# What software to use filters created with REW?



## Bayilokanto (Sep 29, 2009)

I am running a multichannel sound system straight from a Windows 7 computer (with ASIO installed) directly to an 7 channels Emotiva amplifier. The whole sound pre-amplication process is done within Windows (in my computer) and not through a sound processor. So any sound filter has only to be applied within Windows. My question now is: Is there any software (or pluggins) with multichannel capabilities that I can use in my computer to apply wave filters (obtained during room equalization) in order to reduce room effect on my whole system?


----------



## Zeitgeist (Apr 4, 2009)

I've seen some EQ/correction software for Windows..... but nothing fully multichannel, and any EQ they had seemed to be entirely proprietary.

I'm curious to know if there is any good multichannel EQ software for Windows. Perhaps the REW filters could be adapted.


----------



## Barleywater (Dec 11, 2011)

What are you using for a media player? Is this theater setup? Or active stereo?

Key to implementing filters is convolution. Convolver works with Windows Media Player, and has VST version that works with host such as Console, which works with ASIO. JRiver Media Center hosts VST, and has own convolution engine. Foobar has something too, but I don't know if it is multichannel.

Regards,

Andrew


----------



## fusseli (May 1, 2007)

Yes, "doing" filtering is simply convolving filter coefficients with raw data.

I have heard of some software but it's escaping me at the moment. For whatever reason the idea doesn't seem like it's caught on much which is kind of surprising.

In all honesty it seems like it should be really simple for someone to write a program that does this. Unfortunately I have no knowledge of how to take control of hardware in windows and make it do something like that. I also have no idea what multichannel limitations there might be. Logic tells me that if you can filter one channel, you can filter seven others. It will all come down to CPU speed and filter order at that point.


----------



## AudiocRaver (Jun 6, 2012)

Bayilokanto:

One option is to use a DAW - Digital Audio Workstation - as your EQ engine, following your media player within your Windows PC. I use Reaper for this, which costs $60 for a non-commercial license, includes convolution and parametric EQ plugins (unlimited # of bands for the parametric), will accept pretty much any available VST plugin (unlimited quantities, combinations, instances) - there are free and inexpensive VST plugin options galore for equalization, bass management, signal analysis and display. And multichannel capability is unlimited - 24.6 anyone? Reaper itself is not a memory or cpu hog, but your ultimate configuration will determine how much horsepower you need. And it is extremely stable and well-supported, including a helpful user community and pretty-good documentation.

The downside - the rascal is quite complex, as is the nature of DAWs in general, and it could take awhile to get your arms around. Its target audience is the professional recording scene, but it is implemented in a stripped-down, build-it-up-however-you-want-it modular fashion (think of an audio erector set) that has led to it showing up in all kinds of odd audio corners.

Reaper works best with ASIO input & out drivers, but can work with Direct Show or pretty much anything else I/O-wise. I use foobar2000 with Reaper, but other media player options exist with ASIO capability. The complexity question is a big one for you to consider. If you enjoy digging into into something new and different like that, you might end up appreciating its elegance and become a fan like me. If you just want something that works so you can move on to other matters, than the DAW/Reaper route is probably not the way to go.

If it helps, I'd gladly share Reaper project files with you as a starting point for you to modify at will, so you don't have to figure out EVERY little thing about Reaper from scratch. Plan on it taking some time either way.

Anyhow, there's an option for you to consider.

AudiocRaver


----------



## Bayilokanto (Sep 29, 2009)

Thank you for your well informed perspective on my post. I am not intimidated by complexity and will adopt your advice. It is exactly what I have been looking for. I will install it as soon as I get back home tonight and play with it a little bit. Do you have any idea how to make it work with Windows Media Center (WMC) or Windows Media Player? Does it connect to WMC through sound plugins? Does it show up on Windows System as an additional sound output device that I can control through the Control Panel.


