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[Stream_HD]

At this point, for my Stereo Musical listening experience, how can I now setup 2.2 for my stereo music?  There is no dedicated subwoofer track on these tracks (unlike movies which do include a dedicates sub track) so how would I route the left signal to the left speaker and left subwoofer while doing crossovers on both?  So I would want:

Left Speaker - cutoff 80hz and below
Right Speaker - cutoff 80hz and below
Left Sub - cutoff everything above 80hz from left stereo signal
Right Sub - cutoff everything above 80hz from right stereo signal

Would I just do a crossover to the left and right speaker, and then send each 80hz and below to the outputs that have my subs on them?

Thanks!

[Matt]

This is the sort of thing Parametric Equalizer is made for.

It's basically:
Copy Left to SW1, the lowpass
Copy Right to SW2, then lowpass
Highpass Left
Highpass Right

[Matt]

I might add that if you don't want stereo subs, just mix to 2.1 in DSP Studio > Output Format.

Then copy the SW channel to the second subwoofer output.  Reduce both by 3dB after to maintain the proper energy balance.

[Stream_HD]

Thanks Matt.  Which output format am I picking though to start with?  If I try anything more than stereo or 2 channels, it tells me it can't play it back.  I tried 2 channels in 7.1 output container, 4 channels etc.  But it seems the ones I picked won't let me work with stereo tracks.

Edit - I figured it out.  The problem was I had setup a zone for 2 channel.  Well, I made my output driver the 2 channel ASIO of my DAC.  As soon as I changed the output to the 8 channel ASIO version I was able to choose 2 channels inside a 7.1 container and the bass came blasting in .

[JAVA Alive]

Hi, I have similar needs but a bit moire complex.

I whould like to do 2.2 from a simple stereo file but with 2 stereo sound cards for output (because I use 2 high quality external dacs taking only stereo inputs) :

 Stereo flow =>
   => stereo flow with low pass on output card 1
   => stereo flow with high pass on output card 2

Other questions :
  How to manage frequency cuttoff setup ? If I put same on both high and low pass, I will have a bump at that freq, right ?
  How to add phase and delay management ?

Hehe .. Quite complex. Thanks !

[mwillems]

Hi, I have similar needs but a bit moire complex.

I whould like to do 2.2 from a simple stereo file but with 2 stereo sound cards for output (because I use 2 high quality external dacs taking only stereo inputs) :

 Stereo flow =>
   => stereo flow with low pass on output card 1
   => stereo flow with high pass on output card 2

Other questions :
  How to manage frequency cuttoff setup ? If I put same on both high and low pass, I will have a bump at that freq, right ?
  How to add phase and delay management ?

Hehe .. Quite complex. Thanks !

You may run into troubles syncing those two external DACs, unless they allow for an external sync.  How are you controlling the two external DACs (And what kind of DACs are they)?  If it's a USB interface, unless the DACs will allow you to master/slave them or you have a wordclock, they will probably not sync correctly.  If you're using SPDIF outputs on the first DAC to drive the second DAC, it may sync, you just need to find a way to output to those SPDIF outputs in JRiver.  If that's what you're doing I can provide some additional advice.

As to your other questions:

1) Yes if you just use the default low pass and high pass you will have a "lump" at the crossover point because they are butterworth filters.  If they're symmetrical and even order, it will be a 3dB lump.  The easy way around that is to cascade two highpasses or lowpasses of the same order in parametric equalizer to create a higher order linkwitz-riley crossover, which sums flat at the crossover.  Two 6dB lowpasses will create a second order linkwitz-riley lowpass on the sub channel, which would then sum flat with two 6dB highpasses set to the same crossover point on the speaker channel, etc.  For more info on butterworth v. LR, see http://www.rane.com/note160.html

2) Delay is easy, parametric equalizer has a "Delay" module, and you can dial in exactly how many milliseconds you need and on which channels.

3) Phase adjustment is harder and, long story short, requires using convolution to really do it.  All of JRiver's parametric EQ fitlers are IIR filters (i.e. minimum phase filters), so they cannot change frequency response without also adjusting phase and vice versa.  To do frequency response-neutral phase adjustment you need to use an FIR filter, which JRiver supports through convolution.  But you'll need to create FIR filters in other software, and it's a complicated subject.  Search the forum for "convolution" and you'll find many threads, here's a recent one where a user discussed his method for creating FIR filters: http://yabb.jriver.com/interact/index.php?topic=84252.0

[JAVA Alive]

Hi mwillems,


 Thanks for quick answer.
 Yes, I've read about convolution and setup looks quite clear.
 But the thing I still dont understand is how to send outpout to several soud cards.

 Regarding DACs, I have a W4S DAC2 (only one at the moment).
 No way to synch them, I guess. But why whould it be necessary ? I need anyway to setup a delay in a 2.2 system, right ?

[mojave]

JRiver doesn't support outputting to two DAC's unless you link two zones. Also, because of how USB and the clocks of the DACs work the timing can be different between the two DACs each time you play something.

Since the W4S DAC2 supports AES input, you could get a Lynx AES16e card. This will be the audio device in JRiver. It then sends the data to both DACs. Since AES carries the clock info, both DACs should always be in sync.

[JAVA Alive]

Ok, clear, thanks a lot !

[JAVA Alive]

Hi, I'm bringing up this old topic because I have additional questions regarding previous answers given.

This is the sort of thing Parametric Equalizer is made for.

It's basically:
Copy Left to SW1, the lowpass
Copy Right to SW2, then lowpass
Highpass Left
Highpass Right

Yes, it seems to work but I'm a bit frustrated because I want to use convolution filters, not the ones of parametric equalizer.
Maybe the filters used by Parametric Equalizer are convolution filters. Anyone can confirm ?
If not, in MC18, I don't se how I can use 2 convolution filters : 1 for the low pass, 1 for the high pass.

Any idea ? Update to MV20 improves this topic ?

Thanks.