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Sound engineer's monitoring and HiFi + HT system with active Yamaha NS-1000x, multi-subs and FIR QSys processor in a particular and treated room

jlo

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I (we) need very careful look on it for understanding what's going on and what you can do with the QSys configurations! I fell it would be almost impossible for me to fully understand it by the single diagram you shared, though; I will further try to "read" the diagram.
The diagram looks a bit complicated because it has been designed to listen and compare many possibilities :
- without/with FIR/IIR EQ for front/subs/surrounds
- compare various EQ
- switch between music and movie (X-curve) together with LFE level at +10dB
- compare 1/2/3/4 subs
- compare real central loudspeaker with phantom center
- listen independantly each channel or speaker
- add a tilt curve, inspired by Quad preamps
-....
 

jlo

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It would be really amazing that you do it in full/complete clock synchronization. You definitely have no micro drift of synchronization (i.e. time dependent micro asynchronization) all over the so many channels SP drivers, right? Please let me also ask you about how you could establish and objectively measured/confirmed the complete time alignment at your listening position (hopefully in 0.1 msec precision over 20 Hz to 22.05 kHz) for all over the multiple SP drivers...
Synchonisation clock : we use the QSys internal clock. It is the same for all DSP channels and I see not reason why there would be any drift between channels.

The time alignement has been measured at listener place and aligned for all channels and speakers at 0.1ms (maybe 0.2ms for subwoofers as it is more difficult to exactly measure with low frequency signals). The time alignement is automatically adjusted when you switch FIR on/off or other type of comparisons.
 

dualazmak

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Synchonisation clock : we use the QSys internal clock. It is the same for all DSP channels and I see not reason why there would be any drift between channels.

The time alignement has been measured at listener place and aligned for all channels and speakers at 0.1ms (maybe 0.2ms for subwoofers as it is more difficult to exactly measure with low frequency signals). The time alignement is automatically adjusted when you switch FIR on/off or other type of comparisons.

Thank you for your quick and kind responses.

Yes, I assumed that the common and precision internal single clock in QSys can be referred from all the channels/procedures ensuring complete synchronization. Thank you for your confirmation.

I now understand your time alignment establishment (and objective measurement/validation?) in your such a big complicated system. Just for your interest and reference, I did it manually using my rather naive and primitive approach/methods as summarized in my post here.

(In case if you and/or any of ASR friends would be interested, I will be happy to share the test tone signals I prepared for these naive primitive measurements. )

In any way, I am very happy to know we have our common goal(s) and approaches for excellent total sound quality at our listening position using still-amazingly-wonderful NS-1000 and NS-1000x!
 

jlo

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I now understand your time alignment establishment (and objective measurement/validation?) in your such a big complicated system. Just for your interest and reference, I did it manually using my rather naive and primitive approach/methods as summarized in my post here.

I had a look to your interesting methods.
I'm using a signal similar to your single sines : it is 5 single sine burst with the middle of each sine separated to the next by a known number of samples.
This gives following graphs.

timealign.jpg


More explanations can be found in https://www.loudspeakers.audio/en/graphs-explained/
 

dualazmak

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I had a look to your interesting methods.
I'm using a signal similar to your single sines : it is 5 single sine burst with the middle of each sine separated to the next by a known number of samples.
This gives following graphs.

View attachment 263154

More explanations can be found in https://www.loudspeakers.audio/en/graphs-explained/

Thank you again for your quick response.

Yes, your method above and one of my approaches look very much similar with each other based on common principle;
- Precision measurement and adjustment of time alignment for speaker (SP) units: Part-2_ Energy peak matching method: #494

At least for me, I feel my another approach of "time shift method using super-tweeter tone as time zero marker" would be also (better?) feasible and accurate since I can/could easily precisely identify the kick-up timing of each of the SP drivers;
- Precision measurement and adjustment of time alignment for speaker (SP) units: Part-1_ Precision pulse wave matching method: #493
- Precision measurement and adjustment of time alignment for speaker (SP) units: Part-3_ Precision single sine wave matching method in 0.1 msec accuracy: #504, #507

I am very happy having this invaluable discussion on "time alignment methods" in our multichannel multi-amplifier audio systems with you two (you and Igor); it is really amazing we are still fully utilizing Yamaha NS-1000 and NS-1000x!
 
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dualazmak

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Furthermore, for the best fine tuning of sub-to-woofer and woofer-to-mid time alignment around XO Fq, the "wave shape matching" using single-wave-tone-burst would be very much useful not only for the smooth transition but also for selecting the best XO Fq (especially sub-to-woofer) in excellent transient characteristics by looking at Adobe Audition's 3D sound color spectrum, as shared in my post #504 and #507 on my project thread.

