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Multi-Channel, Multi-Amplifier Audio System Using Software Crossover and Multichannel-DAC

Igor Kirkwood

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"IIR induces post ringing (less audable as the sound masks the ringing) FIR minimum causes preringing (ringing sound comes before the actual sound event- more audable), FIR linear phase (perfect). As far as i'm aware, that's basically it. "

Minimum-phase IIR or FIR will have the exact same time response (ie only post ringing).
Linear-phase FIR will cause preringing if a filter is not matched with a complementary filter t form a proper crossover.

@dualazmak, I did not read all the posts here, but what matters it to consider the acoustical response, that is the "natural" response of the driver in addition to the electrical filter.
Getting an acoustical LR filter is typically not simply a matter of using electrical LR filter. To be clear, using electrical LR filters is very unlikely to produce acoustical LR filters. You need to measure what you get, and adjust using EQ and filters to obtain your acoustical target. Simulation software can be used for this.
When FIR can be used then it become easier as targeting a linear phase response is much easier than trying to match phase shifts between drivers, especially with a 3+ way system.

Hello pos :)

If I don't mistake you are pos the creator of Rephase software.
We are OK with Jean-Luc Ohl : It's possible to avoid any pre ringing with FIR linear EQ if matching a complementary filter.

So the dualazmak argument against FIR EQ fall completly.

So is possible to built FIR crossover more than 48 dB/octave without phase error.

It sound better, and especially you can cut your loudspeakers (without no more distorsion) lower, for exemple Yamaha Beryllium medium not at 500 Hz with 12dB/octave but 375 Hz with a breakwall cut of 80 dB/octave .
 
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dualazmak

dualazmak

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Hello Pos and Igor,

Although quite belated, now I fully aware and understand you two are "famous sound engineers"!

As an amateur beginner in software crossover multichannel system, I would highly appreciate hearing your discussion, comments and suggestions under the title of this thread "Multi-Channel, Multi-Amplifier Audio System Using Software Crossover and Multichannel-DAC".

Your general discussion and suggestions should be very much valuable and informative not only for myself but also for many people visiting here.
It is a great surprise for me that this thread has been visited and viewed more than 14,000 times in rather short period after I started on April 9;
https://www.audiosciencereview.com/...mber-area.36/&order=view_count&direction=desc
 
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dualazmak

dualazmak

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Hello Igor,

During my current trials with various multi-channel amplifier(s), I am using the protection capacitors, 68 micro-F for Beryllium SQ, and 10micro-F for Beryllium TW and Super Tweeter FOSTEX T925A, all Jantzen Audio Standard Z-cap 400V DC;
WS000650.JPG


WS000639.JPG


What do you think about these?
 
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Igor Kirkwood

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[QUOTE = "pos, post: 438551, member: 970"]
And if I am not mistaken you are Mr Kirkwood the famous sound engineer ;)
How have you been doing lately?
I saw you moved house, is your Yamaha active system still kicking?
[/ CITATION]

Hello pos :)

Happy to meet you here !

My new active system now in Briare (130 m3 / 5 m3 acoustic treatement Hih celling)

https://www.homecinema-fr.com/forum...-q-sys-core110f-4-subs-ns1000x-t30094413.html

For pre ringing FIR yesterday after a blind test no pre ringing FIR EQ
pano.jpg
95.15CL.2.90bBC.1-p3.png
95.hybrid.5.90dBC.1-p3.png
 
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Igor Kirkwood

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Hello Igor,

During my current trials with various multi-channel amplifier(s), I am using the protection capacitors, 68 micro-F for Beryllium SQ, and 10micro-F for Beryllium TW and Super Tweeter FOSTEX T925A, all Jantzen Audio Standard Z-cap 400V DC;
View attachment 70413

View attachment 70461

What do you think about these?

Sorry dualazmak I don't know what is better :(

But I prefer active filter withstrong breakwall in linear FIR
 
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dualazmak

dualazmak

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Sorry dualazmak I don't know what is better :(

But I prefer active filter withstrong breakwall in linear FIR

Thank you.
You said about 375Hz -80dB low-cut for Be-Squawkers (Be-SQ), and I would like to try it. Taking the possibilities of some damages on my beloved Be-SQs, however, I should be very much careful if I would go down below 450 Hz even in not so large sound volume. In any way, I will keep using the protection capacitors, at least until I fully deciding the multi-channel amplifier(s) in my project.
 

Igor Kirkwood

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Thank you.
You said about 375Hz -80dB low-cut for Be-Squawkers (Be-SQ), and I would like to try it. Taking the possibilities of some damages on my beloved Be-SQs, however, I should be very much careful if I would go down below 450 Hz even in not so large sound volume. In any way, I will keep using the protection capacitors, at least until I fully deciding the multi-channel amplifier(s) in my project.

