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[beginner44]

Hi there,

since at least 33.0.60, my sound is KO. I've been using the same setup for almost 10 years (12 channels soundcard + convolution from Audiolense).

I've been running MediaCenter330052-x64 since January witout any issue, but yesterday there was an update to .60 and the issue showed up. I installed this latest version 0.70 => same. I uninstalled  MC33 and re-installed MediaCenter330052-x64. Issue is still there !

So I installed MediaCenter320042-x64 (which I was using before moving to MC 33 last January) : everything works fine !

When I say my sound is KO, I mean it is quite muted, and in the analyser it looks like no high frequencies are routed to the tweeters.

I do not know what other information I can provide to help you correct the issue. Just let me know if you need my convo files or any log files.

Many thanks for your help.

BG44

[mattkhan]

Sounds like you are impacted by

33.0.67 (2/13/2025)

1. Changed: Revised the layout of 12 channel VST3 to be the standard 7.1.4 and 13 channel to be 7.1.4 + extra LFE.

Details of the change in https://yabb.jriver.com/interact/index.php/topic,140559.msg975271.html#msg975271

[beginner44]

Hi mattkhan,

thanks for the information.

Yes that looks very probable.

But does this mean that I have to re do my convolution ? Or is there any hope of fixing this issue while offering the new functionnality for those who need it ?

[mattkhan]

 I would guess you need to edit the cfg file to change how channels are routed

I don't know if they would add an option, one for Matt to comment on

[Matt]

I would expect the indexes in the CFG file to be unchanged.

But other processing you perform in Parametric Equalizer might need the channels updated.

I'm sorry for any hitches our changes caused. We're simply trying to adopt the standard for channel routing.

Thanks.

[whoareyou]

And here's link to how everything is mapped with the new system:

https://yabb.jriver.com/interact/index.php/topic,139490.msg968285.html#msg968285

But I thought these changes began in like build 33.0.14

[beginner44]

@Matt :

I understand the need to adopt new standards. But I don't have my Audiolense program anymore, so redoing the convolution would require buying a new licence...

To make everyboday happy, would it be possible to have new and old version of the 12 and 13 channels mapping available in the drop list, so that user can choose which one to use ?

Plus the new one says 7.1.4, but my setup is a 7.1 that goes as follows (channel numbers not necessarelly accurate, just for the sack of explenation):

Front Right : tweeter ch1, medium ch2, sub/LFE ch11
Front Center : tweeter ch3, medium ch4
Front Left : tweeter ch5, medium ch6, sub/LFE ch12
Surround right : ch7
Surround left : ch8
Rear right : ch9
Rear left : ch10

I doubt very much that I will be the only one to have this issue, as JRiver is one of the few media center out there to manage convolution (and this was one of the reasons I chosed it 10 years ago !)

Thank you very much for considering this suggestion.

BG44

[mattkhan]

Can you post your convolution cfg file?

[Matt]

Posting a DSP Preset would be good too because it could be something other than convolution having problems.

[beginner44]

Please find below the DSP's screenshots for one of my zones (this one is for music ; I've got 2 more : one for true 7.1 soundtracks, on for 5.1 that I transform to 7.1 by copying the surround to the rear). But the issue is the same.
See how in the analyser screenshot, there are no frequencies above 1500Hz.

Also attached are the convolution files.

Many thanks for your help !

BG44

[Matt]

I'm wondering if channels #9, #10, and #12 are now actually something different.

The channel choosing interface probably needs to be updated to allow 7.1.4 channel selection.

[beginner44]

Thanks Matt.

Can you elaborate on "The channel choosing interface probably needs to be updated to allow 7.1.4 channel selection." ?

Is this some kind of modification in JRiver ? Or a manual modification I would have to make in the convolution config files ? Or using the DSP / Parametric EQ to route channels differently ?

Thanks again for your patience.

BG44

[Matt]

We switched so 7.1.4 is the layout and that uses these channels:
CAudioFormat::Channel_HeightFrontLeft
CAudioFormat::Channel_HeightFrontRight
CAudioFormat::Channel_HeightRearLeft
CAudioFormat::Channel_HeightRearRight

There's simply no way to address those channels in the DSP Studio today.

[Mitchco]

Just a quick note to say that Audiolense has been updated to conform to the correct Atmos layouts as well. See release notes: https://juicehifi.com/audiolense-download/

[mattkhan]

I do not understand your convolution setup at all, you said it's a 7.1 setup as follows

Front Left : tweeter ch5, medium ch6, sub/LFE ch12
Front Right : tweeter ch1, medium ch2, sub/LFE ch11
Front Center : tweeter ch3, medium ch4
Surround right : ch7
Surround left : ch8
Rear right : ch9
Rear left : ch10

firstly, it might be deliberate is discarding the LFE channel?

secondly the above, in convolver cfg terms, would be

0 -> 4,5,11
1 -> 0,1,10
2 -> 2,3
4 -> 6
5 -> 7
6 -> 8
7 -> 9

but what's actually present is

0 -> 0, 3, 8
1 -> 1, 9, 11
2 -> 3
3 (-6dB) -> 11

it's not remotely the same, it's radically different

are you sure this is the right cfg file?

