All-pass filters can be a very helpful tool to get two speakers to work together, especially when they are from different families or brands. Unfortunately, they are normally only available on higher end output processors and amps like Meyer Sound Galileo, Lap Gruppen PLM series, or Lake.
There is a free solution, though, which I’ll be testing out in this post.
The Setup
After my interview with Michael Curtis I’ve been looking at Reaper more and more. For these tests I’ll use the ReaEQ that comes with it.
Although Reaper isn’t technically free, it does come with a 60-day evaluation license. I did do some research into other available VST plugin hosts, but haven’t tested them, yet, so I’ll save that for another post.
A few years ago I published a video called The Poor Man’s Galileo. You can see that I’ve been interested in replacing various parts of my hardware setup with software for years. I wish I was a flute player. I love gear, but I hate how heavy and bulky it all gets. I love tools, but it’s hard to justify carrying them everywhere if I I’m not that I’ll actually need them. I always have my computer, though, which makes the software solution attractive.
I set up two computers with Reaper. One to host the plugin and proces the audio and the other to record. I created two record channels. One that was a simple loop from out to in so that it would experience the same A to D process, effectively removing it from the measurement. The other went to the OCTA-CAPTURE for live processing.
I ran a sine tone for verification and discovered that there was no block size too low. With only the ReaEQ inserted I was unable to interrupt processing with any errors. Once I dropped below 29 samples, though, Reaper stopped reporting a lower latency. In a later test, when I tried inserting an Ozone 9, I had to bring the block size back up to 32 to perform without any audible hiccups.
Latency
The main thing I wanted to confirm here is that the latency would be low enough for live use. How low is low enough? That depends on everything else in the system and the application. For an IEM monitor rig, 20ms might be too much after other possible latencies in the system, but probably wouldn’t make that much different for a big outdoor gig.
I guess the big question is whether the benefit is worth the expense. If you making a big improvement to a two octave wide crossover region and the latency seems acceptable, it’s probably worth it.
With no plugins inserted I measured a round trip latency of 5ms for my test setup. It stayed the same after inserting the ReaEQ. Just to confirm that I would see a change, I tried inserting an Ozone 9 with 3 modules and the latency jumped up to 75ms.
Conclusion: I would be happy to use this setup as a backup solution in the field. If there’s an internet connection or cellular data connection, I can get Reaper installed on any machine in a few minutes.
Have you tried live processing with plugins system calibration in the field? Let me know in the comments below.
Hello Nathan
first of all thanks for what you do , a ton of good informations on your channel.
Can you explain a little more in detail how can i set up reaper for use it as an “all pass filter” in live processing?
would be very appreciated
thanks
Luigi
Hi Luigi, thanks for reaching out. It’s been a while since I wrote this, but the it should be pretty straight forward. Route your IO through Reaper and insert the filter. How you route the IO will vary on your setup. Could be through Dante Virtual Soundcard. Could be through your USB audio interface. Maybe give it a shot and let me know what challenges you run into?
Hi Nathan thx for the quick reply
here i have an old presonus firestudio as my soundcard connected via firewire to a macmini (late 2012).
so let’s say i need ad allpass on the delay line of my main L/R… i have to send my matrix (delay line) from my console (x32 compact) to the input of my sound card and then in reaper i have to create two inputs and route them to the outputs (let’s say i only need two outputs) and insert ReaEq on each output.
would this be a good starting point or am i totally wrong ?
Yes, exactly. Some way to route IO through ReaEq. If you have X32 USB card you could also pass audio that way.
really really thanks Nathan
i will try this as soon as possible :)
just another question (out of this topic)
in wich situation would you place the microphone on the ground for measurement?
just sub+top allignement?
or would you lpace it on the ground even when delaing speakers?
thx a lot
Luigi
hmmm, I can’t think of a common reason that I would ever always go to the ground. I typically try to always measure at head height to capture the sound where the people are. Then, if data is not actionable because of the floor bounce, I might try a ground plane measurement.
What about you?
thx again Nathan it’s a pleasure to talk with you.
Some years ago i was advised to use the ground plane measurement for allignement stuff and the measure at head height for the EQ of the system… can’t remember , maybe i just read it somewhere over the net.
Anyway i saw many engineers going with the head height method so you are probably right :)
in these days i’m taking into consideration to use the FTW function in Smaart 9 at least for the xover design of my system wich is a 3way+sub .
Do you think this would be a good approach in desingin a xover ?
If not what method would you use?
Sure. Ground plane is a solid approach, but it has its place.
I haven’t played around with FTW much. I never found it very helpful, but I never really tried to use it. That being said, I’m in favor of any tools or features that help us the see the data in some new way. If some additional smoothing and attenuation of late arriving reflections is helping you, that’s great!
yes i think FTW would be a great tool but just for xover desing .. i would never use in live situations.
anyway i still need a lot of practice on the field to understand how to get the best data reading .
thanks for your patience Nathan
you are the man
Luigi
Great! If you are designing crossovers, does that mean you are building speakers?
yes ! i’m usually involved in small events where i have full range speakers (main+delay)+ subs .
i don’t have so much experience so i’m trying to improve , that’s why i’m asking to you some good advices. :)
But.. i built a sound system years ago (it’s a 3way+sub) and i’m always searching for the best way to allign the several components… the problem is that when i have the system stacked , it’s not easy to get a good data reading and so i don’t ever know if i’m taking the right decision when delayng high – high mid – low mid and the sub under them.
That said i started to think that a good starting point would be to simulate an anechoic chamber at home through the FTW and allign the three way above the sub… then one in the field i can treat the 3way as a full range speaker and more easily allign them with the sub.
This would save me a lot of time
Ah, ok, that makes sense. Please take a look at the Tracebook Measurement Procedure. It describes a method for taking ground plane measurements at home, in lieu of an anechoic chamber.
By the way, if you want to use your speakers with SubAligner, just send me your measurements and I’ll add them.
Tracebook:
https://drive.google.com/file/d/13OwwvV1-rYZu__gscDj53vyQ3KRk_X72/view?usp=sharing
https://www.youtube.com/playlist?list=PLsfnoBcHUm222UrwdC6zXS2cA-KQCcOQ1
Great !
really thx for the link finally i have a guide line to follow and to put my trust in.
In regard to SubAligner i don’t have any measurement to send you right now but as soon as i get the time to do some measurement i will send them to you.
What do you exactly need ?
the measurement of each way and then the combined response of the entire system?
These videos will explain: https://www.youtube.com/playlist?list=PLsfnoBcHUm222UrwdC6zXS2cA-KQCcOQ1