Smyth Research Realiser A16

[Richter Di]

Was it because the headphone set up let you hear more of the speakers and less of the room?

I prefered the headphone setup because of the better audio quality.

 

[jgazal]

So for me at least, even though the standards are such that you should have some reverb / avoid direct field from the surround channels in regular setup, I actually seem to prefer PRIRs taken in rather dry room (so I need the personalized xfeed but not so much the acoustic imprint of the room).

So to my idea: am wondering if it would make sense to get my own "PRIRs" (more like hrtfs) with the speaker rather close and stuffing the room walls to attenuate reflections as much as possible so that direct field dominates.

I'd prealably calibrate the speakers using the AV amp correction software so that response is reasonably flat at measured location.

Then, I'd "simply" rotate the head to get the various headings, including elevation channels. The angle might not be accurate but, as long as the head is steady during the recording, it would be fine perhaps.

So I am trying to figure the options to apply digital room correction during the PRIR measurements with only one studio monitor speaker.

I have considered:

A) A calibrated USB microphone such as minidsp UMIK-1 and a digital audio interface with spdif input and output such as minidsp USBstreamer B (toslink) or minidsp ministreamer (coaxial) with an standalone room correction software such as Dirac Live.

More details in this board: minidsp DDRC-22D with a single center speaker

B) A digital amplifier with room correction routine such as Paradigm PW- AMP;

C) I feel ashamed to ask more customization to Smyth Research since they already improved the A16 so much, but once generated in stand alone software, the room correction filter could be loaded into the own Realiser DSP and then applied directly to the frequency sweeps chirps digital or analog output (including the class d amplifier for direct passive studio monitor speaker connection).

Does anyone suggest alternatives?

Cheers!

 

[arnaud]

How about the ability for the realiser to compensate for the speaker response just like their HPEQ?
 
e.g. the actual definition of an HRTF is pressure measured at the ear canal with presence of the head / pressure measured at the head center location without its presence.
 
At the end of the one speaker PRIR test, we'd just need to place the microphones on a stand at the head center location, have the realizer perform one last measurement and then use that to compensate (slight) deviations from speaker response.
 
Given this is above 1kHz (hence a small speaker / tweeter is ok for this kind of measurement) and that direct field from the speaker should dominate, one expects the compensation to be trivial using similar technique to HPEQ filters.
 
This would even compensate for actual microphones used for the recording. In particular, Smyth mentioned one cost-cutting measure for the A16 was to find cheaper mic capsules but we all know that good sensors just are expensive. In the end, I do not know if the mics are simply paired (phase/gain matched) or they also ship with a specific A16 which then contains internal compensation filters for the microphones (compensation for mag / phase mismatch between the capsules which is a given for cheap / tiny piezo capsules).
 
But having the ability to have the A16 automatically compensate for capsules / speaker imperfections for the user who wants to use the A16 as a realtime HRTF convolver, would be just ideal for me (and probably many who share the same issue I have with non-natural reverberation tail in PRIR measurements).
 
cheers,
arnaud

 

[jgazal]

How about the ability for the realiser to compensate for the speaker response just like their HPEQ?

e.g. the actual definition of an HRTF is pressure measured at the ear canal with presence of the head / pressure measured at the head center location without its presence.

At the end of the one speaker PRIR test, we'd just need to place the microphones on a stand at the head center location, have the realizer perform one last measurement and then use that to compensate (slight) deviations from speaker response.

Given this is above 1kHz (hence a small speaker / tweeter is ok for this kind of measurement) and that direct field from the speaker should dominate, one expects the compensation to be trivial using similar technique to HPEQ filters.

I have been reading that room correction software acts in the frequency and time domain.

While I guess that once you alter one you necessarily affect the other, I am not sure if some room aberrations can be measured and corrected using a filter based solely in frequency domain..

I have found this article from Gerzon very interesting: http://www.audiosignal.co.uk/Resources/Digital_room_equalisation_A4.pdf.

I am always admired when I read anything from Gerzon.