----------



## AudiocRaver (Jun 6, 2012)

Bayilokanto said:


> Thank you for your well informed perspective on my post. I am not intimidated by complexity and will adopt your advice. It is exactly what I have been looking for. I will install it as soon as I get back home tonight and play with it a little bit. Do you have any idea how to make it work with Windows Media Center (WMC) or Windows Media Player? Does it connect to WMC through sound plugins? Does it show up on Windows System as an additional sound output device that I can control through the Control Panel.


As an application, Reaper will not show up in your Windows Sound control panel as a recording or playback device, just like Windows Media Player does not. So your first challenge is getting signal flowing between applications, which Windows doesn't help you with directly. Virtual Audio Cable (VAC) or Virtual Audio Stream (VAS) can do it. I haven't used VAC, but it sounds very capable, handling different bit depths and rates and providing advanced features. VAS is simpler, as of my last installation it only handles 16bits @ 44.1KHz, but works great if that's all you need.

I'm going to start a new thread for using Reaper as an audio EQ engine, but other obligations won't allow me to put much time into it for a few days. I know I promised help - you very well might not even need it - and there are numerous configuration tips for using Reaper that I have been meaning to write up and share, and where better than our beloved Shack? - but it will be a few days...

HOWEVER... A couple of quick things that might keep you from spinning your wheels in the mean time...

If you get into the land of ASIO drivers, there's another option. Reaper can be installed (this has to be activated at installation, but if you missed it, just re-install) with a feature called ReaRoute - it is the _only_ install option box not checked by default, so dig for it. This creates 16 ASIO inputs and 16 ASIO outputs available to any application on your PC, as long as both Reaper and the other application(s) have ASIO selected as their I/O method. I'm not aware that WMP works with ASIO, but foobar and JRiver do, and there are others.

One installation quirk. It seems you have to install both the 32-bit and 64-bit versions of Reaper to get the ReaRoute feature to show up in 64-bit Reaper, no idea why. Both installations are small, so no big deal. Then mind your shortcuts so you're always starting the version you want. If you install 32-bit first, then 64-bit, I THINK the 64-bit version will be the default started from a project shortcut. I _think._

****
A foobar2000 note, if you consider going that way: its ASIO driver is not perfect. When you hit *stop* or *pause* while playing a track, it stops, then a fraction of a second later there is one more little bleep of sound from that track, and THEN it stops properly. A word of warning to the perfectionists. I just tuned it out long ago. Other than that, I really like foobar. Hooking it up to ReaRoute is a little tricky, and that will be my first writeup.
****

And you'll need ASIO4ALL installed if you go the ASIO route.

And be sure you download the Reaper Quick Start guide, it will save you hours. I stubbornly avoided it at first, and it was not pretty:crying:

That should get you started. I will reference the new thread here when I get it going.

Have fun with Reaper.

AudiocRaver


----------



## Bayilokanto (Sep 29, 2009)

Thanks very much for your advice. My system is based on Windows Media Center (WMC). I have found this ASIO plugin that works with Windows Media Player as well as WMC: asiowmpplg_sourceforge_net (replace "_" with " ."). I used it once and worked fine, but I had hard time trying to adjust the volume through Windows which became impossible since everything was going through ASIO at a maximum power. I ended up removing it for lack of volume control. I will again give it a second chance with your well advised settings. I will keep you informed about everything.


----------



## EarlK (Jan 1, 2010)

Hi,


- I like AudiocRaver's suggestion of using *REAPER*  ( as a host for VST based EQ files ) . It has a nice GUI , something sadly lacking in most of the important ( to me ) DSP areas within MC17 .

- I look forward to your thread on how to make this work with WMC or WMP ( though, it is a cinch with jRiver's MC17 ) .

As with all such things , the devil is in the details . A few speed-bumps that one must overcome ;

(i) *ReaRoute*  ( the ASIO driver in REAPER ) needs to "communicate" with the media players via ASIO , therefore those players must support an ASIO output . 
- adding the DSP plugin called  *ASIOwmpplg*  nicely solves this problem for the WMC &/or WMP players .

(ii) *ReaRoute*  must be set to use the same sample rate ( & bit depth I do believe ) as the file being played by / or else WMP ( for one ), throws up a caution flag just before crashing .

