Setting up a PC-based multichannel DSP system

[mimi123]

External software would be PowerDVD or the like......External to MC.

Very thanks for the clarification.

The loopback works by "grabbing" the audio stream sent to the Windows audio driver and processing it through MC's dsp. This way you can use MC to process any audio on your PC. No virtual driver used but I hear it's in the works.

Now I see it. Then we need a non used 7.1 sound card to send the audio of the external player and another card to play the sound from MC.
It's like this?

I've never tried min phase filters with convolution.....I use the peq for that but I'm sure latency issues would be more manageable. I played with the delay in media payer classic while using linear phase convolution but I could never get it right.

I thing this is the same problem that has ra7, every time you stop and play the sound the delay is different. Have you noticed this?

Regarding more channels, again, I haven't got to that point yet. But I believe the development team at miniDSP said that you can combine two USBStreamers to function as one soundcard under Windows.

This is very interesting.
Can you show to us how the usbstreamer is presented in Windows?
Is one device with 8 channels and another with 2 or is various stereo devices, or is a 10 channels device, or...?

The loopback through JRiver has been problematic for me. It works fine when playing music or sweeping an entire loudspeaker. But it has severe latency issues. Even with convolution disabled, its latency varies with time. For measurements, I ended up measuring the raw response and convolving the filter response in Holm. So, the steps are:
1. Measure raw driver response bypassing JRiver and USBStreamer altogether. Capture relative phase between drivers.
2. Import raw response in rePhase. Create filters. Export filter response from rePhase as .txt file.
3. Import filter into Holm.
4. Convolve to see the final response in Holm (C = A*B).
Then after you've got the amplitude and phase for all drivers lined up correctly in Holm, set up JRiver and do a final sweep with all drivers playing and see if it matches up. It usually does.

We know that JRiver guys are continuously working and improving their software. If it's possible, I'm sure they fix it.
Have you reported the problem to them?

 

[ra7]

This is what the USBStreamer control panel looks like. 10 channels, one device. Individual volume controls. It has 10 input channels also.

 

[natehansen66]

Now I see it. Then we need a non used 7.1 sound card to send the audio of the external player and another card to play the sound from MC.
It's like this?

Yep. The card that's used as the default Windows device doesn't need to be multi-channel, though most are.

I thing this is the same problem that has ra7, every time you stop and play the sound the delay is different. Have you noticed this?

I haven't dug into it at all so I don't know if the delay changes like ra7 is saying.....I suppose it could. My thought is that if you are using the same filter then the delay in theory should be the same from one moment to the next. If you use convolution and play a video within MC the software corrects for the latency of the filter. When playing video in external software and using MC's convolution there is no latency compensation so things are noticeably off.

 

[ra7]

Once the stream is started, the latency is fixed. So, if you start a movie, it will play with say 10 ms of delay. But then you come back tomorrow, and play another movie, there is no guarantee that the delay will again be 10 ms. It might be 11 ms, 12 ms, 5 ms... anything.

Maybe this is why you had problems with syncing the movie? You'd have to adjust delay everytime you played a new stream.

 

[natehansen66]

Ahh gotcha. That is strange. I would think it should stay the same as long as the filter hasn't changed.

 

[mimi123]

Yes, that's the problem for me too.
With one card it's a minor problem, but for two cards it's impossible to fix it if the two cards takes two different and random delays every time you play a movie. I think that the driver must to do it.

Windows is not a real time OS and is difficult to solve this.

 

[bohdan1232000]

Once the stream is started, the latency is fixed. So, if you start a movie, it will play with say 10 ms of delay. But then you come back tomorrow, and play another movie, there is no guarantee that the delay will again be 10 ms. It might be 11 ms, 12 ms, 5 ms... anything.

Maybe this is why you had problems with syncing the movie? You'd have to adjust delay everytime you played a new stream.

Hi Ra7,

How do you measure your latency?.

Best Regards,
Bohdan

 

[ra7]

I don't measure it at all... The precise value is not needed, just that it should be stable. As noted in the previous posts, I couldn't get around the JRiver loopback variable latency problem. I decided it's eaiser to do the crossover work outside JRiver. The minidsp 2x4 setup had me addicted to live measurement and tweaking of the crossover. So, this is bit of a step back, but not a big deal.

 

[DorinD]

Hello guys,

I'm using my USBStreamer in ADAT mode and got 8 channels out to an 4-way stereo system. For crossover duty I use AllocLite VST plugin-in jRiver and the system sounds pretty well and consistent. As a next step I am thinking to get a FIR of my room and load it into jRiver convolver.

Can I use --for example-- an HOLM generated impulse file for this purpose? Is anyone here already tried this? Or maybe have other ideea how to get as easy as possible the FIR filter of my room? I have no intention to spend hundreds on "Accurate" or something like this.

Thank you very much.

 

[natehansen66]

You can use any software for making measurements......you just need a way to build the filters. I'd recommend rePhase like ra7 is using here. Basically you're trying to do the same exact thing he is, no? He's even using Holm.

 

[DorinD]

I already have sorted out all crossover filters by Allocator Lite VST (if that's what you mean). I don't try to use FIR filters everywhere, it's kind of nonesense for me.

