David Morison

As long as the monitors you're currently using for vox are reasonably flat in their frequency response and have enough headroom to handle the extra content put through them, then yes, it should be as simple as you hope. Ideally, you'd have at least one spare aux output available on your desk - that way the new guitarist gets a mix to themselves as they'll likely want more of themselves in their mix an anyone else will need. For monitoring, mono should be fine and, as you note, for a lot of gigs that's all that's really needed for FOH too. It can be nice to have stereo, and devices like the Helix will give plenty of opportunity to exploit that, but if you're a pub/small venue band then going mono shouldn't be a deal breaker. The other guitarist should be able to carry on as normal; as I say if you're able to give the helix user their own mon mix that will help avoid having to have that guitar too loud in everyone else's monitor mix so you should be able to balance the 2 reasonable easily.

If you really like the tone of the Elf & don't mind carrying both and you weren't needing the Elf's DI for normal FOH feed, you could run the Elf's DI into the fx return of the Eden to take advantage of the latter's increased headroom. Would need an XLR to TS adaptor cable, and it's a bit clunky, but might get you the best of both worlds.

Yup, that's exactly it 👍

Yeah, it's a little counter-intuitive, for sure. Your last line has it right - basically, all of the input channels feeding each mix can be pre-fade, but as the FX come post fade before they reach the IEM mix, they would remain post fade. So, if you turned up the vox in the main mix, the level of the dry vox would remain the same in the IEM, but the level of the vocal reverb would still increase in the IEM. Given that you hopefully don't need to make massive adjustments to vocal levels once the show is underway, and that the preferred reverb level is hopefully several dB lower than the dry vocal level in the first place this is unlikely to be a deal breaker, but it's an easily overlooked quirk that's worth being aware of. HTH, D.

Hi Al, That's not quite the full picture re reverb in the monitor sends. You're right the send to the reverb unit will be post fade - that's because we generally want the relative level of reverb & dry signal to stay the same when we adjust the fader for the main mix. However, each monitor output can still be set to pre-fade - so normal input signals will stay at the same level even when the main mix is adjusted. This does throw up the slight complication that the reverb that is being sent to the monitor will still be post fade, as that is where the FX unit is getting its signal in the first place, so if any big changes to the vox levels are made in the main mix, those will result in the amount of reverb in the monitor send changing too.

If you're only using one mixer, then the faders' main function is to control the main mix (desks with fader flip notwithstanding). As we usually don't want out monitor mix changing every time we tweak the FOH mix, pre-fade is the default for monitor sends. If you don't have a dedicated FOH engineer, and you rely on your monitors to get an idea of what the main mix is sounding like, then you might be better with post fade.

I only had a brief look, but I'm pretty sure the CQ doesn't have traditional, analogue style insert sockets. Agreed on the potential downside of using the Aux Output EQ to do your "de-clanking" - you might be able to make the bass sound better, but as you said you were happy with the tone of everything else in the IEM mix back in your original post, you risk making everything else sound worse in the process. It is however, free and quick to try, so no harm in giving it a go before buying/making a splitter for the inputs.

Sort of. I believe the OP used the term "pan" to refer to a continuously variable blend between 2 sources prior to sending the signal to the transmitter, rather than the more common sense of panning in stereo. The issue with this is that doing it with a hardwired combiner can electrically stress the outputs of the desk, which makes finding a solution based on working with the tone of the input(s) more important. Splitting the bass into 2 channels on the desk is the easiest & most reliable way of doing this, IMO. As @Chienmortbb has confirmed the CQ can't do that internally, a hardwired solution such as This for less than four quid is all it takes, assuming a spare input channel is available.

Hi Ian, Many devices do not like having their outputs mixed with other devices' outputs via a simple hardwired joiner (essentially, a Y-cable used backwards) as the outputs can interfere with each other. There's a good tech note on it called "Why Not Wye" from the former Rane company Here if you want to dig into a possible way of alleviating that problem. A simpler solution, if you have a spare input channel on your desk, is to split the bass input into 2 channels - EQ one for FOH and the other for IEMs, and only include the latter in your monitor mix. Many desks can do this kind of double patching internally, so you might not even need any external hardware; check the manual for the CQ to see if this would work for you. If the desk can't do it internally, then a simple y-split of the cable going into the desk would still be cheap & easy. HTH, David.

Looks to be perspective - if you scan across from the speaker mounting point on the ceiling to the brick pillar far right, it seems a little in front (downstage) of the front of the pillar, and even the back of the pillar appears to be a little in front (downstage) of the lectern, per the line of hazard tape.

Hi, thanks for sharing this. I've never had a mixer with such presets (or felt the need, having learned the hard way on analogue mixers) so it may help me and perhaps others to get a bit more info about how well they work please.... How different were the results from the presets vs setting up each channel manually (assuming you did that too)? Are the presets "black boxes", or can you see what they're doing in terms of EQ, compression & effects etc? If you can see "under the bonnet" as it were, are there any gotchas to be aware of such as EQ boosts or compression that would raise the risk of feedback in a live situation? Ta muchly, David.

To be fair, the D112 is one of the less heavily contoured of the typical kick mics (only about 6dB before accounting for proximity effect), so better for other instruments than say, a Sennheiser e902 with over 20dB of shaping built in.

Fs, Qts and Vas are all higher in V3 vs V2, so in a given cab & tuning, there will be a more prominent upper bass hump and less low bass.

I mean, given we're talking about 90's era Trace, it's probably just a typo and they really meant 47 tonnes rather than kg, right?

