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[James Duncan]

I am hoping to find something in this market that has at least 3 inputs so I can run stereo to the tops and then run an aux-fed sub mix into the 3rd input.

It appears that the Behringer has this unit, but when looking closer, the 3rd input is the one that is used for the reference mic to do the auto-aligning function. I am not sure if this is a valuable feature or not, but it sure would have been more valuable (to me, at least) if there was a dedicated reference mic input.

Plus... "it is a Behringer."

Sorry folks, but I have had such bad experiences with Behringer in the past. I know folks are saying that this piece does not sound bad, etc., but you are still getting low-end components like power supplies and cheap caps on the boards. They may sound fine, but I am more worried about the long-term reliability. In a live rig, this is even more important.

Those are 2 strikes in my mind (no ref mic input and perceived reliability).

The Peavey also scares me from past experiences with the brand, but it also is getting some pretty good reviews around here. The soon-to-be-released VSX 48 looks promising as it has 4 ins and 8 outs...

As an alternative, the Sabine Nav3600 (3-in, 6-out) has been recommended to me from a few folks around here, and retails for about $1,000.

[E. Lee Dickinson]

Input C can be routed anywhere you want. The other nice thing the Behringer does is treat the Sum input (selectable among A, B, and C) as though it were a separate input. That is to say, if you adjust the input gain on A, it does not affect SumAB. We have the Behringer in our B rig.

I currently have a DR260 in each of my amp racks on the A rig (left rack, right rack) which allows me to run aux-fed subs, since I'm only using one input (L or R, respectively) for program and the other for sub. 3 inputs would let me split the system in half much easier, so I've been thinking of replacing with the Sabine Nav360 (which also has better remote control options) or, for cost reasons, the Behringer.

[John Roberts {JR}]

Josh Billings wrote on Fri, 13 April 2007 15:41

The difference isn't in the sample rate as it is with all the stuff you are doing with the signal. In the studio side of things you sample at 24/96 to get a super clean sample to really mess around with (compress/EQ/limit/etc) The more you mess with that signal the higher that sample rate helps you in the end.

There are several conditions for archival studio recording that don't apply to live SR. If you need to later convert to a different sample rate you can down convert which is discarding data more accurately than up converting which is essentially inventing data points between real samples.

The increased resolution in the time domain literally allows for a signal pass band up to 1/2 the sample rate (or 48 kHz for 96k clock). Since there isn't audible information that high, the practical benefit is relaxed anti aliasing filters and their impact on the top octave. Since the nominal audio passband is typically defined as 20-20kHz the difference between 24kHz and 48kHz, on paper at least is moot. This is also complicated by how some A/Ds use oversampling in the conversion process.

While perhaps slightly beneficial for EQ envelope symmetry in the very top octave, level manipulations such as compressing or limiting will be more affected by word length than sample rate.

Josh Billings wrote on Fri, 13 April 2007 15:41

CD Audio is only 14/44 but almost always down sampled from 24/96 to get the cleanest sound possible. Granted weak DACs could throw the behringer out right off the bat and the sample rate wouldn't matter, but i do think it would make a big difference if all others were equal. It's like the small differences between 320KBPs mp3s & wavs (14/44).

The difference between 320kbps and 16b/44.1k is roughly 2:1, similar to the 2:1 ratio between 48k and 96k. But the very audible difference at the lower end (320k to 16b/441k) is because it is literally discarding audible audio data at the lower bit rate. The difference between 48k and 96k are far more subtle because both are higher than they need to be to capture the full audio passband with decent sample rate and resolution.

Josh Billings wrote on Fri, 13 April 2007 15:41

Anyways what do you guys think about going Digital from CD-> Mixer -> Signal Processor. Is that better than sending the unit an analog signal? WHat do you think is more stable & would sound better? -Josh Billings

In principle avoiding unnecessary D/A-A/D conversions should reduce opportunities for signal degradation, but with modern gear that degradation is pretty minor, so do what works.

JR

[Rob Spence]

Going all digital is good EXCEPT if you do too many format conversions. It will work best if all the digital connections are either the same type or at least running at the same speed.

Converting from SPDIF to AES, in my opinion would not be a good thing - you may introduce unpleasant artifacts.

While the high sample rate is very important for studio work, in live audio, or worse, playing commercial disks, no one is going to hear the difference.

Who here knows what the ambient background acoustic noise floor is at the venue you work at? I bet it is greater than 70dB

[Mike Pyle]

Does the VSX26 add any audible noise to the system? Any fuzz when it's idling with the amp gains up?

