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[zz]

I have found no way to route audio through my DSP chain before serving the audio to a network client. Is there a way?

I have successfully used the Media Center convolution engine to do a stereo crossover with FIR filters, separating bass from treble, producing 4 channels. The parametric equalizer allows me to map output channels wherever I want. ASIO4ALL allows me to route the treble to one stereo USB DAC and bass to a different stereo USB DAC. All this seems to work after a fashion under both Win 7 and Win XP.

My DACs are connected to an old XP laptop sitting next to my amplifiers. The XP laptop is fed from the Win 7 box as either a library or a DLNA server. If I use simple FIR filters at 44.1, the XP box processes things ok in real time. Any higher bit rates or more complex filters produce dropouts in the audio. The processor can't keep up.

So, if I move the DSP to the more capable Win 7 box and use the XP laptop just to receive the 4 (or 6, or 8 ) channel audio and feed the DACs, all should be well, right? Signalyst's network HQPlayer does something like this; I experimented with HQPlayer, and it worked for me. The problem is that I find HQPlayer's library management atrocious compared to JRiver's. Also, it costs three times as much.

Eventually I'd also like to add digital room correction into the convolution. I think doing all the DSP on the server would be great, but I cannot find a way in Media Center to serve the result to my simple client machine short of using something like a Shoutcast server. Is there a way to accomplish this all within Media Center?

[bob]

The only DSP functions available in DLNA are volume leveling, stereo downmix and sample rate selection. They are under the advanced audio options in the DLNA server configuration.

[zz]

I have tried endlessly to find a way to invoke the DSP chain before transferring audio to the DLNP renderer. No success. I thought for sure that the ability to send the output to a null render and linking a zone to the player's zone would accomplish this. But, NO GO! The link linked the player's input instead of the output. Again, I thought for sure that I could run a second instance of MC and through a loopback accomplish what I want. NO GO! again.

You guys have lots of hidden, undocumented features. The consensus I see, though, is that you don't want to make MC too complicated. Well, it is already too complicated, but it seems to do everything except what might be required here! This seems like nonsense. JRiver seems to listen to it's users, except in this case. Please consider this another vote for this particular feature request.

I read with interest this thread:

http://yabb.jriver.com/interact/index.php?topic=50760.0

I would really like to have a look at what I might be able to do without JRiver folks adding this feature. Can you send me the header file for #include "AudioFormat.h"? I am a retired software developer, and I don't mind exercising my chops after several years.

But, still, this is a serious feature request!

[maniac0r]

I would welcome DSP (convolution) for DLNA streaming too, it was actually main reason why I have tried and bough the MC18 for MAC even I was not sure if I'll be able to make it working - similarly as you I was believing that the flexibility of MC18 will allow this. Hopefully this feature will get once implemented. Seeing the good progress of the MAC version I'm pretty sure JRiver team can make this happen 

[mwillems]

I have found no way to route audio through my DSP chain before serving the audio to a network client. Is there a way?

I have successfully used the Media Center convolution engine to do a stereo crossover with FIR filters, separating bass from treble, producing 4 channels. The parametric equalizer allows me to map output channels wherever I want. ASIO4ALL allows me to route the treble to one stereo USB DAC and bass to a different stereo USB DAC. All this seems to work after a fashion under both Win 7 and Win XP.

My DACs are connected to an old XP laptop sitting next to my amplifiers. The XP laptop is fed from the Win 7 box as either a library or a DLNA server. If I use simple FIR filters at 44.1, the XP box processes things ok in real time. Any higher bit rates or more complex filters produce dropouts in the audio. The processor can't keep up.

One thing that might help your situation is to try and do less of the work of bi-amping with convolution and use JRiver's PEQ for the heavy lifting.  I also have a bi-amped system using JRiver, and I can confirm, for example, that the PEQ module crossover filters and EQ are much less processor intensive (and latency producing) than doing the exact same things with convolution filters.  

You could do most of the brute work in PEQ and then just apply limited convolution on top for room correction or phase linearization (which works pretty well in my experience).  For example, on my PC a simple convolution filter that linearizes the phase of a PEQ-applied crossover runs at something like 80x real time in the DSP Studio performance display.  A convolution filter that does my whole crossover and EQ runs at 20x real time (i.e. four times slower).  You might be able to get your settings arranged so that your XP laptop can handle the whole load.  If you haven't tried bi-amping using PEQ, I can describe my settings in detail.

BTW what kind of DACs are you using, and did you take any special steps to sync them to each other?  I never had any luck getting two separate USB DACs to sync up well enough.  I'm curious how you got it working (or whether there is hardware that will cheerfully do this without a word clock/ master slave relationship or similar).