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[Juniorjbl]

Ok, so I have a MOTU 24Ao hooked up via USB and I can not figure out how the channels are mapped to the device. For example JR only allows me to select left right etc for crossovers and eq settings. How do I know what channel of the 24Ao these are mapped to? I have set the channels section of the output format to 24 but I am lost here.

Second I cant seem to control the volume of the device either. I have tried internal, application, and so on.

Third it would seem that all system sounds and so on are bypassing JRiver. If that is the case then my fully active setup will all be for not if a sound is accidentally played outside of JR and destroy my High Frequency drivers. What can be done about this if anything?

Any help would be appreciated.

[AndrewFG]

Look at the channel ID table on this page https://en.m.wikipedia.org/wiki/Surround_sound

[Juniorjbl]

Maybe I am a bit challenged here but I am not finding a 1 to 1 chart of what is mapped to what on my 24Ao. Example channel 1 on my 24Ao is ? left? Is the mapping hard set in JRiver? Can we re-route like I can on my BSS BLU 160 where 1 is just a number on the piece of hardware that can be renamed.

All the chart told me was the standards but no mention of User 1 and User 2 and the other options in JRiver that I can not find any way to map these.

[JimH]

You could use Windows Control Panel > Sounds to test.

MC can re-map channels.  It's in the DSP Studio > Output Settings.

[mattkhan]

Maybe I am a bit challenged here but I am not finding a 1 to 1 chart of what is mapped to what on my 24Ao. Example channel 1 on my 24Ao is ? left? Is the mapping hard set in JRiver? Can we re-route like I can on my BSS BLU 160 where 1 is just a number on the piece of hardware that can be renamed.

All the chart told me was the standards but no mention of User 1 and User 2 and the other options in JRiver that I can not find any way to map these.

The mapping to a physical output channel will be done in the motu mixer app. If you use jrss then jriver mixes to a 7.1 target in the 24 channel output mode. If you don't use jrss then what ends up on channels 1-24 is entirely up to you based on how you mix channels in the peq blocks and/or the convolver.

You need to state exactly what you are trying to achieve if you want more specific guidance.

[Juniorjbl]

I would like a BD rom to play on the PC in say 7.1 and output to a fully active system ie: Left HP channel 1 left LP channel 2, Right HP channel 3 Right LP channel 4, Center HP channel 5 Center LP channel 6 and so on.

I would like to build this in JRiver without buying $2000 BSS BLU 160's for speaker management. I have Mojave's post about it but I think the only active XO's he is using is sub woofers where as my entire system is active XO's.

Thanks for any guidance you folks may have.

[mojave]

You need to set your routing in the MOTU control panel first. You can even label the channels to make it easier. On the Device page > Computer Setup, you put how many channels you want to use. This makes it easier to manage in the Routing page.

On the Routing Page, you route all the From Computer inputs to the Analog outputs. It is easiest if you just route sequentially from 1-24.

Now in JRiver you use Mix Channels to copy and create the active crossovers. Here is the channel order for 7.1:
1-Left
2-Right
3-Center
4-Subwoofer
5-Rear Left
6-Rear Right
7-Surround Left
8-Surround Right

Since you can't change names in JRiver, it can get confusing. It is best to write it down first and then use Mix Channels. For example, you might want to do the following:
1.Left High
2.Left Low
3.Right High
4.Right Low
5.Center High
6.Center Low
7.Subwoofer
8.Rear Left High
9.Rear Right Low
10. Rear Right High
11. Rear Right Low
12. Surround Left High
13. Surround Left Low
14. Surround Right High
15. Surround Right Low

So you do the following Mix Channels in this order starting with the highest channel:
Move Surround Right to 14
Copy 14 to 15
Move Surround Left to 12
Copy 12 to 13
Movie Rear Right to 10
Copy 10 to 11
Move Rear Left to Surround Right
Copy Surround Right to 9
Move LFE to Surround Left
Movie Center to Rear Left
Copy Rear Left to Rear Right
Move Right to Center
Copy Center to LFE
Copy Left to Right

You end up with the following:
1.Left = this is Left High
2.Right = This is Left Low
3.Center = Right High
4.Subwoofer = Right Low
5.Rear Left = Center High
6.Rear Right = Center Low
7.Surround Left = Subwoofer
8.Surround Right = Rear Left High
9.Rear Right Low
10. Rear Right High
11. Rear Right Low
12. Surround Left High
13. Surround Left Low
14. Surround Right High
15. Surround Right Low

Now add your high and low pass filters:
Left - High pass for Left High
Right - Low pass for Left Low
Right - High pass for left Low if necessary
etc.

