[SobiC]

In synchronous mode the Realiser starts to play the sweep and expects the recorded sweep ideally to start at the same time.

Yes, but then you ALWAYS end up with a lot of unwanted system delay (input lag + speaker distance + speaker delay etc.). I hope (but don't know) the Realiser has some kind of impulse response shifting function. But maybe it doesn't and that's the reason why some users here report about extensive delays...

 

[romainhc]

It works flawlessly

Another addition to the wish-list would the possibility to save listening rooms and presets to SD card

It seems like a priority for me, when you see how much options and settings and time it takes to have a great listening room / preset.

 

[romainhc]

[OFF]How do u delete a post ? lol[/OFF]

 

[audiohobbit]

impulse response shifting function. But maybe it doesn't

I don't think it has and it shouldn't. A real loudspeaker in a big auditorium 10 m away from your seating position would have delay of about 30 ms. Of couse for movies this has to be taken into account, but should not be done in the impulse response. This would be my opinion. Of course I could be wrong.

that's the reason why some users here report about extensive delays...

Yes I also thought about this and maybe you're right.
Would be interesting what PRIRs these users use, since I only use the factory PRIR of the Surrey room.

 

[Eich1eeF]

I don't think it has and it shouldn't. A real loudspeaker in a big auditorium 10 m away from your seating position would have delay of about 30 ms. Of couse for movies this has to be taken into account, but should not be done in the impulse response. This would be my opinion. Of course I could be wrong.

I'd say you are wrong. You could argue that it should not have automatic delay removal, but if the AVR used to to record the PRIR has some significant delay (maybe because it's setup with an appropriate delay for an especially slow display), that should most certainly be removed. At the very least, the lowest delay of all speakers in a room could and should be removed from all speakers automatically. Right now, I can't think of any reason why _any_ delay should be retained, since I don't see how that could possibly improve localization.

 

[Richter Di]

If I find some time I may write a tutorial for this, because I think it's important.

Yes, please, or a video?

 

[Erik Garci]

Did you just connect the line out to the mic in, without any adapters (impedance, voltage)?? That's brave...
The mic in would need a very low voltage. So the level of the line out has to be set very low, which deteriorates S/N ratio.

I found that the A8 provides +5V DC power on the tip for the microphones. I used a TRRS to 3 RCA adapter, where the RCA for the tip is not connected to anything. The A8 automatically calibrates the levels when you press CAL-OK.

 

[Got the Shakes]

I ran through the manLOUD procedure in the video from the Smyths from a month back and it did make the Surrey Atmos setup have better separation for sure.

 

[bcwang]

I don't think it has and it shouldn't. A real loudspeaker in a big auditorium 10 m away from your seating position would have delay of about 30 ms. Of couse for movies this has to be taken into account, but should not be done in the impulse response. This would be my opinion. Of course I could be wrong.

I think the impulse response must be shifted, that is the only way for async to work as it's impossible to have the timing right by hand. It would also be the only way to combine PRIR profiles together from different systems that have different system delays. It would also be the reason you can measure in one system which has variable amount of delay, and playback in another system and have it perfect. I measured my PRIR on my dedicated stereo system with analog output (no delay) and play back in my TV room through all the HDMI stuff, etc..In fact, I'd say it must be auto shifted because there are no controls to change channel delay, meaning the A16 already makes sure every channel arrives at your ear at the exact same time relative to other channels.

In fact, I always believed that this capability is one of the big advantages of the A16 over an actual speaker system, that it can equalize all delays and volumes for all 16 channels. So each listener can have perfect delay, volume, and soundstage like they are in the prime seat of a surround system. In a speaker system, especially a small home cinema, everyone other than the prime seat will be closer to some and farther from other speakers thus getting varying volumes and delays compared to the prime seat.

 

[bcwang]

I ran through the manLOUD procedure in the video from the Smyths from a month back and it did make the Surrey Atmos setup have better separation for sure.

It's too bad the HPEQ can't be applied to specific speakers while a different HPEQ applies to other speakers. I'd like to improve my Surrey room overheads I used with my direct measured 7 speakers but the only way for that to work is to keep the 7 speakers on my autoEQ profile while having the manLoud for the surrey room overheads. It would be a cool feature if the mixing and matching could be done.

 

[jgazal]

@jgazal ("cc" @Erik Garci): so the user A and B headtracking can be linked, that's one step in the right direction for my dual user cross talk reduction/cancellation idea. (Now we still need individual speaker level control per user. Or try to create dummy silent speakers or find some other trick.)

