Anyone else mounted the Stage to a pedalboard yet? If so, photos please. Pedaltrain Classic 1 looks like it would do the trick without being unwieldy. I only have a Classic JNR at hand atm, which is too narrow. Think the Mission EP-1 could just stay on the floor as it’s already quite tall.
I think the acoustic sounds of 30-year-old records sound better than the acoustic guitars recorded today on pop-rock CDs. That old sound, in my opinion, is more realistic. The only exceptions are the recordings of the great guitarists who mainly play the acoustic guitar (tommy emmanuel, andy mckee, etc.). Most acoustic recordings are too "plastic" compared to the true sound of the guitar.
I think perhaps what you are hearing is the change in approaches to mixing and mastering, more than the changes in the way the acoustic guitars are captured. Especially in the modern pop/rock genres, there can be huge amounts of dynamic compression and filtering of frequencies to allow acoustic guitars to occupy a space in the mix, without taking away from anything else, sometimes reduced to little more than a percussive strum sound. What you hear from Andy McKee and Tommy E is the sound of a miked guitar, for sure, but their music has the guitar as the sole occupant of the complete mixed track, in terms of arrangement, dynamic range and frequency bandwidth. If you've got some specific track examples, I'll have a listen and see whether I think what I'm saying still holds true.
... although I'm not sure his Dante vs. AVB arguments make much sense... "Dante does have some serious networking problems which are being solved in AVB"? Dante uses DSCP to give audio higher priority - something supported by any decent switch, without need for a more expensive AVB capable switch.
But when you say "DANTE please", why not ask for the most obvious (which I believe would be yet another step ahead of the competition for Kemper): plug your guitar into the Remote, and have the signal streamed through Dante over the network cable to the Profiler, along with the Remote commands. Perfectly feasible - no latency - no risk for network priority issues - one single cable to the rack...
Yeah, I disagree with much of what this chap is saying too. Dante has no serious networking problems, otherwise most of the national broadcasters I work with around the world would be in trouble. DSCP queuing is indeed supported by off-the-shelf switches. Prioritising e.g. clock multicast data is only relevant when the network bandwidth is limited, and it's easy to provide sufficient bandwidth across even very large networks to avoid this, but when required, it's there. All professional Dante networks I encounter are physically separate, or separated by switch VLANs, but QoS and DSCP is utilised on occasions where network traffic is mixed (I see this more in music studio networks with lower channel counts).
Audinate publishes following Dante specs :
1 switch hop: 0.15ms, 3 switch hops: 0.25ms, 5 switch hops: 0.5ms, 10 switch hops: 1ms, +10 switch hops: 5ms
These values are not affected by the number of devices or the number of channels used.
Anyway - I'm not at all an expert in this matter, I have no experience with AVB, and therefore I have no preference whatsoever for one or the other technology. I'm confident that Kemper will pick the right one for their Profiler2+Remote2
These values were quoted in Dante Controller Latency settings, and were conservative - they are no longer stated as such. Realworld performance was/is considerably better in most cases, and most large broadcast Dante networks I commission are set to 0.5ms network latency with more than 5 switch hops. This is the max latency threshold you're setting before a network packet is considered late, and dropped. When you set 0.5ms latency, most of the time you are getting better performance than that e.g. 0.3ms.
I support Dante all the way, and so do the vast majority of manufacturers in audio over IP. The reason is that it is a complete technology, that uses the AES67 transport stream (as do the other protocols including AVB), but also has accompanying software, control, routing, device discovery and device announcement capabilities. No-one else has an offering that is as complete, and out of all of the AoIP protocols (because there is no complete standard yet), Dante is the leading horse in the Broadcast and Live touring sectors.
Cool band. I get the whole need for separate mixes - it's the same deal in the bands I play in, with totally different wedge or IEM mixes per player. I now just play as a dep in one of the bands, so I went to see them tonight as an audience member. I've become so used to the monitor mix I prefer that it sounded rather strange hearing a full band mix!
(P.S. I think you've got your TLAs mixed up - Enterprise Information Management probably isn't that helpful when gigging )
More recent Expression Systems on Taylor guitars go a long way towards good sound via DI from stage imo.
Most people I know recording acoustic guitars are still using mics, but there are plenty of mix processes and decision after that that can limit dynamic range, bandwidth and overall impact of a well-played acoustic guitar part.
As an engineer, and large console user, I really recommend having a conversation with your next mix engineer in advance of your gig (when you send your technical spec, perhaps) about what you want out of an IEM / monitor wedge mix. Unless they are severely limited on the number of auxes available, there’s no barrier to driving monitor mixes from FOH for many mid-sized venues, especially with a heads-up to assign some busses in advance. A dedicated monitor mixer is a nice thing, sure, and yes you’d use mic splits, but having a band try and mix their own through the show without it being near anyone sounds less effective than having the FOH engineer look after things for you.