Sorry, just realised it was the 1528 that dclaasen was asking about, that 47kg was for the 1524 from the previously linked ebay auction, my bad. FWIW, while the same catalogue does show the 1528, it doesn't include specs. The 1248, which was the same size as the 1528 but with 4x12's was 45kg for the version without an HF driver; I'd imagine the 2x15 would be a few kg less than that.

47kg according to the 1998-99 catalogue I have.

To Heat. You forgot the 2 most important words... 😉

You've more or less answered your own question - there are so many pedals that do drive/dirt precisely because lots of amps don't. Given that lots of styles of music work fine with clean sounding bass, it's logical that a big chunk of the amp market doesn't go there. Couple that with the fact that lots* of dirt/drive type sounds can relatively easily be created without designing & building a whole amp, it's logical too that that sector of the tonal spectrum finds more expression in pedals than amps. * Obviously, there will always be those purists that aren't happy with anything other than output valve distortion & even transformer saturation, so I don't think SVT's are going away any time soon, but even they are just a subset of the overall drive/dirt spectrum of preferences.

Hi again, Yup, your basic signal flow concept is fine. You're right too that the "extra" outs like headphones, control room etc on that desk just allow you to monitor the main mix (or any solo'd channel); the only other output that allows you to properly tailor what's being heard is the single Aux. As a worst case bodge, IF you A: never use the EFX and B: never adjust your main mix once it's set at soundcheck, you could theoretically use the EFX send as a second aux mix. Obviously the potential for frustration abounds if you ever did need to adjust the main mix during the set - whoever was getting the EFX derived Aux mix would find it tracking the changes to the main mix.

No worries, you're welcome. Running as you are isn't going to blow anything up, but you're not getting the best out of your system as-is for sure. At the moment, the Titans and the Peaveys are both getting fed the same signal, so they'll be overlapping each other. That usually doesn't sound as good as separating them so that only one type of speaker covers any given part of the audio spectrum. Putting in a crossover would take a load off the Titans by filtering out everything below a set frequency (lets assume 100Hz for the sake of the example), meaning they do less work and (if need be) could maybe be pushed a little louder. It would also ensure the Peaveys are only trying to reproduce content below that frequency, meaning they're less likely to contribute unwanted muddiness in the (low) mids.

Short answer, no, unfortunately it wouldn't work like that. Why - because the high pass filtering in the Peavey bins occurs after the signal has already been amplified, meaning you'd be passing a full power signal to the input of the Titans, which they very much won't like. The right way to do this is to run your signal from the desk to an active crossover, which splits it by frequency and gives you separate outputs to feed one set to the Titans and another set to the amp that's driving your Peaveys. The modern way of things (that gives you maximum control of what goes where) is to do it digitally - entry level units start around a hundred quid: Thomann Mini DSP. For similar money, you can get new, analogue units that offer less control, or if you really want to save the shekels you could look for second hand rack gear. Behringer, Peavey, dbx have all made relatively affordable crossovers that should be findable for 50-100 s/h with a bit of patience. HTH, D.

Fair point, for some reason I'd defaulted to assuming we all tend to leave cabs set up ready to use but obviously, not. Several of the Ashdown Rootmaster series of cabs are shallow enough - the 414, 210 & 112 are all 336mm deep, the 115 343mm, so would all fit. I haven't used them however, so can't comment on their tone.

Note that for most cabs, you'll need an extra few cm behind it to clear the speakon & cable, which usually sticks out straight backwards. That will reduce your useable cab depth to perhaps 30cm or so, which will really narrow down your options unfortunately. @Downunderwonder's suggestion of a curtain rather than a door is looking like a really good idea at this point, if you can persuade Mrs Grapefruit.

Cool, well for the time being I'd say leave it as is at least till you've had the chance to turn it up a bit. The Equal Loudness effect means that as you get louder your perception of bass improves, so even if the low end were to sound a bit light at livingroom volume it might still be viable for your rehearsal use-case, even without adding ports.

As Bill says, there is a gain to be had from porting, but it's modest - less than 3dB. By the way, how does it sound at the moment? Are you noticing it being particularly light in the bottom end at all? Also - how's the back panel & its latch and seal holding up? If you port the cab for more bass, you're also increasing the chance of extraneous noise if the hinges/latch don't hold it closed tightly enough. Anyway, as I am an inveterate WinISD tinkerer, Blue is sealed, Green vented with 2 of these: [url=50mm port]: That's theoretical however and there are a couple of caveats: 1; WinISD's port calculation assumes the backside of the port is at least one diameter away from any boundary such as the sides of the cabinet - you will be struggling to achieve that, so the ports may tune lower than predicted. If they were to tune as low as 40Hz rather then the theoretical 46.5Hz, the curve would look like the bold Green trace here - not much different in this case: 2; Vent Air Speed may be a problem. For best performance, the general rule of thumb is to aim for 17m/s or lower, though some sources do say that for non-hi-fi applications, up to twice that may be acceptable. At the ~33V your BAM200 can put out, the predicted airspeed is this (again, bold = 40Hz tune - in this case that actually plays in your favour): In practice, this is probably less of an issue than the modelling suggests, due to a couple of factors - first, we very rarely play with the amp flat out the whole time and allowing for dynamics in our playing, average power levels should be a lot less than max and second, those graphs assume full power at each frequency, which for an instrument like bass which is rich on harmonics, doesn't happen - our power is divided up over several frequencies for each and every note we play. So, overall, more work for marginal gain - if you're an inveterate tinkerer you may still want to do it but don't expect night & day differences from the current sealed condition. HTH, Good luck, D. Edit - link formatting