This would even compensate for actual microphones used for the recording. In particular, Smyth mentioned one cost-cutting measure for the A16 was to find cheaper mic capsules but we all know that good sensors just are expensive. In the end, I do not know if the mics are simply paired (phase/gain matched) or they also ship with a specific A16 which then contains internal compensation filters for the microphones (compensation for mag / phase mismatch between the capsules which is a given for cheap / tiny piezo capsules).

But having the ability to have the A16 automatically compensate for capsules / speaker imperfections for the user who wants to use the A16 as a realtime HRTF convolver, would be just ideal for me (and probably many who share the same issue I have with non-natural reverberation tail in PRIR measurements).

cheers,
arnaud

This is what the manual says:

Microphone balance calibration

The microphones supplied with the Realiser are matched by hand at the factory within a fraction of a decibel. Any slight remaining difference in sensitivity will become part of the HPEQ measurement, so listening to a PRIR and HPEQ made with the same microphone pair will be fully compensated.

The Realiser provides a microphone balance procedure in order for microphone imbalance to be compensated independent of HPEQ, so that any PRIR can be used with any HPEQ regardless of the microphones used. Because the microphones in each pair have already been carefully matched, this procedure may be superfluous, but it provides further assurance that any residual imbalance, effect of microphone aging or mishandling, etc. is accounted for.

Place the microphones close together and stationary. Diffuse ambient sound is best for this measurement. Only after the microphones have been placed, press MENU-CAL. The screen will say:

MIC CALIBRATION
BALANCE: 1.0 (L/R)
PRESS EXIT TO ABOUT
...OK TO REBALANCE

While this menu is on screen, the Realiser is accumulating the average signal level received at each microphone. When OK is pressed, the balancing factor is calculated and the accumulators are reset. After perhaps fifteen seconds, press OK and observe the number. Do this several times and you should see similar numbers each time.

After several brief observations, let the measurement accumulate for several minutes and press OK. This final, longer-accumulated factor will be the one to use. If during the brief trials you get very different numbers each time, check the cabling and try to avoid loud transient sounds during the measurement.

The range of factors is 0.5 to 2.0, with 1.0 indicating that no compensation is necessary. Intermediate readings will be compensated, but a reading of 0.5 indicates 0.5 or lower and a reading of 2.0 indicates 2.0 or higher, so if either of these numbers appears, it should be considered that there is a problem with the microphones.

The last-calculated factor will be applied for new measurements until power-down; to save it for future sessions, press (!)-SAVE SYSTEM CONFIG.

http://www.smyth-research.com/downloads/A8manual.pdf

I am not sure if such level matching is as precise as a frequency measurement file from a given capsule loaded into a dsp to compensate the frequency domain variations.

Anyway, you know more about this subject than I do.

Cheers.

 

[RRod]

I have been reading that room correction software acts in the frequency and time domain.

While I guess that once you alter one you necessarily affect the other, I am not sure if some room aberrations can be measured and corrected using a filter based solely in frequency domain..
 

 
The two domains are dual, but the frequency domain involves both frequency response and phase. If you're using a minimum phase filter (like many EQs), then the phase effects of your filter are determined by the frequency response, and thus you can't manipulate non-minimum phase behavior (say in a room) by such a filter. See here for more.

 

[jgazal]

How about the ability for the realiser to compensate for (...)

Thank you, Arnaud! I believe you are right.

12. My AV receiver supports Audyssey room equalisation and I intend to personalise my room with the EQ in circuit. Are there any problems with this?
Not at all. We have used the Realiser in conjunction with both Audyssey and Trinnov room equalization hardware to great effect.

http://www.smyth-research.com/faq.html

Q. I would like to know if, in your experience/opinion, there are any advantages to using an external digital room correction (DRC) algorithm while measuring a PRIR, mainly for users with little to no experience with acoustic settings. Would the existing A16 processing facilities (such as bass management/direct bass, reverberation and equalization user settings) allow to achieve similar - or hopefully identical - acoustical quality that one would achieve when using the first DRC/PRIR method?