I am going to update the firmware this weekend and report.

 

[sander99]

Warning: I am going into full-nerd-mode again for a moment:

It's too bad the HPEQ can't be applied to specific speakers while a different HPEQ applies to other speakers. I'd like to improve my Surrey room overheads I used with my direct measured 7 speakers but the only way for that to work is to keep the 7 speakers on my autoEQ profile while having the manLoud for the surrey room overheads. It would be a cool feature if the mixing and matching could be done.

In fact this additional "corrective" filtering should ideally not be in the HPEQ, but in the PRIR. Then it could be different for each speaker, and even more: different for each individual impulse response (that is for each speaker-ear-lookangle combination).

It's too bad that not any speaker/channel can be choosen to perform the manloud procedure with.
If we were able to first determine per measured lookangle for each speaker which additional filtering it needs then with a trick we could create a new PRIR in which on each individual speaker for each measured lookangle the corresponding additional filtering is applied. (So it is applied in the PRIR instead of the HPEQ).
As follows:
First play the test sweeps (non-overlapping) through the A16 using the original PRIR and a neutral HPEQ and record the headphones output (while holding the head tracker in the appropiate angles for the different lookangles). The result would be a series of sweep responses, for all speakers and lookangles.
Next we need to do some "audio editing" on this recording with a pc.
First identify in the recording the sweep response for each speaker/lookangle combination (and of course for the left ear if is in the left channel, and for the right ear it is in the right channel). Then for each of those sweep responses apply the apropiate additional filtering to that segment of the recording (using some audio editing software).
Finally during an asynchronous PRIR measurement feed the altered recording with sweep responses into the mic inputs of the A16. (Or ideally: if Smyth implements the PRIR generation function that takes its input from an SD card instead of from the mic inputs then use that method.)
The result is an altered copy of the original PRIR with the additional filtering applied to the individual speakers/lookangles combinations.

This technique of
1. reproducing the underlying sweep responses of a PRIR
2. altering the sweep responses
3. generating a new PRIR from these altered sweep responses
can be used for many things by the way. For example for creating cross talk free or limited cross talk PRIRs from existing PRIRs.
Also it can be used to "add" PRIRs together (by adding the underlying sweep responses), which also has several applications.
For example this can be used to create virtual coaxial speakers by first measuring PRIRs from individual drivers positioned in the same spot.
Say you have a conventional - non-coaxial - 2 way speaker with bi-amping terminals. You could first measure a PRIR with only the tweeter branch connected to the amplifier. Next move the speaker up such that the acoustic centre of the woofer occupies the exact same spot where the acoustic centre of the tweeter was before. Now measure another PRIR with only the woofer branch connected to the amplifier. Add the 2 PRIRs together and you have a virtual coaxial speaker.
You can also create virtual bipole and dipole speakers.
For example combine bipole/dipole with coaxial:
Repeat the steps for creating a coaxial speaker, but with the speaker facing in the opposite direction (but with the acoustic centre in the same position as before).
Add the two resulting "coaxial" PRIRs together and you have a virtual coaxial bipole speaker.
Add the two resulting "coaxial" PRIRs together but with the backfiring one inverted (times -1) and you have a virtual coaxial dipole speaker.
(One practical complication: to make this work your head and ears should also be in the exact same positions for each of the PRIR measurements.)

 

[Flash676]

Has anyone with both a Realiser and a home theater receiver/processor had any success placing the Realiser after the receiver? If so, what is the brand of the receiver?

I've tried the following setup, but it seems my Pioneer receiver's HDMI "passthrough" is only providing two channel audio to the Realiser.
~10 sources -> receiver -> Realiser -> TV

 

[dsperber]

Has anyone with both a Realiser and a home theater receiver/processor had any success placing the Realiser after the receiver? If so, what is the brand of the receiver?

I've tried the following setup, but it seems my Pioneer receiver's HDMI "passthrough" is only providing two channel audio to the Realiser.
~10 sources -> receiver -> Realiser -> TV

My guess here is that the HDMI negotiation was between the SOURCE and the TV, deciding that 2-channel stereo is all that the TV can accept. Hence your result. It's not the A16 which is the destination, but rather the TV.

 

[Flash676]

My guess here is that the HDMI negotiation was between the SOURCE and the TV, deciding that 2-channel stereo is all that the TV can accept. Hence your result. It's not the A16 which is the destination, but rather the TV.

Thanks. I tried this again without the TV connected but I'm still only getting the receiver to pass through two channel audio.