Is there a way to host a VST plugin between the DAW and ASIO outputs?

[elmagoo]

Hello,

I did a search on GS but couldn't find this one answered (someone did ask the same question, but instead of answering it, they were told why they shouldn't use the plugin...

So I have the IK Multimedia ARC system, and it's designed to work as a plugin you put at the end of your signal chain inside your DAW. What I wanted to know is if there's a way to host a VST plugin within Windows 7 before your ASIO outputs that you can route other programs through? Basically I want everything the computer outpus to go through ARC so I can do reference mixes, OS played sounds, as well as hear my direct monitoring through it.

I'm using an RME RayDat with the Hammerfall DSP software. I have all of my OS related sounds going to one set of ASIO streams, and my DAW outputs specifically going to another (so I can balance the volume between them).

My currenty work around is I found a VST plugin host plugin for Winamp. So I can play my reference mixes in Winamp through ARC, but this doesn't solve the problem of hearing the direct monitoring sources (like my mic'd guitar cabinet).

Any idears??

Benson

[sickie]

Yo!
You could try VSThost; I don't know if it can take audio inputs though, but probably it does. Just rout everything you need trough it with TotalMix (loopback/hardware inputs). If it doesn't take audio inputs you can try with reaper (the DAW) as it's quite lightweight.

[electro]

Elevayta - Stream Boy

[pedrogoya]

Hi benson!

Try Soundflower, although im not sure if there's a windows version.

Hope it helps

[Hardtoe]

Buy the KRK Ergo instead.

[sickie]

With RME's totalmix he's able to move streams between hosts with loopback functionality, so he doesn't need such applications as streamboy.

[elmagoo]

Thanks for the posts all!!!!!

I'll have to investigate this loopback functionality in TotalMix as I've never had a need to use it, sounds promising .

Also I did try the KRK Ergo as I felt that would've been the ideal solution, but it acted REALLY wonky. I ran the setup routines with it and the output was suffering from really bad phasing problems. I ran the mic test like 5 times, both via digital input and the analog inputs, all yielded the same problems. The ARC software worked right the first time, so I took the Ergo back. :(

[Hardtoe]

Originally Posted by elmagoo

Thanks for the posts all!!!!!

I'll have to investigate this loopback functionality in TotalMix as I've never had a need to use it, sounds promising .

Also I did try the KRK Ergo as I felt that would've been the ideal solution, but it acted REALLY wonky. I ran the setup routines with it and the output was suffering from really bad phasing problems. I ran the mic test like 5 times, both via digital input and the analog inputs, all yielded the same problems. The ARC software worked right the first time, so I took the Ergo back. :(

Too bad - mine has worked as advertised from the start....

[elmagoo]

Originally Posted by Hardtoe

Too bad - mine has worked as advertised from the start....

Yeah I know, I was really wanting it to work out. I'm guessing it was user error on my part probably, but I thought I followed the instructions accordingly. Eh....it allowed me to upgrade my monitors to Adam A7x's by returning it .

[elmagoo]

Just wanted to give a follow up to how I solved the problem!! Thanks again for all the replies as it definitely pointed me in the right direction

Alas I wasn't able to find a loopback option inside of my version of Totalmix as there seems to be a difference between Totalmix and Totalmix FX. The FX version being the one that comes with the newer interfaces and actually has a nice GUI. Old school Totalmix that comes with HDSPe cards (like my RayDAT) I couldn't find a loopback feature. Also I'm not sure if it would do what I wanted as it seems the loopback is to put the audio back into your DAW, rather than routing it back into an actual physical input, which is what I needed in my case. At least this is my understanding of how I needed to get things to work, so I could be wrong about that bit.

So to give a background on my setup (I hope this isn't too confusing , which I'll admit might be a bit unique, I'm using a Liquid Saffire 56 as an out-board AD / DA converter + mic pre's (which I highly recommend to anybody on a budget as you get 2 liquid pre's, and excellent quality converters for around $1k), and the RayDAT as my interface card into the computer. Hence I only use the Saffire Mix program to setup routings and Liquid Pre's, then I save it into the hardware and disconnect the FW so it's purely running in stand alone mode. I have the 8 mic pre's on the LS routed to the 8 ADAT outs of the first port, and the 1st ADAT port inputs routed directly to the LS physical outputs 3 - 10 (I use physical outputs 1 and 2 for the actual monitors, and I have the SPDIF input routed to them). The 2nd ADAT ports are hooked up to the RayDAT (LS ADAT Outputs to RayDAT Inputs, and RayDAT Outputs to LS Inputs), and internally in the LS I have all of the 2nd ADAT inputs routing directly back to the 2nd ADAT port outputs. Basically, THESE are my loopbacks in physical form .

