msmcleod

Have you tried disabling the High Definition Audio Controller driver on your video card? According to Microsoft: https://support.microsoft.com/en-gb/help/2832829/0x00000019-stop-error-when-you-try-to-shut-down-a-windows-based-comput#:~:text=Not all "Stop 0x00000019" errors,sys driver.

Send an email to support@cakewalk.com - they should be able to get it sorted for you.

It sounds like your master.ins has somehow got corrupted. It's located in %APPDATA%Cakewalk\Cakewalk Core

*.smf isn't supported at all. *.midi is treated the same as *.mff

IMO the music one is worth it for SoundForge 12, which at $10 is a steal. AFAIK SoundForge 12 is 32 bit though - so something to bear in mind.

The transport buttons on the M-Audio Oxygen only work with DAW's that support DirectLink, which CbB currently doesn't: https://m-audio.com/kb/article/1688

This has been fixed and will be in the full release.

"...Cakewalk (browser) on my mac..." @Maurice - are you sure you're using Cakewalk? This is Cakewalk: This is BandLab:

Exactly the same happened to me today.... twice.... a complete waste of 3 hours. The first time it failed with a driver issue. So I downloaded the Windows Upgrade utility, shut down and removed all my PCI cards, unplugged all USB devices and tried again. Failed again with the same driver issue. Never did find out what driver was the issue. It rolled me back to 2019.09, but I've just finished restoring my backup , as I don't fully trust the uninstall.

Yeah, I had 2 separate licenses. When I went for the upgrade, it said it would merge both my existing licenses to a single license (with a limit of up to 5 activations), for the price of a SINGLE upgrade from Editor v4 to Editor v5. So yes, two v4 licenses upgraded to one v5 serial number.

CTRL + D duplicates the selected clips. You may or may not want Ripple Edit enabled.

As soon as I hear Jon Meyer / Jon Mayer / John Mayer... I get confused as I never know who they're talking about. There's the Jon Meyer that this thread is about, there's Jon Mayer jazz pianist, and there's John Mayer the guitarist.

I thought I'd post this here in a new topic, so it's a bit more visible than the other Melodyne threads. I've just upgraded Melodyne to version 5. It merged my two existing Melodyne 4 licenses into a single license (which was great, 'cos I only got charged once 😊 ), and it automatically moved 2 x Melodyne 5 licenses to my iLok account. The upgrade was seamless on my Windows 10 boot, and worked in CbB right away (I didn't even need to re-scan). However there are some gotchas... Melodyne 5 does NOT work on Windows 7 - if you're planning to stay on Windows 7, don't upgrade. Your Melodyne 4 installation AND your Melodyne 4 licenses will be removed - and actually, your v4 licenses become invalid at the point you purchase your upgrade Melodyne 5 is 64 bit only. So if (for example) you use an older 32 bit only version of SoundForge, it's not going to work with Melodyne any more.

When clips are cropped, the original clip data isn't lost... it's still part of the clip, allowing you to move the crop point to retrieve the data. The same thing happens when you split a clip - i.e. both clips are essentially copies of each other with different crop points. If you do a "Apply Trimming" or "Bounce To Clip(s)" on a clip, this will remove the original event data. So if the original data is still there, this is probably the cause of what you're seeing.

MixControl doesn't affect what inputs/outputs are available, it only affects what is routed to those inputs/outputs. For example, I switch between two routing configs within MixControl: Config 1: Analog inputs 1-8 to the first 4 stereo devices, ADAT inputs to the last 4 stereo devices ( I use this for tracking ) Config 2: Analog inputs 1-6 to the first 3 stereo devices, SPDIF to the 4th stereo device , and ADAT inputs to the last 4 stereo devices ( I use this for mixing, so I can use the SPDIF as an outboard effects return ). But what you're seeing in preferences is correct. The preferences dialog always shows the stereo inputs & output devices. So stereo input 1 is comprised of inputs 1 & 2, stereo input 3 is comprised of inputs 3 & 4, and so on. However, when you're choosing a track's input or output, it's at that point you decide whether you want stereo or mono. If you're recording mono drum mics (and assuming the default routing within MixControl), then the inputs will be as follows: Left Focusrite USB ASIO Mic 1 = Channel 1 on 18i20 Right Focusrite USB ASIO Mic 1 = Channel 2 on 18i20 Left Focusrite USB ASIO Mic 3 = Channel 3 on 18i20 Right Focusrite USB ASIO Mic 3 = Channel 4 on 18i20 Left Focusrite USB ASIO Mic 5 = Channel 5 on 18i20 Right Focusrite USB ASIO Mic 5 = Channel 6 on 18i20 Left Focusrite USB ASIO Mic 7 = Channel 7 on 18i20 Right Focusrite USB ASIO Mic 7 = Channel 8 on 18i20 On my rig, my keyboard rack is connected to inputs 3 & 4 as a stereo pair. So I would pick Stereo Focusrite USB ASIO Mic 3 to record the stereo pair.