- one ( pretty lousy ) solution is to match the sample rates between the media player and the ReaRoute ASIO driver , and then make sure that ones library only contains that one file format ( not very workable IMO ) .

- another solution is to resample all files as they are played back. I looked around the net , but couldn't find an on-the-fly resampler that was a dsp plugin . FWIW, MC17 will resample all audio files on the fly to whatever sampling rate ( & bit depth ) your hardware will support ( or what the DAW is set to ) .

- perhaps the solution is to get WMC or WMP to utilize  ffdshow-tryouts  as the overall Codec so that one might use it's on-the-fly resampling . I tried this but got stone-walled by my on ignorance of this codec. :unbelievable:

(iii) Another challenge is to get all these apps working together without hearing stutters, ticks, etc. .
- That means employing the dark art of buffering between all the programs being used ( & their audio drivers ).
- If the media players lack adjustable buffers, then one is at a real disadvantage here . 

:sn:


----------



## AudiocRaver (Jun 6, 2012)

EarlK said:


> (ii) *ReaRoute*  must be set to use the same sample rate ( & bit depth I do believe ) as the file being played by / or else WMP ( for one ), throws up a caution flag just before crashing .


Here's how I handle sample rates. Reaper allows for a default sample rate at startup (I use 48KHz), and then a separate project sample rate. A project then, or in our case, an "EQ Settings Project," will consist of the desired plugins, channel configuration, input and output choices, sample rate, convolution impulses, EQ settings - whatever is desired for a given setup. I have one at 44.1 with EQ for my Polk speakers and the foobar2000 source, another at 44.1 with EQ for another set of speakers with foobar, another at 48KHz with flat EQ for headphones and inputs from the audio interface for external sources -- you get the idea. Each project represents a completely different configuration. Then...
.
Reaper allows you to have as many projects as you want all open and "tabbed," with one being active. To switch from one to another, a single mouse click on the appropriate project tab in Reaper switches the entire configuration instantly, including sample rate (Reaper has taken exclusive control of the audio interface's sample rate and switches it automatically). And foobar follows the Reaper sample rate.

I keep my media files segregated by sample rate, only a slight inconvenience for me. So I stop foobar, load up a playlist of tunes with at a different sample rate, click on the Reaper project for that sample rate, hit play, and away it goes.

It's not perfect, but it's not that bad, either, for me anyway. I have no idea how one might accomplish automatic switching between sample rates as the media player pulls up a file with a different rate, if that is desired. If that is a priority (it WOULD be really cool), and if someone has a suggestion how to try to make it work, I'll sure pitch in and help where I can. A programmer I am NOT, but I do get lucky once in awhile fiddling with configurations and utilities and drivers and such.

Oh, and if you try to play a 48K file in foobar while Reaper is at 44.1K, foobar throws up an error message, there is no big crash or anything horrible. I realize that other media players may have issues with this. Sample depth does not appear to be an issue. I have Reaper set at 24 bits all the time and it handles the normal 16 bit stream from foobar without appearing to even notice the disparity.

Hope that helps a little,

AudiocRaver


----------



## Mitchco (Apr 12, 2011)

You might want to consider the new version of JRiver MC 18 and its 64-bit Convolution engine. One of it's new features is automatic filter switching based on source sample rate. JRiver also has an excellent resampler.

There is a loopback feature in JRiver 18 that makes its audio and convolution engine (plus host to other VST's) available to any audio source, essentially making it like a standalone audio playback engine.

Another new feature is the convolution engine adapts the target latency. So I can use JRiver to host FIR filters, and using the loopback feature, I can run REW to perform before and after filter acoustic measurements, running the sweeps through JRiver.

I run a simple stereo setup using Audiolense to generate FIR filters hosted in JRiver's Convolution engine. However, others on this thread have tri-amped, 7.1 systems with subs and using JRiver's convolution engine successfully: http://yabb.jriver.com/interact/index.php?topic=68828.0

I don't work for JRiver and a fan of Reaper as well. Reaper and will do what you desire as decribed by AudiocRaver. The new MC 18 version of JRiver/Convolution approach may be simpler...