To send all system sounds through JRiver, you need to go to General > Features and check the WDM Driver. Reinstall JRiver or upgrade to latest version and install the WDM Driver. Go to Sounds in Windows Control Panel and set the WDM Driver as default. For the 24ao, it is best to use a sample sample rate for all content. Set JRiver to output 96 kHz and set the WDM Driver in the control panel to output 7.1 at 96 kHz.

You can use leftover channels in the 24ao for other Zones in the house (deck, garage, kitchen, etc.).

[Juniorjbl]

Thanks for the information Mojave!

I am still not sure where this "Mix Channels/copy" operation happens. I have been looking through all of the menus but I must be missing this.

I had previously mapped/labeled the MOTU as I am familiar with digital mixing consoles and speaker management systems just not the JR layout.

[Juniorjbl]

Never mind I found it.

So next question do I need to have this "EQ" list in the order I want the processing to happen? This seems like it would add quite a bit of latency to my audio path.

[Juniorjbl]

Also what is "User 1" and "User 2" in the drop down channel list and what number channels do they occupy? They are right after 7.1 and before what would be channel 9 on the MOTU. I sure wished you could rename these "Channels" in JR.

[mojave]

So next question do I need to have this "EQ" list in the order I want the processing to happen? This seems like it would add quite a bit of latency to my audio path.

The most complex Mix Channels only takes a millisecond or so to compute. No latency. Think of a spreadsheet doing math.

Also what is "User 1" and "User 2" in the drop down channel list and what number channels do they occupy? They are right after 7.1 and before what would be channel 9 on the MOTU. I sure wished you could rename these "Channels" in JR.

User 1 and User 2 are virtual channels for DSP without disrupting other channels. They don't occupy channel numbers because they need to be routed back to real channels. Someone might want to copy Left to User 1, perform some EQ on it, and then add it to Right. They want Left to stay the same, but need the EQ before adding to right. This is a nonsensical example, but you get the point.

[Juniorjbl]

Is there a benefit to using "Parametric Equalizer 2" as well as the First PEQ in DSP Studio? What is the purpose?

[mattkhan]

Is there a benefit to using "Parametric Equalizer 2" as well as the First PEQ in DSP Studio? What is the purpose?

One use is to apply eq before and after convolution, ie to the input channels and then to the output channels.

[Juniorjbl]

Thanks for the reply!!

That makes sense.

[AudioMunchies]

JRiver should add an advanced feature which allow us to rename the channels to make it easier to work with. I'll be working with 16 channels in an active XO setup.

[gj_1970]

i have motherboard with on board sound card and 5.1 analog home theatre attached to the pc. Can anyone tell how to map correct channels with parametric equalizer

[Matt]

i have motherboard with on board sound card and 5.1 analog home theatre attached to the pc. Can anyone tell how to map correct channels with parametric equalizer

Ideally you'll already have the right channels hooked up to the right stuff so just picking 5.1 as the output is all you need!

But if you need to customize the channel ordering, just use Parametric Equalizer > Add... > Order channels.  Then just drag the channel list into the right order (trial and error should get you there).

[Paul W]

Edited to update mapping table.  (I found two different "standards" for rear and surround channel mapping...had me very confused.)

Setting up Acourate for the first time and would appreciate any input.

To understand the system:  It is 11.1 channels with bi-amped LCRs plus SBA subs in a baffle wall, 8 passive surrounds with 4 of them standard, the other 4 derived from the standard 4 using delays and frequency tapering.  The center channel is designed for a little wider polar pattern than the left & right channel so the ACs are a couple of inches closer to the MLP...definitely not the same timing.  The player used for all sources is JRiver with a Lynx AES-16e output to a Lynx Aurora 16 DAC with everything run at 96k.  Processor is a 3.5GHz i7 with 16G RAM.

Blowing expensive tweeters in the LCRs is scary so I need to get this right from the start.  Does this mapping look right?



Latency is a concern because we use the WDM driver to watch a lot of Netflix etc.  Should processing be changed to 48k or will latency be okay at 96k?

My DSP knowledge would fit in a thimble so all input is appreciated!



PS:  Sorta related....can JRSS be used with this mapping???
PPS:  Adding a single number in front of JR channels 1-8 would be helpful when mapping in PEQ....1 Left, 2 Right, 3 Center, etc...