[Matt]

Convolution should be unchanged I think. It's just that addition of 25 dB in Parametric Equalizer won't work for the new channels.

[beginner44]

Hi mattkhan,

thanks for your feedback.

First, as I said before, the number of the channels were for demonstration purpose only. Since then, I was able to find my documentation (dating back to 2012 !) and will use the correct numbers from now on.
Second, I may have confused you because, on the one hand, I provided the config file for the stereo setup, but I gave the details of the channels for the 7.1 setup.

So if we stick to the simpliest configuration, which is music / Hifi, as per the config file provided, here are the channel numbers (in "" are the names of the channels in JRiver) :

Front Left : tweeter ch9, medium ch1("left"), sub ch4 ("subwoofer")
Front Right : tweeter ch10, medium ch2("right"), sub ch 12

Looks that this is coherent with what you see in the config file (assuming this file is using a 0 base)

See the tab attached (sorry it's in French but I guess it is quite self explenatory)

I appreciate very much you triyng to help me.

BG44

[mattkhan]

Convolution should be unchanged I think. It's just that addition of 25 dB in Parametric Equalizer won't work for the new channels.

certainly the symptoms do correspond to that 25dB being missing but why won't it work? how do those channel names (e.g. Channel_HeightFrontLeft) relate to use of either a convolver or DSP studio PEQ when PEQ just uses named channels or numbered channels (L, R, C etc and C9-C32)

[whoareyou]

Without knowing how AL is configured for this convolution, I'm not sure one can understand why 25db parametric setting is necessary? 
If you've measured with AL (at least the current release), each driver levels would be sorted out by the generated convolution. 

The cfg file appears to apply proper convolution to each input channel, and also outputs to what is assumed to be the proper output DAC channels.  Also, 7.1.4 should open up the proper channels so it seems like this should work (at least for 2 channel) without the parametric settings. 

I think the question is why parametric needed at all.  Shouldn't it all be handled by the convolution?

[beginner44]

Hello whoareyou,

convolution tends to lower the output level quite a bit. that's why it is ofter necessary to boost the signal afterward in the chain.

These convo files were generated in 2012. Since then it has been working good enough for me.

I'm happy to provide information to help debugging the issue, but I just want it to continue to work the way it has over the last 10+ years. I don't have the time, money and energy to do all the process again.

Now I hope JRiver will fix this issue, that's all.

Cheers

BG44

[whoareyou]

Ah, I missed the left & right with the 25 db increase . Maybe I haven't seen enough convolutions, but based on my own AL convolutions (which are for a difficult room),  I've never experienced anything approaching this amount of gain loss. 

Curious as to what you are using for a DAC and if its gain is already maxed out ?

If you are comfortable with turning off the parametric equalizer perhaps you can at least determine if the convolution works without the parametric's added gain ?

[beginner44]

My DAC is a MOTU 1248. Set at -10db, so almost at max

I will try your suggestion and let you know.

Cheers

BG44

[beginner44]

Ok so I've done that : I inactivated the Parametric Equalizer 2 in the DSP, so no more + 25db on the 6 channels. I had to raison the volume to the maximum (versus 80% I usually use) to have a decent volume level.

And guess what ?

Sounds good !

So Matt was right :

Posted by: Matt
« on: Yesterday at 04:15:04 pm »Insert Quote
Convolution should be unchanged I think. It's just that addition of 25 dB in Parametric Equalizer won't work for the new channels.

What I do not get is why it was working before, and not anymore...

Removing Parametric Equalizer 2 from the audio path is not really a viable solution for me because for 5.1 and 7.1 I use it for more operations (high pass filters on my subs...).


So I hope that Parametric Equalizer 2 will soon be fully functionnal for all channels.


@whoareyou : thanks for the suggestion, very appropriate !

@Matt : what do you think ? easy fix ?

Thanks again all for your support,

BG44

[mattkhan]

this file is your stereo file? t-b&k linear phase f-default (true time domain) m-2.2 bis 10 janv 2017 - 480002.0_441.cfg

it uses channels 1,2,3,4,7,8 of the wav file where 2 & 4 are clearly the tweeter

the cfg file routes as follows (named MC channel -> wav file channel -> named MC channel)

L -> 1 -> L
L -> 2 -> C9
R -> 3 -> R
R -> 4 -> C10
L -> 7 -> SW
R -> 8 -> C12

the tweeter channels have ~9-10dB of headroom so if you add 25dB to them then if you either don't use internal volume or your internal volume goes beyond ~-15dB then you're into clipping if playing a full scale signal

