msmcleod

The best way to get it to play what you want is to provide it with a MIDI file of just kick & snare. Jamstix will then "paraphrase" your MIDI file in the style of the drummer you choose:

Remember that Kontakt 6 is fairly new. If you buy additional libraries (which are more than likely designed for Kontakt 4 or 5), there's always the possibility that it might not work properly in Kontakt 6.

Yeah, I've got it on three machines so needed the extra license.

Personally, I'd keep them. You never know when some instrument isn't 100% backwards compatible.

Right click on the transport record button, and make sure the following two options are set: If you've got Auto Punch enabled, make sure "Mute Previous Takes" is NOT checked.

If you find the drums are sounding a bit thin, try adding this: https://kazrog.com/products/true-iron/ It's a transformer emulator that really adds girth to the track.

If you've got complex routing in your project, it can be a pain when a track automatically has it's MIDI echo on. For example, cases where one synth is controlling another (Jamstix & AD2 spring to mind). Having the track being echoed can have undesirable results when you want to select the track to adjust something else. I normally have this option enabled during tracking, but quite often switch it off for more complex mixes.

Things to check: 1. Have you got input echo enabled on your midi/audio tracks or instrument track if that's what your using? 2. Check you've not got any surface controllers defined in Preferences that are assigned to these devices

Funny you should mention this, I tried to import a video file whose audio was recorded at 22Khz (a camtasia presentation), and it did exactly the same. I thought that changing the import file depth to match the project depth might help, but it doesn't. Is your .wav file recorded at a sample rate smaller than 44.1Khz? [Edit] - I still can't work out why the import does this, but this is how I fixed it afterwards:

The error message is a bit vague, as it's saying either: 1. It can't open the audio device at all; or 2. It can open it, but not at the given sample rate (i.e. 44.1Khz / 16 bit) In either case, it looks like some other program (or Windows) is using your Octa-Capture and/or setting it to a different sample rate or bit depth. So first, check you've not accidentally disabled your Octa-Capture, and that it's definitely assigned to Cakewalk. Then check that Windows isn't using your Octa-Capture (check input devices as well as output devices).

If you've got an on-board sound device, make sure that is enabled and assign Windows sounds to that. That way only Cakewalk will be using your Octa-Capture. Your on board sound device may have other output "devices" listed, e.g. a digital output as well as the analogue one (usually labelled "Speaker output"), or an HD output. Disable all of them apart from the speaker output, and make sure the speaker output is what Windows is set to using. I'm unsure as to why ASIO isn't working as well as WDM. Usually, WDM performs far worse than ASIO. But if you are using WDM, make sure Cakewalk is only using your Octa-Capture. You'll now have to work out how to listen to the sound output from your on-board card if you need it. Personally, I plug mine directly into my 27" monitor as it has audio inputs. An external mixer (even a passive one) could work - e.g. previously I had it plugged into the 2nd input on my Mackie BigKnob Passive so I could switch between them, but plugging it into a spare input on your Octa Capture could also work I guess.

Yeah, that's the way I use it most. I write the MIDI drum part I want, then it plays it like a real drummer

Yes. Think of EZD & SD3 as the sounds, and Jamstix as the drummer. Jamstix doesn't use preset patterns - it uses "models" of real drummers. I use Jamstix to drive AD2. The sounds in Jamstix are OK, but not as good as AD2. IMO they're better than SD3 though. [Edit] - A note of caution though... the learning curve for Jamstix is steep. Until you understand how it works and how to leverage it, you won't be blown away by the results... at least not at first. This guy has done a fantastic collection of in-depth tutorials: http://www.djbolivia.ca/videos.html

Make sure this is set:

Absolutely true.

Just noticed the following on Wikipedia: (Note Celemony's "get out" clause in bold) Version 2 of ARA was announced during NAMM in January 2018, introducing new features such as the simultaneous editing of multiple tracks, transfer of chord track information, and undo synchronization with the DAW. DAWs which use ARA version 2 are not automatically backwards compatible with plug-ins using version 1. The first DAWs to support ARA version 2 were Logic Pro (version 10.4, released in January 2018)[6] and Studio One (version 4, released in May 2018). Of course Cakewalk is the other way around - it's a DAW using version 1, which is using plugin version 2. But it does raise doubts about backwards compatibility.

