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How to generate latency in live situation?

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...doing something other than turning down one of the two bass guitar signals
Most sound engineers receiving two signals from a bassist have likely mixed the band with one muted anyway rendering the whole thing pointless.

If they use both signals and the mix sounds thin in the comb filtering regions they’ll compensate with EQ on the bass or some other relative aspect of the mix.
 
Latency would be more for a comparable unit run at 44.1 kHz (likely 3 msec-ish). Consoles that run less latency are typically running 96kHz sampling, which reduces latency in the A/D's and D/A's. Latency of .7 msec for a unit running 44.1 kHz is odd - I doubt that's signal input to output. A latency of .7 msec (16 samples) for processing in the digital domain I can see, but not input to output, which is what the user here is fighting.

I'm not sure where you are getting that figure from. Here's how I know what kind of latency I am seeing through the behringer:

 
Thanks for your answer
Indeed this out of phasing was audible on recording , not sure it was audible in live with the band .

By the way we never played with sound engineer yet, the only time I used those two signals, we were alone , ( so impossible to adjust on the fly in between songs )
I will turn off the DI if needed !
 
Hmm.. One way to fix the problem would be... to adjust the phase? Something like this : Phazer - Radial Engineering . Or simply something that reverses the phase 180 degrees - might make it better or not.

Houuu the phazer seems Nice ! Also seems not cheap

I think a 180 phase reverse will make the thing worse, because it is only a little little delay between the two signals, they are almost sync'd
 
Houuu the phazer seems Nice ! Also seems not cheap

I think a 180 phase reverse will make the thing worse, because it is only a little little delay between the two signals, they are almost sync'd

The phase difference (in degrees) is relative to frequency of the sound. The 0.7ms delay is equal to 180 degrees phase shift at one given frequency - I believe it's around 500Hz-1Khz. I suggested it, since it's might be easier option for most ppl to try out. I must admit I don't fully understand what I'm talking about and may be totally lost here - trusting there are more knowledgeable people to correct me.
 
Most sound engineers receiving two signals from a bassist have likely mixed the band with one muted anyway rendering the whole thing pointless.

If they use both signals and the mix sounds thin in the comb filtering regions they’ll compensate with EQ on the bass or some other relative aspect of the mix.

I agree that most sound engineers will just use the signal that sounds the least crappy. But using EQ to fix comb filtering is not the best idea and I doubt it will give you a good result.
 
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But using EQ to fix comb filtering is not the best idea and I doubt it will give you a good result.
No one is looking for "the best" in that scenario, just a serviceable solution. The engineer is simply trying to make the live mix sound good as quickly as possible so the audience enjoys themselves. If the OP were describing a studio setting the engineer would likely choose a different solution, but since it has been stressed that this is a live mix issue the solution changes to fit the situation.
 
Swap the 6" patch cord to a 25' or longer cord. You'll start to get some latency. If you want to dial in latency the Line 6 G-30 wireless unit has a know while adds in latency, I suppose you could do that too.
 
There are rack mount delay systems that can delay a signal in units of 1/100 of a millisecond. They usually have two channels. You could put each signal into one of the channels and dial in the amount of delay you need to get the signals back in phase. That's exactly what they're designed for.

Here's a link to the first one I found that fits the bill:
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No one is looking for "the best" in that scenario, just a serviceable solution. The engineer is simply trying to make the live mix sound good as quickly as possible so the audience enjoys themselves. If the OP were describing a studio setting the engineer would likely choose a different solution, but since it has been stressed that this is a live mix issue the solution changes to fit the situation.

I understand. My day job involves running a large studio and theaters. I'd be interested in seeing what type of console the OP is working with. The best solution would be to delay the DI signal in the console. The second best would be some type of delay between the DI and the console. The serviceable solution is just using the Radar output. I feel that changing the EQ to address the problem will more likely emphasize the problem. Of course, I come from a background of using EQ as a last resort. I got to do a master class with Al Schmitt years ago and it changed my whole outlook on audio.
 
The serviceable solution is just using the Radar output.
Agreed

I feel that changing the EQ to address the problem will more likely emphasize the problem.
I agree that it definitely could, but knowing how most soundguys in clubs, bars, small theaters, etc. operate I'm sure the first they would do is try and address it with EQ.

I got to do a master class with Al Schmitt years ago and it changed my whole outlook on audio.
Al is a genius to say the least. I read an article he put out on gain staging in the box when working in a DAW that was absolutely fantastic. It basically confirmed some things that I had read previously and been implementing. My thinking was "if it's good enough for Al, it's good enough for me."