I could also successfully measure and assess the transient characteristics of sub-woofer and woofer using 8-wave tone burst and 3-wave tone burst;
- Measurement of transient characteristics of Yamaha 30 cm woofer JA-3058 in sealed cabinet and Yamaha active sub-woofer YST-SW1000: #495, #497, #503, #507

I wrote there:
I was really surprised that the kick-up and fade-out patterns are still really nice in 1 kHz, with both of the 8-wave and 3-wave excitation.
I am very happy seeing and confirming that my 40-year old Yamaha JA-8058 30 cm woofer (WO) driven
(directly and dedicatedly without LC-network) by A-S3000 amplifier is still working very fine and doing really nice job throughout the 55 Hz to 550 Hz Fq region in which I always use it all the way through my multichannel audio project.
 
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tehas

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@jlo and @Igor Kirkwood what is the delay that the QSC adds in the system? I mean just the QSC and an eq without FIRs. Do you have any video-audio sync issues when playing back video content? I'm asking to get a better understanding of how much latency different products add to the system, because I have no experience on this.
 

dualazmak

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The delay is not only a matt of hardware. The choices made at the level of the filter banks will trigger different delays wether you are using LR24 our à FIR 4096 taps. So you have to refine your delays once you have frozen your filtering strategy,

dominique

And, the relative delay between all the SP drivers as well as between L-SPs and R-SPs, front SPs and rear/side SPs, should be objectively measured by independent method using measurement microphone at the listening position.

I wrote here;
In the digital signal processing, we have so many buffers or latencies; JRiver output buffer, ASIO4ALL's I/O buffers, EKIO's processing buffer, DIYINHK USB ASIO driver's buffer, and so on. Consequently, it is not straightforward to exactly measure the "absolute delay" between the JRiver's "shout" and the final air sound kick-up by SP.

I usually set all the buffers in the digital domain in rather large size, so that I should not have any latency or delay problems; in our audio setup, we have no problem at all if all the bunch of the digital and analog signal (15Hz - 30 kHz) have identical amount of delay time from the signal origin at JRiver, and this is always the case in our digital (PC based) audio system.

The relative delay between the SP units, or "time alignment" in multiple SPs, however, is always one of the critical issues in audio system, especially the multichannel multi-driver multi-amplifier system, as you may agree.

It is important and indispensable that we need to fully "validate" the method; the "forced delay settings in 0.1 msec precision" should be accurately "measured" by the method we would apply.
 
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Igor Kirkwood

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And while it looks like good "not-super-expensive-from-hiend-POV" scheme, it have at least one flaw: most probably, one will need a specialist qualified like Jean-Luc Ohl to achieve such results...
Finding a "Jean Luc Ohl" in the USA is not as difficult as you think.
Jean-Luc Ohl's website "Loudspeaker audio" allows you, with Jean-Luc Ohl's MMM measurement method, to obtain the same curves and measurements as those published on ASR and many others (like this measurement curve impulsive with Step response. Yes You can get it !



Example Step response Yamaha NS 1000x without FIR correction and no subs

imp no fir no sub.PNG




impddme 141j carrés.PNG
 
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FeddyLost

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Finding a "Jean Luc Ohl" in the USA is not as difficult as you think.
Anyway, for such system one will need some experienced professional working with it until success.
Even if end-user can measure something, building such system with a lot of variables is not an easy task.
 

jlo

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@jlo and @Igor Kirkwood what is the delay that the QSC adds in the system? I mean just the QSC and an eq without FIRs. Do you have any video-audio sync issues when playing back video content? I'm asking to get a better understanding of how much latency different products add to the system, because I have no experience on this.
In our case, from analog input to analog output and without FIR, the delay is near 3.17ms.
With our 8192 taps FIR and uncentered FIR impulse, the total delay is 45ms.
 
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tehas

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thank you all for the information and perspectives, it helps me understand :)
 
OP
Igor Kirkwood

Igor Kirkwood

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A decisive advantage for the QSC 110f interface (designed by Jean-Luc Ohl) is the possibility of carrying out true instantaneous blind tests between several configurations.
Examples: comparison of 6 FIR equalization target curves, of several configurations of the 4 subs; and especially 2 target curves of the installation.
And this thanks to the collaboration with Pos (inventor of the RePhase software and well known on ASR).
Attached is a photo of the switch made by Pos.
Switching speed of listening + blind listening is critical for a Sound Engineer (ask Tuga ).
The human brain having a weak auditory memory but a colossal and extravagant aptitude to invent the imaginary auditory differences often read in the forums...
poss3.jpg
MUSICe.PNG
 

gnarly

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Hi guys, another Q-Sys user here....
Love what you've done!