So what I do is in Totalmix, I use one of these sets of ADAT ports as my master out (what everything's routed to). Specifically RayDAT output 2.1 and 2.2 (second ADAT port on the RayDAT, channels 1 and 2). These feed into the LS ADAT inputs 2.1 and 2.2, which are then routed internally to LS ADAT outputs 2.1 and 2.2. This then goes back into the RayDAT on ADAT Inputs 2.1 and 2.2. So I have all of my sources now looping physically through ADAT cable, but still remaining digital with no extra conversion steps, back to a RayDAT input. I believe I could also just theoretically hook ADAT cables to loop the RayDAT input / output ports as well without going through the LS, but since everthing is wired into my desk, and it's all remaining digital with no additional conversions, I didn't feel like ripping apart cables and spelunking

Now I just needed something like VST Host to feed off of the new inputs 2.1 and 2.2 to run all of the audio through. I'm sure VST Host could do this, but I found the interface very confusing, so rather than mucking with it, I downloaded a different program called Live Professor. Currently in beta form and free, but this software is designed to specifically be a very low CPU usage VST effects rack for live usage. Hence I just told it to feed off of inputs 2.1 and 2.2, and then had it output to a separate ASIO stream that's not used by anything else. I then take my SPDIF physical outputs on my RayDAT, and route the dedicated ASIO stream coming from Live Professor (with nothing else routed to it) which then goes to my Mackie Big Knob...then the speakers.

This works like a charm, I just have to have Live Professor running in the background always to hear anything . But all of my signals go through it, including my direct monitored signals, anything I play in the operating system, or from my DAW.

So to boil all this down to simplistic terms , I used a discrete physical output as my new master out and put all signals to it. I then routed that back to a dedicated input. Used the Live Professor program to feed that new input through VST plugins of choice. Then route the output of the program to my actual speaker out.

The really cool thing about this is now I can run not just the ARC program, but also my analysis plugins that I use for mixing as well (Voxengo SPAN in this case, and some iZotope stuff for extra metering). This way I can see the analysis of my incoming signals as well as reference material played back through the OS. Everything gets the ARC room correction.

Now if audio interface manufacturers would just "get it" and put the ability to host VST (or AU or RTAS) plugins directly in their software mixers it would save a lot of hackery head aches and such!!!

[electro]

loopback would not accomodate clipping over 0dbfs and 32bit float

Originally Posted by elmagoo

Just wanted to give a follow up to how I solved the problem!! Thanks again for all the replies as it definitely pointed me in the right direction

Alas I wasn't able to find a loopback option inside of my version of Totalmix as there seems to be a difference between Totalmix and Totalmix FX. The FX version being the one that comes with the newer interfaces and actually has a nice GUI. Old school Totalmix that comes with HDSPe cards (like my RayDAT) I couldn't find a loopback feature. Also I'm not sure if it would do what I wanted as it seems the loopback is to put the audio back into your DAW, rather than routing it back into an actual physical input, which is what I needed in my case. At least this is my understanding of how I needed to get things to work, so I could be wrong about that bit.

So to give a background on my setup (I hope this isn't too confusing , which I'll admit might be a bit unique, I'm using a Liquid Saffire 56 as an out-board AD / DA converter + mic pre's (which I highly recommend to anybody on a budget as you get 2 liquid pre's, and excellent quality converters for around $1k), and the RayDAT as my interface card into the computer. Hence I only use the Saffire Mix program to setup routings and Liquid Pre's, then I save it into the hardware and disconnect the FW so it's purely running in stand alone mode. I have the 8 mic pre's on the LS routed to the 8 ADAT outs of the first port, and the 1st ADAT port inputs routed directly to the LS physical outputs 3 - 10 (I use physical outputs 1 and 2 for the actual monitors, and I have the SPDIF input routed to them). The 2nd ADAT ports are hooked up to the RayDAT (LS ADAT Outputs to RayDAT Inputs, and RayDAT Outputs to LS Inputs), and internally in the LS I have all of the 2nd ADAT inputs routing directly back to the 2nd ADAT port outputs. Basically, THESE are my loopbacks in physical form .