A. We have some experience using our previous Realiser A8 with both Trinnov and Audyssey room correction systems. Essentially when the A8 was connected upstream of these DRC systems a PRIR measurement would capture the equalised sound room to good effect. So in general DRC is a good thing. The A16 will incorporate both room amplitude and reverberation equalisation systems so the need to attach an external DRC will not be so pressing. In my opinion our hybrid equalisation method will not just match but exceed anything achievable with DRC alone.

https://www.kickstarter.com/projects/1959366850/realiser-a16-real-3d-audio-headphone-processor/comments

 

[AppleheadMay]

Just preordered the A16, going for the rack mount version.
Will see how it fares with the standard Atmos profile and try to find a place to calibrate it for Atmos, DTS:X and Auro not too far in Europe.
 
I have some questions though and would appreciate your opinions.
 
As far as headphones go I hear the HD800S is a great choice due to it's spatiality.
I wonder how a TH-900 would do as it's one of the most spatial closed phones. Or are closed phones a bad choice?
What about the Utopia or Pioneer M1? Any others worth mentioning?
I do not want to go for Stax.
 
How is the amp part in the Realiser by the way? Would it be a good idea to hook up my Black Widow?

 

[462429]

I have Sennheiser HD 598SE and don't plan to upgrade soon. As I understand the Realiser philosophy, headphones don't have to be the "best of the best", and their amp too. Everything here is magically processed and if that magical processing can bring a million dollar audio system into "your bathroom", it can certainly bring your decent headphones to the high enough level too.
 
What I'm most concerned about is - will I be able to listen to my multichannel flacs through the Realiser A16/headphones? If flac is really not compatible, is there some indirect way to make it work? Does anyone have any idea or positive experience with the A8?

 

[bmoura]

  I have Sennheiser HD 598SE and don't plan to upgrade soon. As I understand the Realiser philosophy, headphones don't have to be the "best of the best", and their amp too. Everything here is magically processed and if that magical processing can bring a million dollar audio system into "your bathroom", it can certainly bring your decent headphones to the high enough level too.
 
What I'm most concerned about is - will I be able to listen to my multichannel flacs through the Realiser A16/headphones? If flac is really not compatible, is there some indirect way to make it work? Does anyone have any idea or positive experience with the A8?

Actually Smyth will tell you that the better the headphone is, especially it's frequency response flatness, the better your results will be with the Realiser.
That's why they bundled Stax headphones with the original Realiser A8 product.

 

[462429]

 
Actually Smyth will tell you that the better the headphone is, especially it's frequency response flatness, the better your results will be with the Realiser.
That's why they bundled Stax headphones with the original Realiser A8 product.

 
Yes, yes, but even flatness is being improved here ... In the same virtual way   

 

[dsperber]

What I'm most concerned about is - will I be able to listen to my multichannel flacs through the Realiser A16/headphones? If flac is really not compatible, is there some indirect way to make it work? Does anyone have any idea or positive experience with the A8?

 
The A8/A16 are intended to replicate the sound of real loudspeakers and room/electronics as they are heard by your brain and ears just the way things sounded when you actually sat there during the PRIR measurement process, except played back through stereo headphones where the digital SVS processing in the Realiser causes you to believe you're hearing that very same original room loudspeakers and total listening environment (good or bad).  That's what it's for, to duplicate a given listening environment as your own ears heard the calibrations sounds in that room as recorded in the PRIR, and as played back through your particular headphone/amp you use as measured through the HPEQ.
 
The A8 was fed already upstream-decoded discrete multi-channel input, either analog (via 8 RCA inputs) or digital (via LPCM through HDMI). The A8 had no built-in decoding capability. It was up to the upstream processor(s) (e.g. Oppo AVR) to decode whatever the multi-channel source was and feed the now decoded discrete multi-channel output to the A8. No matter whether the multi-channel source is BluRay movies, HDTV, multi-channel audio, 2-channel anything, doesn't matter.  Whatever you feed to the AVR, it will be decoded and then each of the discrete channels fed to the A8 either through 7.1 analog (e.g. preamp line outputs) or 7.1 digital (LPCM via HDMI).
 