450 Hz low-cut for Yamaha Be medium you are right , but only with IIR conventional filter

No problem with FIR breakwall filter 375 Hz (but I forget wrote than the medium Be hight cut is only 1800 Hz instance 6000 Hz for passive Yamaha NS-1000)

The evidence ?

The distorsion curve of active Yamaha NS-1000x (with 4 subs SVS PC2000) ) at 100dBC and distance earing 3,10 meter :
dsto 100.PNG
 
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dualazmak

dualazmak

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Hello Igor,

Thanks a lot again for your kind suggestions, valuable information and the wonderful evidence data.

Since at present I have no plan to use other SP units for tweeters and super tweeters, I will move forward in this project with current SPs, i.e. two sub-woofers YST-SW1000 (active with powerful amplifier in it), NS-1000's WO, Be-SQ, Be-TW and FOSTEX T925A.

>450 Hz low-cut for Yamaha Be medium you are right , but only with IIR conventional filter

I am now using software crossover EKIO in this project which uses IIR filters; the processing is done using a cascade of second order transposed direct form II biquad sections. Every calculation is done using 64 bit floating point numbers. (please refer to post #139 and #140).

Consequently, as you kindly suggested, I will try "WO to Be-SQ" crossing around 450 Hz using EKIO's IIR with 12, 24 and 48dB/Oct slopes (48 dB/Oct is the steepest with EKIO). Since I use the YAMAHA original Be-TW, the cross of "Be-SQ to Be-TW" will remain around 6,000 Hz, if otherwise suggested by you.

My listening position is about 3.8 m from the surface of L&R NS-1000, and your distortion curve at 3 m with NS-1000X is encouraging me to go forward with my step-by-step configurations in this project with current SP units.

I am waiting for the arrival of two of DENTEC DP-NC400-4, rather heavy duty 4-channel amplifier with four Hypex NC400 modules in it, for my free trials scheduled in late July, and I will try the crossover at 450 Hz after the arrival of DP-NC400-4.

In any way, I will keep using the protection capacitors at least until I would fully decide the multichannel amplifier(s) in this project.

Thank you again, and I highly appreciate your continuing kind attention on this thread and further possible valuable suggestions.
 
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dualazmak

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An Interlude of the Project:
Sync connection from DAC8PRO to second DAC8PRO, or to another DAC, possible?

Edited to add on July 7:
In his post today here, Pavel of OKTO RESAERCH kindly confirmed the AES/EBU digital sync connection from DAC8PRO to second DAC8PRO or to another DAC.

Hello friends,

As you well aware by looking at many schematic diagrams of my current project, OKTO DAC8PRO, an amazing 8-channel multi-channel DAC, is the core of my multi-channel system.

One of my audio-enthu friends in Japan is now very much interested in building stereo 8-way 16-channel system with software crossover and two of DAC8PRO, and he has recently placed his order for two DA8PROs! As you aware, one DAC8PRO is just enough for my current project, while he and myself are interested in how we could establish two DAC8PROs in sync with each other.

Recently, I had a series of interesting discussion with @Zooqu1ko in the Review thread of DAC8PRO, from here to here. We agreed, and also the Owner's Manual of DAC8PRO tells, that if we connect the AES/EBU digital OUT (CH1+CH2) of DAC8PRO into AES/EBU digital IN of second DAC8PRO, or into another second DAC capable of AES/EBU digital IN, then the second DAC would be automatically in sync with the first DAC8PRO, without using the internal clock for DA processing, but using the sync signal provided by the AES/EBU digital input.

As described here, I tested and confirmed such sync connection to second DAC, and let me share that trial and result again here, and some additional features in this project will be also shared afterwards in this post.

I tested the "sync connection" by putting AES/EBU digital OUT of my DAC8PRO into second DAC, not DAC8PRO, but ONKYO DAC-1000(S) which I have and capable of AES/EBU digital input;
WS000678.JPG


Before my actual trial with DAC-1000(S), I sent my inquiry to ONKYO's customer support center asking; "If I would connect DAC8PRO's AES/EBU digital out (CH1+CH2) to DAC-1000(S)'s AES/EBU input, DAC-1000(S) would be fully in sync with DAC8PRO for the DA processing, right?".

An ONKYO engineer quickly responded by email that "If DAC8PRO's AES/EBU digital out is fully compatible with the IEC90958 of AES3 Standard issued in 1985, then DAC-1000(S) would be automatically fully in sync with DAC8PRO, without using DAC-1000(S)'s internal clock."