@Promidi is correct - you're selecting everything from 1:0:0 Go up to the timeline at the edge of the green selection indicator, when the cursor changes to an "S", move the selection start to 6:02. The white dots indicate that there's event data further on in that track - in your case at 6:02.

IIRC the old Cakewalk tuner is a 32-bit DX plugin, which won't work in CbB (because it's 64 bit). There's quite a few free tuners out there, that are much better than the old Cakewalk one. I recommend Melda's MTuner, which is part of their free bundle (all the plugins in that bundle are excellent): https://www.meldaproduction.com/MFreeFxBundle https://www.meldaproduction.com/MTuner

First, make sure you've got "Show Mono Outputs" enabled: Then the channels are assigned like this: For drums, I guess you'll probably for the most part be using the mono inputs.

I've currently got Editor, and I'm seriously considering upgrading to Studio. The main reason being that the new Chord analysis feature works across multiple tracks, and takes all of the tracks into account when figuring out what the chords are.

I think the only thing you can do is use your ART as the timing/playback master, and live with the fact that your Shure might be a few milliseconds out. It should be easy enough to nudge the recorded clip once recorded if its timing is slightly out. In reality though, I don't think you'll notice any audible difference in timing. Personally I always advise people against getting USB mics, unless they're ONLY doing podcasts. Most decent mics (and your Shure, being a Shure, probably is a decent mic) have non-USB equivalents, and I'll steer people towards them instead. There's two reasons for this: 1. The ASIO "one driver at a time issue" you've encountered; and 2. With a standard XLR mic, you've got a choice of preamps and interfaces. When you upgrade your preamp/interface, you can still use your mic. With a USB mic, it's also the preamp and the interface, and you've not choice at all. [Edit] - You could use ASIO4ALL or ASIOLink as an alternative way of combining these interfaces. ASIO4ALL will use the WDM mode of both interfaces, and present it to Cakewalk as ASIO. ASIOLink can use ASIO mode for the ART, and WDM for the Shure - presenting them both to Cakewalk as ASIO. The configuration of both can be a bit of a hassle though - ASIOLink especially so.

A CAL script would probably do it... but it would be fairly complicated tho. The split notes to tracks script is probably the place to start, but the logic will be different. The way I would approach it would be to give it a max number of tracks, then go through and individually move the top notes to the 1st track. I'd store the start time/duration of the moved note, so I would skip the others at the same time period (leaving them for the next "round"). Making sure you always get the top note might be tricky, but I guess you could read a maximum of n number of notes and choose the highest one. Then rewind and repeat for the 2nd track, and so on. The logic described above may sound a bit weird... but CAL is no normal scripting language!

FWIW I still use sf2 as my preferred sampling format. My main reasons: 1. They're easy to manage as they're self contained (unlike sfz which has all the samples loose in a separate folder). I can also easily combine many of them into a single sound-font. 2. It's supported by several different players across many platforms. 3. Most players are far lighter on CPU than something like Kontakt. The only thing that I find missing from sf2 is it not supporting release samples, but I've never found this to be a huge issue. Setting the release time on the player usually suffices for me. Sure, if you want massive control over your samples and need scripting, then Kontakt is the way to go... but 99% of the time sf2 does the job. Oh, and I use SampleLord as my preferred player. It's 32 bit so I run it through JBridge in CbB. I also have it running on my SMPro V-Machine.

It's not in the EA release. It'll be in the main release.