[Paul W]

Friendly bump for feedback on edited post.

[mattkhan]

Friendly bump for feedback on edited post.

tbh I don't really understand your notation and/or each path is unclear

If you are using acourate for this, why use peq at all? Just do it in the cfg and/or the filter

You need a low latency filter for wdm btw, do you have one?

[mojave]

Friendly bump for feedback on edited post.

8:  I don't understand what you are doing here.
13:  You need to adjust channels by -10dB before adding to sub. You might consider using Room Correction first for bass management before copying channels in PEQ.

You are using delays and frequency tapering on the 4 derived channels. You might consider using matrixing so that the SL+, for example, is getting part of the signal from SL and part from RL. You can weigh it so that SL+ gets more from SL and RL+ gets more from RL. You also need to measure using JRiver's test signals to make sure you are maintaining equal energy in the room. This means that you might need to lower the level of SL.

If using Audiolense, you don't need to do any of the routing, combining channels, or bass management in JRiver. It is all done in Audiolense and is super easy.   I did 14 channels for The Savoy Theater that had one less set of surrounds and a rear subwoofer for the surround speakers.  

[Paul W]

Hi Matt,

This is a case of "I don't know what I don't know".  As mentioned before, my knowledge of DSP would fit in a thimble, so please bear with me.

The objective is 15 output channels as listed in the right column.  My (weak) understanding is that remapping JR channels is required to achieve the objective. 

If all of the output channels can be made available via the Acourate config file I'm all ears!  Recommended reading?

The mapping above works fine with the WDM driver, JR crossover and EQ filters...but I am concerned about latency after changing to Acourate filters with the WDM driver.  Is there another Netflix solution that doesn't require that driver?

[Paul W]

mojave,
8.  This is just a temporary "fix" to add a different XO frequency for LCR than the LFE.  Shouldn't be necessary in the final config.
13.  Yes, still playing with that.

Interesting suggestion for the surrounds...but I think I should crawl before trying to walk!

Got a link to the Savoy?

[mattkhan]

Hi Matt,

This is a case of "I don't know what I don't know".  As mentioned before, my knowledge of DSP would fit in a thimble, so please bear with me.

The objective is 15 output channels as listed in the right column.  My (weak) understanding is that remapping JR channels is required to achieve the objective. 

If all of the output channels can be made available via the Acourate config file I'm all ears!  Recommended reading?

when you say biamped LCR, do you mean LCR is an active 2 way with an acourate provided crossover or something else?
are your mains running full range or are you intending to low pass them into a mono SW feed?
if they are bass managed then are you doing this in your acourate filters or expecting jriver to do it?
what does "Add L>R+U1" mean? the trail through U1 seems to end at your sub but I'm not really sure what it means

I don't think it's possible to say what the right routing is without knowing what your acourate filters are actually doing

The mapping above works fine with the WDM driver, JR crossover and EQ filters...but I am concerned about latency after changing to Acourate filters with the WDM driver.  Is there another Netflix solution that doesn't require that driver?

you would need to use minimum phase acourate filters or use acourate purely for high frequency phase correction & use minimum phase to correct any frequency response issues or implement an approximation using jriver PEQ. The exception to this is if your display has high latency (like some projectors) which can increase the latency budget available to your filters. IIRC mojave exploits this in some setups.

[Paul W]

The mapping above is just in preparation for implementing Acourate FIR and/or IIR crossovers and EQ via the convolver.  All crossovers, delays, EQ etc are currently implemented via PEQ.

The LCRs are active 2-way with active XO to mono subs.  All currently done in PEQ, will be changing to convolver.

"Add L>R+U1" means left is added to the (empty) right channel to produce the Left Hi output.  U1 is a common output for LCR VLF which is low-passed and then on to the subs.  (As mentioned by mojave, this could probably be done in Room Correction.)

PJ is an Epson 5030UB.

So there is no solution for Netflix other than the WDM driver, correct?

Any recommended reading for creating convolver config file(s)?  I'm an old analog guy.

[mattkhan]

your configuration will change when you move the XOs to the convolver, it will be simpler in fact... you will then just need

PEQ to attenuate main input channels by 15dB + LFE channel channel by 5dB
convolver
PEQ to add gain back to main channels (and LFE channel unless you add the gain back in an analogue stage)

your convolver cfg will be something like this, I added comments using # to start them (though the cfg format doesn't support comments so you'd remove that later).