You can still do that wih the A16... i.e. feed upstream decoded discrete multi-channel audio, either via analog or digital (albeit now up to 16 channels instead of just 8).  So even if the new built-in decoding capability for some reason doesn't support multi-channel FLAC (and I don't know if it does or doesn't), you can always still just decode/play it through your upstream AVR and feed the now decoded discrete multi-channel audio to the A16 through either analog or HDMI, same as it was done with the A8 and which is still possible with the A16.
 
You are not obligated to use the decoding capability of the A16 if you don't need to or don't want to.  But if your wiring schematic is based around using the A16 to be the decoder for still-encoded source multi-channel content, you'll obviously be limited to whatever the A16 can decode. This is no different than whatever the AVR limitations might be for decoding multi-channel source, such that you couldn't handle Atmos source content and feed the A16 using an older AVR that didn't understand Atmos.

 

[esimms86]

I have an exasound e28 multichannel DAC but I no longer have a surround sound system in my house.  My plan is to have jriver and the e28 process my multichannel FLAC/DSD/iso files and then send them via analog connections to the A16. I checked with Lorn over a year ago and he assured me that the A8 can used in this way so I'm sure that the A16 will work in similar fashion. I have a small but growing collection of multichannel SACD rips originally from Audio Fidelity remasters and also nativedsd download purchases that I look forward to listening to via the A16. I also hope someday to go to Timonium, Maryland(location of an authorized A8 seller)and/or the AIX Studio in Los Angeles to have 2.1 and 5.1 PRIRs made. Once I do this I will also likely use one or more PRIRs to have a personalized OOYH preset made my Darin Fong. I have an original Blue Hawaii plus a STAX SR009 to complete my setup.

 

[462429]

...
The A8 was fed already upstream-decoded discrete multi-channel input, either analog (via 8 RCA inputs) or digital (via LPCM through HDMI). The A8 had no built-in decoding capability. It was up to the upstream processor(s) (e.g. Oppo AVR) to decode whatever the multi-channel source was and feed the now decoded discrete multi-channel output to the A8. No matter whether the multi-channel source is BluRay movies, HDTV, multi-channel audio, 2-channel anything, doesn't matter.  Whatever you feed to the AVR, it will be decoded and then each of the discrete channels fed to the A8 either through 7.1 analog (e.g. preamp line outputs) or 7.1 digital (LPCM via HDMI).
 
You can still do that wih the A16... i.e. feed upstream decoded discrete multi-channel audio, either via analog or digital (albeit now up to 16 channels instead of just 8).  So even if the new built-in decoding capability for some reason doesn't support multi-channel FLAC (and I don't know if it does or doesn't), you can always still just decode/play it through your upstream AVR and feed the now decoded discrete multi-channel audio to the A16 through either analog or HDMI, same as it was done with the A8 and which is still possible with the A16.
...

Thank you, but there is a problem if your Oppo's analog outputs are occupied being conected to the amp and it's "HDMI Audio" setting is set to "Off" because you don't want your "SACD Output" setting (which is set to "DSD") to be automatically turned into "PCM", without your permission.
It seems I will not be able to avoid constant switching - either cables or "HDMI Audio" setting

 

[462429]

  ...
My plan is to have jriver and the e28 process my multichannel FLAC/DSD/iso files and then send them via analog connections to the A16. I checked with Lorn over a year ago and he assured me that the A8 can used in this way so I'm sure that the A16 will work in similar fashion. I have a small but growing collection of multichannel SACD rips originally from Audio Fidelity remasters and also nativedsd download purchases that I look forward to listening to via the A16.
...

I don't know if you can have multichannel FLAC and DSD via PC's "digital audio" because they are usually HD MCH and as far as I know HD MCH demands either analog or HDMI.

 

[esimms86]

My files already exist as FLAC or DSD. They are sent to jriver set up for multichannel and then fed to a multichannel DAC. The music is then fed to the A16 via analog inputs. If I was playing SACDs, DVDAs or Blu Ray discs then I would most definitely be using an HDMI input.
 
The exasound website has an excellent primer on how to playback multichannel files via jriver.