Posts by EdwardArnold

    i know that the soundmen like to get signal between -12db to -18 db in digital mixers and signal about 0db in analog mixers.

    How can I know how much db my kemper send out to the mixer? maybe too loud or too low?

    How can I measure it?

    thank you

    Yep, peaking around -12 dBFS and averaging or -20 dBFS is a good starting point when mixing. It’s laudable you try and get things to a sensible output level prior to arriving at a gig. There are many variables though between live sound rigs that will mean you’ll always have to talk to the engineer and they will want to set levels based on real signal. For example, the analogue operating level of different desks can vary by a significant amount. Also, you’ll have almost exclusively mic inputs on-stage, meaning the engineer will PAD down those that you feed with your Kemper, which outputs line level. PAD functions on different desks and their associated stageboxes vary. Many are -20dB but you’ll find -15dB and -30dB. This means the mix engineer’s workflow will be to pad, then increase input gain until they meter where they want them to. I think the most important thing to tell your mix engineer is that you are outputting line level, not mic level, so that they can be prepared.

    What do you think if i lower the gain of the audio interface to zero, That means he won't increase the signal at all?

    Although unity gain on an analogue input doesn’t change the level through the preamp circuit, the operating level of the interface will become a factor when translating to dBFS after the ADC stage. The UK standard operating level is 0dBu = -18dBFS. The US standard equates to 0dBU = -24 dBFS (usually written as -20dBFS = +4dBu), and there are other standards in between. Most audio interface manufacturers plump for something in the middle to try and integrate with as much gear as possible without multiple versions being required, so without looking at the specs, you can’t know how much to offset your measurements by. This also applies to the analogue inputs on professional mixing consoles, analogue and digital, where operating level is either fixed or selectable.
    If you go via S/PDIF, you circumnavigate all of the analogue stage operating level variables.

    Some details:

    It's true that many studio guys believe in 96k. But it's a believe only, and they are wrong.

    In my day-to-day encounters with professionals from the studio, broadcast, live and post production strands of the audio industry, it’s interesting that only the commercial studio userbase puts forward 96 kHz as a consideration in terms of what they hear. In other sectors, it’s only discussed in terms of system latency and interfacing (e.g. MADI or Dante channel counts halving from 48>96k), or network audio bandwidth double if for same channel count etc. Seemingly far more impactive/beneficial to the audible performance of DSP products I work with has been the progression from fixed 32bit to 64bit floating point DSP calculation, greatly reducing quantisation distortion due to the higher resolution of coefficients generated to instruct changes in DSP parameters.

    I've used wide range of analogue and digital consoles for a variety of purposes on both sides of the musician/engineer fence. You've made some valid points there, and I would agree that you should definitely have a standard IEC power input rather than an external supply. The three manufacturers you mention make reasonable budget mixing desks, but I would highly recommend looking at Allen & Heath. I've used a ZED 10 FX for small gigs (solo / duo / trio) and rehearsals for years. It has 4 XLR inputs for mics but the option of TRS inputs for Line, so you don't need combo jacks. Additionally two dedicated stereo channels with 1/4" and/or RCA inputs. The main outputs are XLR but you simply need XLRf>TRS cabling if your destination is going to be 1.4" TRS. Sounds great, but look further up the model range for more i/o for a larger band.

    I used to have a PA20, which I really liked the ergonomics of, plus the output parametric EQ and sensible routing options (background music mode, foldbacks etc.). They discontinued them but I'm looking to pick up a PA12 second hand, I miss it so much.

    I can't say I've been overly impressed with Soundcraft or Yamaha. Mackie Onyx stuff was alright back when I used to use it, but below A&H in terms of preamp quality and ease of use in my opinion.

    As for the 'future-proofing' argument, it's true that you can quickly run out of functionality. I would look at getting more than one aux/foldback output on a desk. For example, two or three pre-fader auxes (may be labelled foldback busses) and one or two post fader (for internal and/or external effects). A&H and Mackie have widely adopted inbuilt effects. In general the basic hall and room reverbs are ideal for live use, and you have the option of hooking up outboard as well/instead if you prefer.

    Never had a Kernel issue on either of my iMacs. Course I run legit software. Whats that like?

    Grey screen, rather than blue. Same thing though; an unhandled exception due to hardware or software renders it impossible for the motherboard to continue running. I’ve used a lot of Macs and they are certainly not immune to it, but it’s rare. On the flip side, I recently rediscovered a PowerBook G4 from 2004 in my loft that I haven’t used for 10 years - powered it on and it booted fine! Damned things won’t die.