I suspect that if and when CbB supports ARA-2, the problem will likely go away. CbB is probably making the same ARA-1 calls it always has. It's Melodyne that has changed, and I suspect as a result, they've subtlety broken ARA-1 backwards compatibility.

Balanced cables are generally used for either longer cables, or cables carrying a weaker signal. The are designed for preventing noise that enters the cable after it has left your audio output. The longer the cable, the more chance of noise being introduced along the way. For weaker signals (such as a mic), any introduced noise will be much more noticeable due to the high gain of your preamp. The balanced cable holds two copies of the signal, one in reverse phase. As any noise introduced will affect both signals with an equal phase, reversing the "reversed phase" signal and summing it with the original will mean that the noise is now reversed in the 2nd signal and will cancel out. So the question is, how likely are you to get noise introduced into your cable if it's a short run between your audio interface & your monitors? I guess this largely depends on your environment. Obviously it's better to use balanced cables if you can, as it will cancel out any introduced noise, but unless you've got a lot of RF being produced around your cables it shouldn't really be necessary. If the computer noise has been introduced before the signal leaves the cable, then balanced cables won't help.

In the UK, it's a total postcode lottery. I'm extremely lucky to come under NHS Highland, but have the nearest hospitals in Glasgow. I've never had to wait for any treatment, and of course in Scotland all prescriptions are free (but we pay a bit more tax for it!).

The switching issue is something that is more likely to occur on offline UPS's, or less so on line interactive UPS's. There's three types of UPS's: Offline UPS (cheapest) - power comes straight from the mains until it detects a power drop, then switches to battery. The time taken to switch can cause power glitches in connected equipment, or as you say major damage in some cases. Line Interactive (a bit more expensive) - basically the same as the offline, but the switching mechanism is much more complex. Some will generate their own clean sine wave when connected to the mains, but they still have a small switch-over window from mains to battery. Inline (most expensive) - This type of UPS always runs off the battery, so no switching occurs. It generates its own clean sine wave. A separate circuit charges the battery in the background when the mains is providing power. If the power goes up and down, all it's doing is switching the battery charger on and off, and won't affect the power coming from the battery powering your equipment.

FWIW, most of the "computer noise" problems I've had have been on laptops. It's only a very minor problem for me on my desktop - i.e. it's very quiet, but audible enough during silent sections to annoy me. I don't use my UPS in any way as a backup power source. With the amount of gear its powering, it only provides around 5 mins backup power in any case. The main reason for getting the UPS over a power conditioner, is that it generates its own clean 240V supply from a battery (which is constantly charging in the background). This means regardless of how "dirty" your power source is, it's guaranteed to supply a clean 240V signal, or no signal at all. A power conditioner may clean up a power signal, but it's still physically connected to the mains, and it won't protect you against brown-outs (i.e. the voltage drops down to 50V or so), which will kill HD's.

Personally, I find no significant advantage in using of balanced cables for output signals, unless your cables are significantly long. It makes much more of a difference for input signals, and an absolute must for low input signals such as mics. There was a link to an article on a related thread explaining the difference/advantages between balanced & unbalanced cables, and it basically said the same thing.

Seriously, try a ground loop isolator between your audio interface and your powered monitors. It totally solved this for me, and it's only £6 https://www.amazon.co.uk/AV-Link-Ground-Loop-Isolator/dp/B000NVWB9O/ref=sr_1_4?ie=UTF8&qid=1552292358&sr=8-4&keywords=ground+loop+isolator

I got a fairly expensive inline UPS - got it 1/3 price at £250 as it was being discontinued. It's one that generates it's own clean 240V sine wave, and always goes through the battery - so there's no switch over on power loss. I do notice a reduction in hiss (i.e, there's practically no hiss now). I used to hear a short buzz/click when the freezer switched itself on and off, but not now. It does absolutely nothing for noise generated by the computer though.

Try CTRL + SHIFT + dragging the bounced audio on to a fresh audio track.