My intention is to obtain somehow a formatted/normalized impulse response of my audition room to have it loaded into jRiver's convolver engine --before the crossover-- and let it corrects for the general phase&amplitude.

 

[Soldermizer]

Will this work? (Cheapskate multi-channel I/O)

I want to explore other options for low- to no-cost multi-channel I/O (mainly "O") on a Windows 7,8, or 10 based PC (I will consider Linux as well but have zero audio experience with said rogue operating system )

Could this be done? Imagine a typical PC, even a super-cheap one like the Pipo X8: they [almost] always have built-in 2-channel analog outs. Now, can I add in addition, an external USB DAC (2 or more channels)? Of course. But how do I arrange the software for crossover and signal routing? This is where my brain gets confused

My signal source would be PC-based (e.g. WinAmp, Foobar, or J River or ...). I do NOT do video. I would like basic functionality at least like a MiniDSP 2x4. The projected use would be to drive a 2-way system (actually a 1-way "Bose 901" type for highs and LF for sub(s)). But if there is merit to my idea, one could add more external DACs for 6 channels, etc.

What are the recommened softwares to do this? Thanks.

 

[jplesset]

The software the already exists will drive any 5.1 or 7.1 USB external "sound card", using the various channels for your crossover outputs.

I've not experimented with doing this on Windows, but I understand JRiver will do this with pluginsl JRiver isn't free, nor is Windows. . .

It's not difficult to do this on Linux. That's where I went. I'm using and old PC with a sinle-core AMD processor, 1 gig of RAM, and it's a great plenty. My system rarely goes over 5% use, 95% idle.

The computer was free, of the, "Can you pull the data from this old thing and recycle it for me?" type. Ubuntu Linux is free, as is all the software in use. I paid about $20 fior the USB sound card off Amazon.

 

[arcgotic]

Check this, http://www.diyaudio.com/forums/pc-based/264307-software-crossover-equalizer.html
It is what you are looking for.

 

[diyaudnut]

Happy to have found this thread. Have just browsed thru it. But i will go ahead and post my question.

I am in the process of building a 4-way active speaker.
1. I am going to use a Fanless HTPC as my source + DSP XO/correction filters
2. I will build the filters using rePhase
3. Will be using convolution engine in JRiver
4. Multichannel DAC has following options
a. Asus Essence STXII 7.1 PCIE soundcard
b. MOTU USB audio interface
c. A dedicated ES9018 multichannel dac using MiniDSP USBstreamer as the multichannel USB interface to the PC
5. My measurement gear includes a EMM6 calibrated microphone, Focusrite 2i2 as USB soundcard with Mic preamp

A concern that i have been having and seems to be valid as indicated in this thread is a reliable measurement setup for the above active loudspeaker system.
Tools like ARTA seem to require input and output channels to be on the same soundcard in ASIO mode. In this case MOTU or the 7.1 ASUS soundcard having both ADC and DAC controlled through same driver and full duplex through the USB/PCIE port seem to be suitable.

But how to use the standalone USB MC DAC with Focusrite 2i2 for microphone? In this case one USB port/driver is for the measured response from the Mic preamp. A second USB port/driver is used to output the measurement sweep through the Jriver convolution engine (XO/filters) and out through the MC DAC -> MC amp -> drivers

Please advice if this can be made to accurately and reliable work to measure and develop the filters in the first place.

 

[DorinD]

@jojip
in case 24bit/48kHz is good to you I can suggest using USBStreamer with ADAT firmware and a professional dedicated ADAT DAC like this. Having inputs/outputs on the same interface will solve you also the ARTA issue. I used such a setup with IIR filters into a jRiver loaded VST (Allocator Light) with perfect results. I change this solution after a while just because I wanted a more flexible HTPC.

 

[diyaudnut]

@jojip
in case 24bit/48kHz is good to you I can suggest using USBStreamer with ADAT firmware and a professional dedicated ADAT DAC like this. Having inputs/outputs on the same interface will solve you also the ARTA issue. I used such a setup with IIR filters into a jRiver loaded VST (Allocator Light) with perfect results. I change this solution after a while just because I wanted a more flexible HTPC.

Thanks Dorin. My preference is a standalone high quality DAC. But i agree mic/dac on the same interface would be easier with measurement software

 

[DorinD]

I have reservation on DIY easy over performing any ADAT box from Alesis, Fostex, Motu or even Behringer (about 70 USD on eBay). USBStreamer driver also permits setting ADAT clock as master to get rid of jitter.

 

[natehansen66]

I use a studio multichannel firewire dac with JRiver, and do measurements with an M-Audio mic pre. This works just fine with ARTA, REW, and Holm. Time lock on Holm doesn't work with this setup, and if you're doing dual channel measurements in REW or ARTA there can be a bit of time drift unfortunately. ARTA and REW can both be used in single channel mode.

 

[diyaudnut]

Thanks Nate/Dorin,

I am very likely going ahead with http://www.diyaudio.com/forums/grou...kbird-multi-channel-es9018-dac-group-buy.html for my 8 channel DAC

Will stick with Focusrite 2i2 for measurement

This is a close contender MOTU.com - Overview for DAC.
Its line input might even allow me to connect the focusrite preamp output, for a single interface IO if i do run into issues.