In a perfect world, we’d all take or consoles and dedicated monitor engineers with us, but that’s the preserve of the larger pro touring scene. The advent of small and mid format digital consoles means in many small and mid size venues that aux outs for monitors are in good supply these days, and if you still want to mix your own monitors, often there’s an app for that...
My Google powers were exhausted yesterday, clearly - apologies.
Did you notice any issues with the taper? It’s not going to make much difference in terms of voltage or current draw through the output stage. The reason impedance is so tonally effective at the input stage is the low impedance, passive pickups cannot simply draw upon more current when loaded, whereas a buffered/amplified signal can do precisely that. There is far less tonal effect from loading a line output to this degree, if any. The main thing that will be affected is the volume taper of the pot.
Looks like the Donner pedal is powered, and therefore buffers the signal coming in anyway. Can’t find specs for what impedance it is, but if it’s post it’s own buffer, it’ll be low impedance e.g. 25K - 50K, or should be.
Impedance matching isn’t critical for low impedance audio, which a line output will be. It’s far more critical when deciding what pedal to put after your guitar. A 250K volume pedal is ideal for guitar pickups, as it doesn’t load them down, which restricts high frequencies - voltage transfer (where the audio signal data is) is maximised, rather than power transfer.
However, if you put a 10K, 25K or 50K passive volume pedal after guitar pickups without buffering, you’ll definitely screw up your tone, losing highs in particular, and the volume taper will be all wrong. After a line output, I’m sure it will be fine. Try it and see/hear!
One question though, is this really just one more point of failure in your live rig, right where you don’t want it..? Depends on how useful you think it will be, and whether you’re using it live or not I suppose.
Perhaps it goes without saying, but first consideration is to ensure impedance matching between the power amp and the speaker(s).
PowerHead and PowerRack:
600 Watts at 8 Ohm
300 Watts at 16 Ohm
Simply add two ambience mics pointed at the audience. Beef those up with a compressor but have them ducked by the FOH band mix. This way you can hear the audience inbetween the songs without messing up your IEM sound when playing.
+1 for this suggestion - the side-chain trick is a great way to achieve a more natural monitoring environment without screwing up your direct mix. tylerhb do you use fig-8 or omni pattern mics to try and include some off-mic stage stuff as well, or just point a pair of cardioids at the front row? Sounds daft, but when you're off-mic between songs and can't hear your own words coming out, it can be a bit weird. Perhaps the answer is, don't ever be off-mic, but I'm interested to experiment a bit more.
For profiling, SPDIF circuitry at the non-Kemper end may be impactful as the system is not calibrated for use with it. You’d therefore also be profiling the sonic characteristics of the SPDIF interface(s), which would be different to using the analogue input of the Kemper to re-capture the transient detail in the signals it sends out to profile. Clearly, Kemper achieves some voodoo levels of accurate profiling, and I’d guess the calibration to account for the known parameters of the analogue input and corresponding analogue to digital conversion stage is paramount to this. I'll admit I'm just guessing though.
The Kemper sends out white noise signals/pulses and works out where the frequency response distorts due to amplifier clipping. This kind of graceful/progressive distortion characteristic is going to be a defining and desirable characteristic of the amp in question, and Kemper’s forte in terms of reproduction. Trouble is, when these same signals instead pass through a SPDIF interface, which is fixed 24bit and therefore headroom, if the amplitude of the test signal clips the digital domain it won’t be full of pleasing 2nd order harmonics or progressive, it’ll be absolute and full of offensive harmonics, as the signal peaks are literally cut off flat with no more bits left to represent them.
Do you/have you already explored profiling your processing chain in your DAW that you want to capture via analogue inputs and outputs? You'd replace the mic with a DI box to take your return line-level signal down to mic level, right?
By the same token, you’re also profiling the response of your audio interface inputs and outputs doing this, but at least if you run at the highest sample rate your system will permit you can retain as much of the fast-transient detail that the profiling system seems to send out and capture as possible, and you have control over the gain stages so you can trim things in a way that they don't clip the A/D stage.
It’s an important topic, as hearing damage is permanent accumulative, and surprisingly easy to achieve if you’re a musician. As Gary_W mentions, the ringing is a sign that damage has occurred - this is the swan song of some tiny hair cells in your inner-ear as they die, having been over-stimulated by frequencies they were designed to vibrate in sympathy with. You’ve got quite a lot of them, but your sensitivity to those frequencies reduces with time and further high-level exposure. It’s worth getting a hearing test each year if you gig a lot. I started to discover a 4kHz ‘hole’ in my hearing response in one ear, and it was due to proximity to drum kit on stage. Ear plugs in sound checks/rehearsals and changes sides of the stage/trying to keep more distance will have helped since.
Most venues I visit (e.g. arts centres, functions rooms, concert halls and clubs) have to adhere to regulations regarding max SPL. You’ll find that most rock/pop band rehearsals will exceed 90dB at audience distances, but some venues can have this set limit (others slightly higher). It can be influenced by venues being near residential areas as well, to try and limit sound leakage out of the building.