    Option 2 as your guitar signal path will be shorter. This means it won’t unnecessarily go through another preamp/converter stage and will be lower latency (if your option 1 plan includes A/D D/A conversion).

    As another option a small analogue mixer could help you out. Something like an Allen&Heath ZED10 has mic inputs, stereo line inputs and USB for audio interface functionality. The difference being that you have all of the mixer routing functionality in the analogue domain (main outs, monitor out, aux send) without the need to use a separate audio interface should you want to perform basic tracking duties. I have a few small mixers for this sort of purpose, but recently I’ve just been patching playback into the aux returns on the Stage (FX returns) as it’s very convenient.

    I’ve got a Pedal Train Classic 1. I haven’t removed the feet, and I’ve put some low strength Velcro on the bottom of the Kemper. I’ve found I can slot the Stage onto the board and it’s in the perfect position and is easy to pull it off when I want to take it off. The vents don’t seem to be covered.

    I’ve also got an expression pedal on it.

    I’d be very grateful if you could post a photo of this as I’m looking at the Classic 1 for my stage, with an EP-1 and possibly some other bits. While I’ve got the dimensions for everything, it’s hard to visualise how much will be leftover around the edges. Thanks

    spdif is easy as you dont have to care so much about the Levels,

    but to me Analog outs from the KPA sounds slightly better than spdif

    the Kemper has great soundingAD Converters

    That's interesting - I can believe it, but I guess the question is where in the signal chain the sounds diverge slightly. Not had my Stage long enough to do an AB on these yet, but I'm interested to have a go at this with some fancy measurement tools and see if I can spot anything :)

    Were you re-converting the main angle outs via a dedicated interface back into your DAW to compare?

    Either option is going to work, they just have different considerations. SPDIF will require a dedicated 75 Ohm cable (i.e. not just an RCA phone you have for consumer analogue equipment), and the Kemper has to be the clock master, as it can't sync to incoming clock. You'll need to pick your preferred sample rate (44.1/48 kHz etc.) to clock your DAW.

    Be careful re-clocking your Focusrite with your monitors un-muted, including sample rate and source changes as I bet it doesn't lift its output relays and will go BLAT when you do this. Better to have the Kemper on first, but this can be impractical if you've already got your computer up and booted.

    Personally, I'd go analogue from the Kemper in a simple rig like this so you can forget about the re-clocking scenario above. It would be nice if there was a sync ref input on the Kemper (word clock for example) so that it could be a slave and better integrate with other systems, but meh.

    As above - you should use the balanced XLR outputs from the Kemper into the TRS sockets of your interface, using XLRf-TRS cabling. Alternatively, you *may* be able to PAD the XLR/Mic inputs down on your interface to bring the level down to something that doesn't clip, or you could run the output of the Kemper into a DI box if you have one, to take bal Line down to bal Mic.

    Either of these solutions is preferable to running TS (unbal) instrument cabling as the latter doesn't reject the interference/noise that can be picked up and the level will be 6dB down from bal Line as the extra XLR pin (3) will be grounded or not connected. If this is just a quick demo recording just to get some ideas down or stick on YouTube at 128kbps I wouldn't worry too much, but if you're committing something to a recording that you're going to want to mix later, stay balanced.

    I guess you have an interface with combo XLR/Jack inputs that auto-switches between Mic and Line modes depending on the physical connector, trying to be helpful? A lot of Focusrite units are doing this now.

    Rig Browser shows 8 rigs per page, in two columns. Given that the Stage has 5 footswitches, could there be 5 per column so that it tallies better? It would also reduce the number of Browser pages, which can only be a good thing. A little 'thinning' of the title bars and I'm sure you could squeeze another text line in there.

    For those with potential footswitch issues straight out of the box, just try 'exercising' the switch rapidly through 50-100 presses and retest before drawing conclusions (much quicker if done by hand and easy to push as they're momentary). I say this as it resolved my issues with switches 4 and 5, which now behave perfectly. I suspect switch contacts have not been actioned much/at all because they are new will be generating some switching noise. If it was part of an analogue circuit, you'd hear the noise in the signal. If this is switch is just being interpreted as binary the noise on that signal could mean it doesn't reliably send 0 or 1, either not triggering or double-triggering on/off. This seemed to be the behaviour I had, and it quickly went away. I don't think this is poor quality componentry per se, as this is common to all electro-mechanical switches/pots/faders including those on new large-format mixing consoles I commission that just need to be used.

    If it's still broken, it's broken, but I'd give it a go first if it really is straight out of the box.