says look up or down.

One last thing I wonder, if the room is already built up of prir profiles and I delete the temp profile, do those room configurations continue working? Reason is this method uses up one profile for one channel. So 12 of the 16 recycle slots are already used up. This one speaker measuring is very bad for the memory storage.

 

[sander99]

The signal output by the A16 should be identical regardless of which analog output it comes out of

That is correct, the problem is that the current version of the user interface doesn't have a special mode yet for this case so we have to improvise with what is possible at the moment.

Someone who is so lucky to have a 16 channel mixer laying around (studio!) could simply mix all the 16 analog output channels together to one channel going to the amp/speaker.

 

[Got the Shakes]

I guess there is a way to use the same output if you reconfigure the sound room between each measurement. That’s quite a few extra steps for each measurement but I guess it works.

What I ended up doing was doing just setting up one sound room with all the speakers. Then I did two separate measurements each time using the left and right speakers where they sat and moved to be directly in front of the one that would be active. Then I only had to move the cable on the back of the a16 to the next pair and choose the active speaker in the calibration menu. It made it decently quick to make a calibration but I stopped at only doing two overheads, could not figure out where the rest actually need to be and didn’t want them to interfere with each other if I got it wrong.

By the way, can someone confirm that when it does the look angles it always says center and then left and then right? I actually can’t hear the command with the mic in my ear and not sure if for some speakers it asks for a different rotation order or actually says look up or down.

One last thing I wonder, if the room is already built up of prir profiles and I delete the temp profile, do those room configurations continue working? Reason is this method uses up one profile for one channel. So 12 of the 16 recycle slots are already used up. This one speaker measuring is very bad for the memory storage.

I never measured any height speakers, but I can tell you for my 5.1 setup I measured that the order was always “look center, look left, look right”.

Not sure about deleting the PRIR in the temp storage as I finally decided to measure my room as a singular PRIR and then saved it to the permanent storage.

 

[audiohobbit]

that when it does the look angles it always says center and then left and then right? I actually can’t hear the command with the mic in my ear and not sure if for some speakers it asks for a different rotation order or actually says look up or down.

Good question. Since you can define multiple look angles I was also wondering that. Up/down would only work if you activate it, IF it is already working for PRIR measurements. For listening vertical headtracking is not working at the moment, this is what they said, but what about creating PRIRs? The menu option is there already (Fig. 8-12) but no one seemed to try this so far?

Concerning command volume: You can boost this up:

8.1.4.5 Voice-Tone rel gain
Adjusts the volume of the voice prompts relative to the sine sw
eeps during the PRIR measurements.

Interestingly I can't remember such problems with my A8.
Did you do the level calibration before the PRIR measurement?