So what I do is in Totalmix, I use one of these sets of ADAT ports as my master out (what everything's routed to). Specifically RayDAT output 2.1 and 2.2 (second ADAT port on the RayDAT, channels 1 and 2). These feed into the LS ADAT inputs 2.1 and 2.2, which are then routed internally to LS ADAT outputs 2.1 and 2.2. This then goes back into the RayDAT on ADAT Inputs 2.1 and 2.2. So I have all of my sources now looping physically through ADAT cable, but still remaining digital with no extra conversion steps, back to a RayDAT input. I believe I could also just theoretically hook ADAT cables to loop the RayDAT input / output ports as well without going through the LS, but since everthing is wired into my desk, and it's all remaining digital with no additional conversions, I didn't feel like ripping apart cables and spelunking

Now I just needed something like VST Host to feed off of the new inputs 2.1 and 2.2 to run all of the audio through. I'm sure VST Host could do this, but I found the interface very confusing, so rather than mucking with it, I downloaded a different program called Live Professor. Currently in beta form and free, but this software is designed to specifically be a very low CPU usage VST effects rack for live usage. Hence I just told it to feed off of inputs 2.1 and 2.2, and then had it output to a separate ASIO stream that's not used by anything else. I then take my SPDIF physical outputs on my RayDAT, and route the dedicated ASIO stream coming from Live Professor (with nothing else routed to it) which then goes to my Mackie Big Knob...then the speakers.

This works like a charm, I just have to have Live Professor running in the background always to hear anything . But all of my signals go through it, including my direct monitored signals, anything I play in the operating system, or from my DAW.

So to boil all this down to simplistic terms , I used a discrete physical output as my new master out and put all signals to it. I then routed that back to a dedicated input. Used the Live Professor program to feed that new input through VST plugins of choice. Then route the output of the program to my actual speaker out.

The really cool thing about this is now I can run not just the ARC program, but also my analysis plugins that I use for mixing as well (Voxengo SPAN in this case, and some iZotope stuff for extra metering). This way I can see the analysis of my incoming signals as well as reference material played back through the OS. Everything gets the ARC room correction.

Now if audio interface manufacturers would just "get it" and put the ability to host VST (or AU or RTAS) plugins directly in their software mixers it would save a lot of hackery head aches and such!!!

[elmagoo]

Originally Posted by electro

loopback would not accomodate clipping over 0dbfs and 32bit float

Yes definitely, but that shouldn't matter at this point anyways. I'm not actually using any of the Live Professor plugins to modify the source for printing, that's all done before hand. I'm only using it to modify the source for room treatment, and also so I can monitor the frequency spectrum in one place (setup pre-ARC in Live Professor).

As for clipping, again it shouldn't be a problem unless you're sending a signal too hot from your DAW (which in that case is going to cause problems regardless of this setup). Whatever signal you put through the ADAT pipe will be the same (i.e. you send a -1 DBFS signal out through the ADAT ports, it's coming back as -1 DBFS). Same for the OS sounds as you have to watch your output volume either way.

The only compensation I had to do was in the ARC plugin since the processing effects the signal. But this is all internal math inside the computer based on how Live Processor handles the VST signal (i.e. it's not going through any ports yet, it's all on the ASIO bus at this point). I only needed to worry about the DA converters, so I took the master output to my speakers and lowered it by whatever was needed to make sure a clipped signal wasn't being sent.

I guess one possible concern based on what you're saying would be does the ARC plugin lose anything being post DAW and filtered down to 24 bits as opposed to being within the DAW running at 32 bits. I did an A/B comparison where I would activate ARC within the DAW, then shut it off and activate it in Live Professor and I couldn't hear any discernible difference. But again, I'm not running any signal out of my daw over 0 DBFS, so it's not exceeding the 24bit resolution anyways.

[tonemaker]

I didn't understand some of the solution, but I was going to suggest Foobar2000 as a media player that supports an add-on component that hosts VST effects.

Also, why not use a DAW instead and just put VST's on the master out?

[elmagoo]

Originally Posted by astroidmist

I didn't understand some of the solution, but I was going to suggest Foobar2000 as a media player that supports an add-on component that hosts VST effects.

Also, why not use a DAW instead and just put VST's on the master out?

That's what I was doing before, and that works just fine. The issue is when you want to listen to reference material to compare your mix to. You want the same room correction that ARC applies to apply to the reference track, else it's an apples to oranges comparison. So you'd have to import all of the tracks you want to reference into your DAW, cluttering up your project and bloating its size. This method bypasses that extra step. Also, the same goes for setting up a sound to track. I use direct monitoring, so it doesn't go through the DAW, which means it doesn't get the ARC plugin if I only use the DAW.

I use Winamp as my media player, and I did find a VST plugin for Winamp, which got me part of the way there (at least for reference material). The thing is being able to play the direct monitored signal as well. Also, with Live Professor, I can create a rack of effects. So currently I put Voxengo SPAN, and then the ARC plugin after it. The beauty of this is now I can use SPAN to check out the reference mixes I want to listen to, and I can also inspect the incoming direct monitored signal as well to see if there's anything funky with it.

So far I'm loving this method, it just made the workflow much easier (although the initial setup was a pain to fully think out .