This covers your L and R channels and assumes; 1) you output stereo wav files from acourate, 2) you create a 3 way crossover in acourate where way 1 = sub, way 2 = main channel low, way 3 = main channel high.

Code: [Select]

48000 16 16 0
0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
C:\your\filter\path\Cor1S48_LR.wav
0       # 1st channel in the filter
0.0     # input channel 1 aka L
15.0    # output channel 16 aka sub
C:\your\filter\path\Cor2S48_LR.wav
0       # 1st channel in the filter
0.0     # input channel 1 aka L
0.0     # output channel 1 aka Left Low
C:\your\filter\path\Cor3S48_LR.wav
0       # 1st channel in the filter
0.0     # input channel 1 aka L
1.0     # output channel 2 aka Left High
C:\your\filter\path\Cor1S48_LR.wav
1       # 2nd channel in the filter
1.0     # input channel 2 aka R
15.0    # output channel 16 aka sub
C:\your\filter\path\Cor2S48_LR.wav
1       # 2nd channel in the filter
1.0     # input channel 2 aka R
2.0     # output channel 3 aka Right Low
C:\your\filter\path\Cor3S48_LR.wav
1       # 2nd channel in the filter
1.0     # input channel 2 aka R
3.0     # output channel 4 aka Right High

You may choose to remove the pre convolver attentuation (in the PEQ block) by attenuating in the filters or attenuating in the input channel line in the cfg. The latter is referred to as input channel scaling. For example, -15dB expressed as a scaling factor is 10^(-15/20) = 0.177828 so to attentuate the left input channel by -15dB you would change

Code: [Select]

0.0     # input channel 1 aka L

to

Code: [Select]

0.177828    # input channel 1 aka L scaled by -15dB

i.e. this line is written as <input channel number>.<scaling factor>

You can't avoid adding gain in the post convolver PEQ block as output channel scaling is not supported by jriver.

You may ultimately find you can collapse a few paths too but if you have the CPU for it then there's probably no issue.

cfg file reference is http://convolver.sourceforge.net/config.html but you may find some threads on interact or the acourate board more useful

[mojave]

Got a link to the Savoy?

Got a working link to the Octagon? 

The Savoy

Home Theater of the Month - June 2014

I don't think there is any difference, but I prefer to use "copy" instead of "add" when going to an empty channel. This makes sure they are identical. For example, "Copy L>R+U1".

So there is no solution for Netflix other than the WDM driver, correct?

Not internally. You can use the Netflix app on an external source (Blu-ray player, STB, Roku, etc.) and use the Hauppauge Colossus 2 and a cheap HDMI splitter (to remove HDCP) to bring the source into JRiver. This works and allows one to use madVR, too. Not all devices output multi-channel sound as Dolby Digital over HDMI, though, which is all the Colossus 2 works with for multi-channel.

Using Zones and ZoneSwitch in JRiver is an easy way to prioritize your sound quality and automate output. The Blu-ray zone gets the full 15 channel Acourate filters. The Netflix Zone might use fewer surround channels and use JRiver's PEQ for lower latency.

Do you already own Acourate? Audiolense is setup much better for multi-channel systems and creates the convolver files for JRiver (Acourate might too, but I'm not sure).

What are you using for a measurement mic? It will work best if using the line input on the Lynx.

[mattkhan]

Acourate might too, but I'm not sure

it doesn't

[Paul W]

Thanks Matt!
You've provided even more help than I was hoping!  To me, configs look like they were written in Martian but, hopefully, I'll be able to noodle through it from what you've provided.



mojave,
Desert Dome?
No working link...sorry  .  Gave up on my free IT guy so, though I have less-than-zero interest in the mechanics, I'll redo the page myself after Acourate has settled.

I bought Acourate almost 2 years ago and, though both were considered, don't remember why it was picked over Audiolense.  The plan was to implement it right after another project was complete, but that was interrupted by injuries from a shipwreck in the Galapagos Islands (long story).  Up & running again with Acourate the last piece before Octagon 2.0 is finished (new baffle wall and WGs).

Yes to multiple Zones & Netflix.  Acourate can generate both IIR and FIR correction filters so I'll try NF via Acourate IIR first, with a fallback to PEQ.  (Netflix has become the #1 play here so it needs to be the best possible.)

Mic is an ECM-8000 calibrated by Kim G.  Pre-amp + cable are so flat I don't even use a cal file for them.  Already made an adapter for Lynx line-in so should be GTG there.

Summer will get very busy soon...so no date for the new page but the long-term plan is to back-fill with a dozen + projects.