[Bob Leonard]

John Roberts  {JR} wrote on Sat, 14 April 2007 10:23

Josh Billings wrote on Fri, 13 April 2007 15:41 The difference isn't in the sample rate as it is with all the stuff you are doing with the signal. In the studio side of things you sample at 24/96 to get a super clean sample to really mess around with (compress/EQ/limit/etc) The more you mess with that signal the higher that sample rate helps you in the end. There are several conditions for archival studio recording that don't apply to live SR. If you need to later convert to a different sample rate you can down convert which is discarding data more accurately than up converting which is essentially inventing data points between real samples. The increased resolution in the time domain literally allows for a signal pass band up to 1/2 the sample rate (or 48 kHz for 96k clock). Since there isn't audible information that high, the practical benefit is relaxed anti aliasing filters and their impact on the top octave. Since the nominal audio passband is typically defined as 20-20kHz the difference between 24kHz and 48kHz, on paper at least is moot. This is also complicated by how some A/Ds use oversampling in the conversion process. While perhaps slightly beneficial for EQ envelope symmetry in the very top octave, level manipulations such as compressing or limiting will be more affected by word length than sample rate. Josh Billings wrote on Fri, 13 April 2007 15:41 CD Audio is only 14/44 but almost always down sampled from 24/96 to get the cleanest sound possible. Granted weak DACs could throw the behringer out right off the bat and the sample rate wouldn't matter, but i do think it would make a big difference if all others were equal. It's like the small differences between 320KBPs mp3s & wavs (14/44). The difference between 320kbps and 16b/44.1k is roughly 2:1, similar to the 2:1 ratio between 48k and 96k. But the very audible difference at the lower end (320k to 16b/441k) is because it is literally discarding audible audio data at the lower bit rate. The difference between 48k and 96k are far more subtle because both are higher than they need to be to capture the full audio passband with decent sample rate and resolution. Josh Billings wrote on Fri, 13 April 2007 15:41 Anyways what do you guys think about going Digital from CD-> Mixer -> Signal Processor. Is that better than sending the unit an analog signal? WHat do you think is more stable & would sound better? -Josh Billings In principle avoiding unnecessary D/A-A/D conversions should reduce opportunities for signal degradation, but with modern gear that degradation is pretty minor, so do what works. JR

John, Finally the correct answers to the digital questions from you and Rob.

RTA has never been my first or even much of a consideration to me. I place more value in RTE, as in REAL TIME EARS.

[James Duncan]

E. Lee Dickinson wrote on Sat, 14 April 2007 09:50

Input C can be routed anywhere you want. The other nice thing the Behringer does is treat the Sum input (selectable among A, B, and C) as though it were a separate input. That is to say, if you adjust the input gain on A, it does not affect SumAB. We have the Behringer in our B rig.

Thanks for that, and yeah, I did understand that.

What I was saying I guess was that if I am running stereo mains, *and* the aux-fed subs, that would take up *all* three inputs on the Behringer, which means that I would have nowhere to put the reference mic.

Plus, I think it is obviously much more convenient to have the ref mic input on the front panel like DBX, Peavey, and others do. This saves having to dig around in the back of the rack to plug in the ref mic when needed.

[Rob Spence]

I think you will find that the reference mic isn't used much by folks who use speaker processors. The builders of these things figure out that for a little extra programming they can include all sorts of features that look good in the marketing adds but it real life don't get used much.

As far as I have seen, delay (input and output), patching/routing, filters (LP & HP), Crossover, Speaker EQ, and limiters are what you need the box for.

If you do use the reference mic, it (in my opinion) should be for finding the characteristics of your speakers, outdoors, so you can set the EQ for them in the box. Then, you use your graphic as needed to adjust for a particular room or mix.

I prefer having clean front panels where some dodo won't plug something in when I am not looking

[Steve Hurt]

Rob Spence wrote on Sat, 14 April 2007 13:15

.....If you do use the reference mic, it (in my opinion) should be for finding the characteristics of your speakers, outdoors, so you can set the EQ for them in the box.......

Exactly what I did today.  Shot my speakers w/RTA outside.
My best tuning's on the DRPA have all started that way.

I just got new bottoms cabs, and needed to come up with new tuning for my rig.
The RTA found a couple things I hadn't noticed which really helped.

Shooting the room inside is not near as much help for me, especially on the low end.  

I've never been able to EQ a room node away.  I doubt the RTA can do it either.  Physics getting in the way I think.

Using the RTA outside really does seem to be helpful for a starting place, despite all the people saying RTA sucks.  I consider the RTA a very useful feature.

[Tim McCulloch]

Steve-

The difference between an RTA and a dual channel FFT analyzer (Smaart, SIM, etc) is this:  The RTA will show you how much level exists in a given frequency band for where ever the measurement mic is placed.  Move it, and the RTA display will change as the mic is moved in and out of comb filter nodes.

With FFT analyzers, the Transfer Function display will change, too, but by looking at the Impulse Response, you can determine if those changes are the result of time-domain issues like relections off of boundary surfaces or grossly mis-aligned systems.

Quote:

I've never been able to EQ a room node away. I doubt the RTA can do it either. Physics getting in the way I think.

That's quite right.  The FFT analyzer helps you see the result of nodes and modes and locate their sources.  The RTA only shows you a bump or dip in amplitude response.

The RTA can be a useful tool, but it helps to understand what it can't show you, as well as what it can.

Good luck, have fun.

Tim Mc