I think the ease of of comparing different speaker setups and/or filter sets instantly, without an audible or hitch of glitch, is one of Q-Sys's super attributes.
I don't know of anything else that can switch custom FIR files on the fly.
I think having high physical input and output counts, without needing a large soundard (for use with a computer) is another.
Along with being able to build GUI remotes that can control about anything...
And most certainly the open architecture.....

Anyway, Q-Sys has added so much enjoyment to my audio world...
Got a Core110f four or five years ago, and now mainly use a 510i.
Here's a schematic for a 3 speaker LCR setup. 4-way mains, each set on a sub stack. So essentially, three 5-ways.
15 channels of FIR, 16,364 taps per channel....showing 43% of capacity. Dang 510i is a monster !!!
syn11 qsys schematic snip 12-14-22.JPG
 
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Igor Kirkwood

Igor Kirkwood

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Thank you gnarly for your important contribution via your ultra powerful QSC.
You rightly insist on one of the essential qualities of the QSC, is the speed of switching between several settings which allows the human ear to really hear the auditory differences between the devices and the settings. You rightly estimated that the "disadvantage" of the QSC: its no lower signal-to-noise ratio compared to other choices did not weigh anything compared to its flexibility and its speed. My Studio has an internal noise floor of 15.5dB (measured with NTI XL2 and class 1 microphone) and yet no QSC floor noise at the 2.5m listening position, with a Marantz 7703 preamplifier. Could you gnarly post a photo of your setup here.
Also note that the POS box which allows blind listening is very easy to install on your QSC.
NOISE briare studio.PNG
 
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gnarly

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Hi Igor, what a nice quiet room you have !

My situation/setup is much more of a laboratory .....I'm constantly trying/comparing DIY stuff, so Qsys is simply ideal.
For instance, the white center speaker is a new build I'm comparing to my current system that uses 3 (blue) synergy/sub stack in LCR mode.
LCR syn11 setup.jpg

Here's the GUI I use for comparisons.
Presets for stereo; mono L, R,or C; L&R dual mono, and several different LCR matrixing methods.
(The volume sliders are not EQs...they actually control the output levels to each of the driver sections....very helpful for testing..not to mention adjusting to taste)
LCR remote.JPG

What is the POS box you mention? I heard QSC made a box for amp comparisons, back when to help sell their amps. Is that what you have?
 

jlo

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What is the POS box you mention? I heard QSC made a box for amp comparisons, back when to help sell their amps. Is that what you have?
POS did a simple toggle switch connected to QSC GPI input to change presets or other parameters manually.
The QSC box is very different : it is an ABX switchbox with remote and display with statistics of right/wrong responses.
Not a lot were built by QSC but by chance, I have one ;)
I have used it to check if QSys input to output was transparent.

I was thinking to program an ABX switch inside QSys but never found enough time....
 
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Igor Kirkwood

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Manual:

It requires the Pos switch to be square in shape with the switch button balanced.
After comparing A and B several times, then you can proceed to
blind test, just... close your eyes, turning the switch several times so as to lose the A and B marks. Then listen to yourself by changing the position of the switch button (position A or B).

If the true A matches the blind guessed A, you win!
It's simple and efficient.

Examples: 2 target curves or 2 sub configurations............

With Pos box It's a blind test.
I wait that Jean-Luc Ohl write a real ABX test for the QSC 110f
 

gnarly

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POS did a simple toggle switch connected to QSC GPI input to change presets or other parameters manually.
The QSC box is very different : it is an ABX switchbox with remote and display with statistics of right/wrong responses.
Not a lot were built by QSC but by chance, I have one ;)
I have used it to check if QSys input to output was transparent.

I was thinking to program an ABX switch inside QSys but never found enough time....
Gotcha, thx. I think can see how to make a GPI switch box now.

Good for you, having the QSC ABX switchbox. Does it have a model number? I might keep a lookout for one out in the wild...
Manual:

It requires the Pos switch to be square in shape with the switch button balanced.
After comparing A and B several times, then you can proceed to
blind test, just... close your eyes, turning the switch several times so as to lose the A and B marks. Then listen to yourself by changing the position of the switch button (position A or B).

If the true A matches the blind guessed A, you win!
It's simple and efficient.

Examples: 2 target curves or 2 sub configurations............

With Pos box It's a blind test.
I wait that Jean-Luc Ohl write a real ABX test for the QSC 110f
And thx to you too.
So far, I haven't really concerned myself with blind test verification....because if in my mind it's gotten close enough to need such a test...there's automatically no difference :D
But that's just me. What i really like, is to live with any new setup for a few days at least, listening from the designed position, and from other rooms as well.
 
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