Cheers, Mitch


----------



## AudiocRaver (Jun 6, 2012)

Mitchco said:


> You might want to consider the new version of JRiver MC 18 and its 64-bit Convolution engine. One of it's new features is automatic filter switching based on source sample rate. JRiver also has an excellent resampler.
> 
> There is a loopback feature in JRiver 18 that makes its audio and convolution engine (plus host to other VST's) available to any audio source, essentially making it like a standalone audio playback engine.
> 
> ...


Thanks Mitchco, it does sound like JRiver is rapidly developing into quite a versatile audio tool and for many users may be better suited for their EQ tasks than Reaper. I've been curious about it but haven't even tried it yet - sounds like it might be time to take the plunge. Appreciate having the additional option to consider.

AudiocRaver


----------



## Bayilokanto (Sep 29, 2009)

Thank you Mitchco for your perspective in using a JRiver MC 18 as an alternative. I have two systems (one in my living room and another one in my basement). I will experiment both of those approaches on my systems and evaluate the quality of the final result and also the easiness of implementing them. As of now I am having problems setting up effectively Reaper through ASIO and also having Windows Media Center to communicate with Reaper through Reroute ASIO. I am not in the hurry and will get to the bottom of this.


----------



## mojave (Dec 30, 2006)

JRiver also has had a powerful Parametric Equalizer for the past couple of years. I've used it at a couple of subwoofer GTG's to EQ the subwoofers and I use it in my own system. Actually, I started using convolution a couple weeks ago and, like Mitchco, use JRiver's convolution engine.

I've had a DCX2496 and miniDSP and JRiver is much more flexible and the perfect solution for someone that wants advanced EQ. It has parametric filter, high and low pass filters, high and low shelf filters, a Linkwitz Tranform for sealed subs, and a subwoofer limiter. You can also copy bass to other channels. For example, I use a phantom center in JRiver and then copy the subwoofer to the center channel. Each channel outputs to one amp and infinite baffle manifold. The Room Correction DSP handles all the delay, distance, crossovers, and crossover slopes. You can use a 24 db/octave crossover for the speaker and a 48 db/octave crossover for the sub.

You can created zones in JRiver for different content (2 channel, multi-channel, and movies). This allows you to different settings depending on what you want for various content. In JRiver V18, they are planning on making the zone selection automatic based on your criteria.


----------



## AudiocRaver (Jun 6, 2012)

Sounds like JRiver rules! I'm definitely going to give it a try.

How flexible is the convolution engine in terms of being able to adjust convolution delay time? I like using linear-phase EQ in an audio-only setup - the processing delays with linear-phase EQ are usually too long to stay in sync in an audio-video situation. But some convolution engines allow you to set the delay time, number of samples or whatever, if your cpu can handle it. Does JRiver's convolution engine allow that option? If so, how short can it go?


----------



## mojave (Dec 30, 2006)

JRiver's convolution engine automatically keeps the audio and video in sync regardless of delays that are part of the convolution. You can also manually adjust the audio/video sync separately from the convolution engine. You can specify delays for 24 Hz video output and 50/60/120 Hz video output in 1 millisecond increments. 

You have a 30 day trial with JRiver, but if you also want a demo on a working system you can come on over sometime. I have GR-Research LS-6 line source speakers and an infinite baffle subwoofer system with eight 15" Acoustic Elegance IB15 drivers.

JRiver also now supports automatic sample rate switching for Audiolense convolution filters. This means if the source has a sample rate of 44.1 kHz, then it will use the 44.1 kHz filter and if the source is 96 kHz then it will automatically use the 96 kHz filter. They will add support for other filters if users provide them with the naming details of the files.


----------



## AudiocRaver (Jun 6, 2012)

mojave said:


> I have GR-Research LS-6 line source speakers and an infinite baffle subwoofer system with eight 15" Acoustic Elegance IB15 drivers.


mojave,

I was just rescanning your earlier post. It had not sunk in before that your sub system has 8 15-inch inch drivers. Zowie! So that's the source of the rumbling we feel over here in Lincoln when there are no storms around!


----------