As DAC8PRO is also for pro use, its AES/EBU digital out (CH1+CH2) should be fully compatible with the IEC90958 of AES3 Standard, and as described in the Owner's Manual of DAC8PRO, the second DAC would be automatically in sync by this AES/EBU connection, I believe.

Then I actually tested by connecting DAC8PRO's AES/EBU digital OUT (CH1+CH2) into ONKYO DAC-1000(S), and listened to both sound, i.e. sound from DAC8PRO's XLR analog out (CH1 & CH2), and sound of ONKYO DAC-1000(S)'s XLR analog out (L & R), for a continuous period of overnight (15 hours). As the result, I found no asynchronization nor drift of sync between the two DACs; DAC-1000(S) has been fully in sync with DAC8PRO.

This in sync connection to DAC-1000(S) gives another option in my project for connection to L & R sub-woofers YAMAHA YST-SW1000 by RCA unbalanced input.

In my previous post #192, I showed that RCA unbalanced input into L & R sub-woofers YST-SW1000 can be achieved by using DAC8PRO's nice headphone output always giving CH1+CH2 line level signal under the control of DAC8PRO's volume and gain controller, just like in this scheme;
WS000681.JPG

In this case, I need to use a "TRS to two female RCA (L&R) adapter" at the front panel of my DAC8PRO;
WS000662.JPG


Now, I have the new option of using second DAC, ONKYO DAC-1000(S), connecting its RCA unbalance OUT into RCA IN of L and R sub-woofers YST-SW1000 as shown here;
WS000682.JPG


with EKIO's I/O setting of;
WS000677.JPG

The main reason for keeping the super-low (SL) 15 - 50 Hz channels in this EKIO I/O configuration is that in a few music tracks in my library ripped from CD or digitized from LP records, unpleasant small volume super low Fq noises exist around 20 - 35 Hz caused by hall air conditioning or by LP records' bending which can be muted (if needed) by the Mute buttons of SL-L and SL-R Output panels. Except for these rare "super low Fq noise" cases, of course I usually need 15 - 50 Hz sound mainly played by sub-woofers.

We should note,however,that DAC8PRO's AES/EBU digital out (CH1+CH2) is the through digital out of CH1+CH2 which is not under the control of DAC8PRO's volume or gain controller. In this setting, I need to control the sound volume of sub-woofers by its own volume controller using an IR remote controller. Of course, even in this setting, I can use EKIO and/or JRiver on PC for master volume and gain controller after I would fix the volume balance in down stream.


In any way, it is nice to know "DAC8PRO's AES/EBU digital OUT (CH1+CH2) into second DAC" enables in sync simultaneous DA processing with DAC8PRO and the second DAC.

Edited to add on July 7:
In his post today here, Pavel of OKTO RESAERCH kindly confirmed the AES/EBU digital sync connection from DAC8PRO to second DAC8PRO or to another DAC.
 
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dualazmak

dualazmak

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Again, Where to Control Master Volume?

Hello friends,

As you aware, I am in the step of trials and decision on multichannel amplifier(s) in my project here. I will soon try (for free) two of DENTEC DP-NC400-4, rather heavy duty 4-channel amplifier with four Hypex NC400 modules in it, for my free trials scheduled in late July.

As shared in my previous post #190, I decided to use the second DAC, ONKYO DAC-1000(S), connected from DAC8PRO by digital AES/EBU (CH1+CH2) for RCA unbalanced input to my L and R sub-woofers YAMAHA YST-SW1000. The scheme of my coming trial with DP-NC400-4, therefore, will be;

WS000692.JPG


Since the AES/EBU digital out (CH1+CH2) of DAC8PRO is not under the volume/gain control of DAC8PRO, I need to use EKIO or JRiver (Roon) as Master Volume Controller.

"Where to control Master Volume?" is always one of the major issues in any multi-channel project, and I have already discussed and shared this topic in my post #58, #62 and #130.

During those discussions, in his post #60 in this thread, @Burning Sounds kindly suggested as;

>From a technical standpoint it is perfectly fine to use the Dac8 PRO volume control to set overall gain. It uses 32 bit floating point, so no problem even after considerable attenuation. JRiver and Roon use 64 bit and dither down to 32/24 bit so no problem there either. My only observation is that DAC8 PRO has a granularity of 1dB whilst JRiver is 0.5dB which I find suits me better. So my take is use whichever suits you better, either will be fine.