If you stay over the limit for a certain amount of time, it can cut power to the stage, or the main power amps if there’s a house system, which is less than ideal. This is a public health and safety limitation, but the trouble is, SPL diminishes with distance, and the placement of measurement mics has a very significant impact on what readings are taken. For example, just one of my small Genelec studio monitors is rated at 97dB at 1m. That metre is important though, as the SPL will halve every time the distance is doubled. So at 2m, it will be 91dB, at 4m it will be 85dB and so on. (For reference a 3dB change in signal is half the signal intensity, and it takes 10dB before the average person perceives half the volume), so whether it is really too loud for someone in the front row is up for debate if the measurement mic is just above the front of the stage (as is common). In pro live tour venues, the FOH engineers/system techs will have a measurement mic just in front of the mix position that is used to measure the difference in signal between what is being sent to speakers and the signal after room acoustics have been involved, plus the SPL levels that most of the audience are experiencing at a known distance. It often hangs around the 100dB point for average at a big rock show. This is loud, but not dangerously loud as long as there is some respite. Peaks can go beyond that, and if you’re right at the front next to a floor stack or just under one of the flown speaker arrays, you’re maximising your chances of getting permanent hearing damage. The reason people get their kicks from this is that over 100dB your inner ear starts to generate the same kind of output as when you’re free-falling, due to overstimulation. Giggity.
What I’m trying to say is, measure at the listening points/distances to get a good idea of what’s going on, both in the audience and on-stage. You also need to consider whether it is the length of exposure to high SPLs that is contributing. For example, if you’re in a death metal band, I’d throw money down that 90% of the set time you’re full tilt, full spectrum in terms of sound intensity due to distorted guitars and plenty of cymbal crashes. This long exposure to high frequency intensity is definitely something to be aware of, and ear plugs are very effective at cutting high end (recommend moulded), although your singer will struggle to pitch correctly.
As nightlight suggests, IEMs are one of the best ways to reduce your stage volume. I’m a singer and guitarist, and I go through phases of loving or hating them, as the sense of being in the moment and hearing the audience is cut down, even with ambience mics at the front (these make a big improvement though). Monitor consistency is great, and if they are well-fitting, they really cut down on the bleed into your ears from others’ backline amps and floor wedges, so the levels of what you’re sending through them can come down. If you’re having to drive your IEMs hard still to get over the levels on-stage it could well be worse than the issue you were trying to avoid in the first place, so it’s best that everyone is on IEMs rather than a combination (unless you’re playing a massive stage and far apart).
I may be able to explain this.
Short version: You’re using the unbalanced output of the Kemper (TS - tip-sleeve 1/4” jack) instead of the XLR (balanced) output (?). Take the XLR output into the balanced line input of your Focusrite and you should have unity gain through the system.
Your DAW uses the dBFS (full-scale) meter scale, with 0dBFS at the very top of the meter. How analogue signals - measured in dBu - line up with dBFS varies depending on the operating level of the analogue equipment involved. This is what others are referring to regarding a pro +4dB operation mode. In real terms, this would mean that -20dBFS on your DAW meters would give you +4dBu in the analogue domain. This is a typical US operating level, and it is easier to think of it as ‘-24dBFS = 0dBu’. This can then be compared to the British standard -18dBFS = 0dBu. As equipment is sold all over the world, many manufacturers pitch somewhere in the middle if they don’t have adjustable operating levels. For example, your Focusrite unit appears to have a line up level of -22dBFS = 0dBu, looking at the specs.
Now, you can consider these specs in the other direction as well, e.g. 0dBFS = +22dBu. This then tells you the (likely) maximum analogue signal level that the device can achieve before distorting. It may be able to go a little further, but this is your defined analogue clip point.
If you look at the Kemper spec sheet, the operating level of the Kemper is also 0dBFS=22dBu (which is good news for you, as no further line up adjustment would be required in your system for unity gain). Shorthand is +22dBu, which is why the manual states “max output level: XLR +22 dBu”
Significantly, you mention the -16dB value. This is the operating level if you only use the TS (Tip-Sleeve) jack output, which is unbalanced, the max output level is +16dBu. You will have balanced line inputs on your Focusrite Clarrett, so take the XLR output of the Kemper into one of those to reclaim your missing 6dB. The reason the TS is lower is because balanced signals comprise an in-phase and out-of-phase copy of the original signal on pins 2&3, in addition to the ground conductor on pin 1. The receiving differential amplifier measures the difference between the two signals (peak to trough of a waveform if it helps you visualise), which would be twice as great as one signal on its own when measured to ground (unbalanced). Essentially, you are halving the amount of voltage difference in the signal if you use the TS output vs XLR, and half the voltage equates to a 6dB reduction when metered. You should also consider that the interference picked up on the TS signal is rejected in the equivalent balanced signal, as it is the same on both the +ve and -ve cores of the cable, and therefore ignored by the differential amp.
Hope this helps! Let me know if I’ve misunderstood your conundrum.