80% is -10dB and most content is mastered such that true peak is close to or beyond full scale so that implies you're riding the limiter (aka compressing) because you're going beyond full scale digitally

if you then have the motu at -10 then your signal chain is basically

* convolve (which cuts somewhat)
* large gain into clipping
* hit the MC limiter (hence compressing)
* attentuate in the motu driver
* to the DAC/amps

i.e. down up compress attenuate

it's a gain structure problem basically

the fix to this is

* scale up all channels in the wav by ~9.5dB
* remove the PEQ gain block
* remove the cuts from the motu
* set the MC internal volume limit if necessary to prevent clipping the amps (if you say what they are then we can calculate that also)


separate to this, it's still not at all obvious where your actual problem lies in MC33

there is a link here - https://wiki.jriver.com/index.php/Verifying_DSP_Studio - which explains how to measure this stuff yourself which would show objectively where the problem lies. If you don't feel confident in doing that then export your dsp config to a file (i.e. dsp studio -> presets -> save to file) and post that file here *with DSP studio configured in a way that has the audible problems*

[mattkhan]

NB: you said you were happy with the sound so feel free to ignore all of the previous post, but maybe it's worth knowing that the current setup has problems and hence now might be a good time to fix it and maybe you get better sound overall as a result

[beginner44]

Hello Mattkhan,

thank you for your message. I'm very impressed, as I do not have all this knowledge !

Now that I'm putting the fingers back in the honey pot, I see that as an opportunity to improve things. The current setup is so much better than what I had before using convolution that, yes, in a sense I'm very happy with it. But if you think that there is room for improvement, I'm ready to take the leap, with your help  ! My only worry is to damage my speakers (especially tweeters), it happened to me before   

The config file is indeed the one for my stereo setup.

I do not see how to "* scale up all channels in the wav by ~9.5dB" ; should I need to redo the convolution measurements ?
When you say "* remove the cuts from the motu" : does it mean I should set it up at 0db ?

Please find attached the DSP file. Also attached a diagram of my setup. Amplifiers are mainly Crown.

Many thanks again for your help, very appreciated.

BG44

[mattkhan]

I do not see how to "* scale up all channels in the wav by ~9.5dB" ; should I need to redo the convolution measurements ?

it can be done in something like audacity, there's no need to redo anything as it's just changing the amplitude of the filter with no other change at all.

When you say "* remove the cuts from the motu" : does it mean I should set it up at 0db ?

yes though the Q is why you set it at -10 in the first place? I see your amps seem to have input sensitivity of 1.4V (though the vzx is switchable, what do you have that set to?) & the motu can output +20dBu (which is ~7.7V rms). 1.4V is ~ 5.4 dBu so if you were clamping the motu to a level that could not possibly clip the amps then it would imply you need ~-15 as max signal level coming out of the motu. Perhaps the -10 was a step in that direction?

If doing this, one can argue it's safest to do it in the motu (as the last step in the chain) if doing it digitally but, personally, i prefer to add analogue attenuation before the amps (e.g. this sort of thing https://diystrat.blogspot.com/2013/08/diy-attenuators.html). Alternatively you can handle it in the amp front end (though those dials tend to be rather imprecise and/or easy to inadvertently adjust)

Ultimately all I'm saying is that going up down up down is generally something to avoid so it's a good idea to simplify this.

[beginner44]

Thanks mattkhan

I'm still trying to figure out how to change the amplitude with Audacity. Will keep you posted.

Regarding why I set the MOTU at -10, it's just it was a kind of middle ground, and to keep some room to increase the output if necessary.

The VZX sensitivity is set at 1,44v

I've to confess you lost me with dBu  and rms LOL


If I understand correctly :  the best set up would be using only the convolution file, no EQ in JRiver, the MOTU set at 0dB, and being sure that even with JRiver volume set at 100% there is no clipping of amps nor danger for the tweeters, etc... and for the audience ears !

I like your analogue attenuation with physical componants, not sure I would be able to use the correct resistance value in my case (given my poor understanding of electronics ; but I can use a soldering der). Do you use this only to protect tweeters, or all speakers ?
On the other end, the amp's voluem button are on the back and the amps in a dedicated closed where nobody goes expect me, so the chance of inadvertently ajust are very low, hopefully.

Do you think that once I've identified the correct settings (physical amp volume (I don't thik I will have to adjust that much), MOTU at 0 dB, no dB boots in JRiver), it would be benefical to redo a convoluton measurement ? Which software is best in your opinion ?

Many thanks again for your assistance,

BG44

[mattkhan]

you may get marginally better performance out of the motu by driving it to full scale and then attenuating in the amps, it's also arguably physically safer and makes it easier/simpler to accommodate different max signal levels per amp/driver combo so, to my mind, it's the preferred approach.

0dBu is just another way of saying the rms level (of a sine wave) is ~0.775V, https://sengpielaudio.com/calculator-db-volt.htm lets you play with some numbers. I only mentioned dBu because that's how motu expresses their maximum output level (+20dBu) but your amps just talk in rms volts terms so we have to translate between the two units.

you only need to redo the filters if you want to change them, if you're happy then no need. I use Acourate myself but the learning curve is (very) steep.