>@dualazmak - If you are using software x-overs and EQ more of an issue is to ensure you don't run into digital clipping. If I remember correctly EKIO is done in a 64bit environment (you can do considerable attenuation on any channel and not suffer quality loss) and has peak level meters for each channel - these should enable you to check that you are not running into clipping. I use JRiver for my x-overs and EQ and use one of the 8-channel Peak meter VST plugins to check. It needs to be after PEQ of course. Typical peaks on the loudest channels for me are around -10dB.

>JRiver (and Roon) enables you to set some digital headroom to help prevent clipping. In neither application is there a loss of sound quality. I use volume levelling in JRiver and this also gives additional headroom. It's all done inside the 64bit audio engine - there is no quality loss. JRiver's Loudness control also works very well for late night listening at reduced volume. If you have a dB meter you can calibrate it and it is seamless in use.

Having the current I/O configuration in my project and above nice suggestions, now I would like to use JRiver (or Roon) as Master Volume Controller where all of the volume balance and gains should be preset in down streams still in rather high gain to minimize the possible bit loss, as shown here;
WS000691.JPG


Yesterday, I carefully tried and decided the presets of volume balance and gains as shown above, and there is no problem in total sound quality even with considerable plus or minus total volume control by JRiver. Now I fully agree with @Burning Sounds' preference of "JRiver has volume granularity of 0.5dB which suits better". I also like the JRiver's keyboard Volume Control functionality by "Ctrl +, Ctrl -" for up and down in 0.5dB step, and also "Ctrl M" to mute the sound.

I am looking forward to my trials with two of DP-NC400-4 soon in late July, using these volume/gain control configurations.
 
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Kal Rubinson

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Yesterday, I carefully tried and decided the presets of volume balance and gains as shown above, and there is no problem in total sound quality even with considerable plus or minus total volume control by JRiver. Now I fully agree with @Burning Sounds' preference of "JRiver has volume granularity of 0.5dB which suits better". I also like the JRiver's keyboard Volume Control functionality by "Ctrl +, Ctrl -" for up and down in 0.5dB step, and also "Ctrl M" to mute the sound.
I use the Jriver volume control but I have added this for convenience:
71J-n40mwpL._AC_SX425_.jpg
 

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All good stuff.
For wireless JRiver volume control I mainly use a BT trackball (w/buttons programed for volume/mute), but also have (several) old Windows Media Control wireless remotes, which work well for emergencies - at least until Pavel releases the software update that adds AES output muting via their remote.
Fumbling for a keyboard or trackball setting (cursor has to be on the volume area on screen) during a volume bump accident is way too slow for me:) (I don't usually have my BT keyboard connnected during listening sessions).
 

Kal Rubinson

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For wireless JRiver volume control I mainly use a BT trackball (w/buttons programed for volume/mute),

The best/worst is the Microsoft Surface Dial. I have one programmed for volume (rotation) and toggle pause (press) and those operations are intuitive. It is also clean and easy to place...................................except that its BT range is horribly short. (Any ideas to help?
03rWJPcVAO4z0of60SR0dcw-7.fit_scale.size_1028x578.v_1569479936.png
 

Neddy

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I saw your posts on that, and nearly ordered one, but didn't feel it was worth the fiddling time investment - otherwise it looks nearly ideal.
I have had intermittent problems with my skullcandy NUC and BT 'reception' with Logitech trackball & keyboard, some (most) of which were due to some older Blue Aura wireless speakers I was trying to use for one surround location.
I've since removed them, but even so still have occasional BT/Unifying Receiver connectivity issues.
(*Another reason to have a 'backup' remote mute device!)
I also read that some NUCs had BT connectivity issues...but haven't fully dug into that (there are some Intel tech notes about it).
The perfect wireless HT/Audio controller still does not exist, as near as I've been able to determine - if only I could just point my finger at it and tell it what to do!!
(Haven't tried voice control yet, but is on my 'someday' list).
Thanks Kal - keep trying on the MS Dial, and let us know if you find any magic!
 
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dualazmak

dualazmak

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Thank you Kal and Neddy for nice info...

Even though I have no problem of using keyboard in front of me on low tiny table, I am also interested in DROK which is available from Amazon Japan;

Just for confirmation, does DROK fully work for JRiver which is running full ASIO I/O, and not using Windows WDM or kernel mixer functions? After installing DROK and its driver, should I select "Application volume" at JRiver's volume property panel?
WS000000.JPG
 

Kal Rubinson

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Just for confirmation, does DROK fully work for JRiver which is running full ASIO I/O, and not using Windows WDM or kernel mixer functions? After installing DROK and its driver, should I select "Application volume" at JRiver's volume property panel?
Cannot recall. We are away for the weekend but I'll check when I get back to NYC.
 
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dualazmak

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