David Baay

For the time being, onboard sound is just fine. And it won't affect the quality of exported audio, as this is a purely 'offline' process, not involving the soundcard/interface. For best results, set 'WASAPI Exclusive' driver mode in Preferences > Audio > Playback and Recording, and set the Buffer-Size slider in Audio > Sync and Caching as low as you can, without getting pops/crackles or dropouts on playback. P.S. Welcome aboard. Lots of old guys around here ready to help with any questions/issues you might have, but it does help to read about the basics of MIDI, Audio and DAWs in Cakewalk docs and around the Web at large. Youtube is a great resource.

A driver is a software component that an application uses to communicate with and control hardware. Every audio interface generally comes with a driver and sometimes a software mixer that should ideally be installed before connecting the interface. The latest version is usually available on the manufacturer's website. Make sure you have the most current driver installed for the Behringer, and if you have ever installed a generic ASIO driver like ASIO4All, uninstall it.

In case it is plugin-related, you might try disabling 64-bit Double Precision Engine in Preferences > Audio > Driver Settings.

Utilities > LANDR Online Mastering If it's not installed, you'll be prompted.

Go to Views > Event List, pick the imported MIDI track, and look for orange controller messages, especially SysxData. Patch Change and Modulation events. You can delete them, or copy only notes to another track, using Select > By Filter. But it will be a good idea to look up any controllers you find first to understand what they do.

I run two 2408mkii interfaces on a PCIe-424 card, and they work beautifully at very low buffer sizes. In addition too uninstalling ASIO4All, I recommend you go into Preferences > Audio > Configuration File, and click Reset Config to Defaults. This makes a backup of your AUD.INI file, and replaces it with a default one. Then verify Playback and Record Timing Masters and other audio settings. If problems persist, you should check your system's DPC Latency: https://www.resplendence.com/latencymon

I'm pretty sure the instrument auditions are generated in real time from internal MIDI sequences as opposed to being recorded audio samples. For fun, I once transcribed the sequence used for the fretless bass audition, and it sounded pretty much identical. Imported tracks may contain controller messages that alter the sound of the assigned instrument. So you'll want to start by checking that. If you can share the MIDI somewhere, it would help to understand the difference you're hearing.

- Ctrl+A to select the whole project. - Alt+A to open the Audiosnap palette. - Click the Enable/Power button. - Select 'Auto stretch' in the dropdown to the right of Clip Follows Project. - Click Clip Follows Project button. - Manually tweak tempos in the Tempos view by clicking above or below existing tempos (not possible to drag) in the map or editing numeric tempos in the list pane. - MIDI and audio will both follow the changes and remain in sync with each other. - When you're satisfied with the timing, save a copy of the project as 'Live Stretching' version. - Bounce to Clips each audio clip independently using an appropriate 'Offline' stretching algorithm to render the stretching permanent. - Save that as a 'Rendered Stretching' version so you can easily revert to the original, unstretched version, or to the live stretching version for more tweaks if needed.

So, if it plays fine, at what point does the crash occur? Sounds like you might have time immediately after loading the project to switch the audio engine off, and remove the offending plugin. If not, Start the project in Safe mode by holding Shift while you launch it, and don't let it load the plugin that's causing the problem.

I don't know what the default is for the Basic Workspace, but it looks like I/O is selected in the Track Control Manager at the top of tracks pane. Change it to Mix or All to see Vol, Pan and Gain/Vel+.

Confirmed selection range is lost on delete in the current relase, and was not in SONAR 17.10. Can't say when it changed, but does not seem helpful so probably not deliberate. Maybe related to introduction of selection markers.

No, I never said that. I may only have ended up with a dozen or two custom bindings, but that doesn't mean I haven't spent a fair amount of time in there over thirty years. Not sure what you mean by 'get deeper', unless you mean create/alter many more bindings. If you have a recipe fo demonstrating a specifiic issue, you should share it. Thank you. Always glad to help.

Interesting site. Of his four conclusions about keybindings, I only minimally agree with the first two (TERRIBLE defaults, Can't serach all 'areas'), and absolutely disagree with last two (slow and prone to corruption) . - Everybody has different opinion about what default bindings are needed and make the most sense. I've only had to make a handful of changes to customize it to my needs and preferences. But I have the benefit of having worked with Cakewalk through it's entire history, and learned new keybindings as they were introduced/changed, so they seem mostly natural to me. Ultimately, I don't see how any software with many hundreds of functions is going to have a completely satisfactory mapping of those functions to a 101-key keyboard. - The division of bindings into different 'area' is a double-edged sword. It can sometimes be difficult to find the exact name of a function in order to bind it, but in general, it seems to me that if you're aware of the existence of a function that you want to bind, it should be pretty obvious in which area you'll find it and what keyword will bring it up. - The UI doesn't seem at all slow to me. The writer says he hates Windows; maybe in part because his system has performance issues. - In 30 years, I don't recall ever having my keybindings get 'corrupted' - possibly another example of issues with his Windows environment. I agree the OP needs to make specific recommendations for changes he would like to see.

Looks like a simple doc bug. I can't see that it would be possible to load a mix scene with a track template into a project structure that didn't match the mix scene.

I'm curious to know whether the issue reproduces with Session Drummer or TTS-1 or SI Drums in your environment.

This happens when you Punch-record with Mute Previous Takes enabled. You can unmute all the notes in one go by sweeping through the top half of the clip with the mute tool (sweeping in the bottom half mutes) .

At the mixing stage, input monitoring should no longer be enabled on any audio tracks, so feedback should not be possible. But if you're using onboard sound as the audio interface, make sure the Input in the onboard mixer app is set to Line or Mic In rather than 'What U Hear' or 'Stereo Mix' which monitors and records from the output rather than the input which can lead to feedback.

How many recording moguls does it take to screw in a lightbulb? Recording moguls don't screw in lightbulbs; they screw in hot tubs.

Right-click the record button in the transport module, and choose Sound on Sound. I also recommend choosing Create New Lane for Take Behavior if it isn't the default.

Hmmm... somethings's not right, but I'm not sure what. I just double-checked using Session Drummer with a long sample loaded into the kick, and with a long-ringing cymbal in a hardware synth, and both play samples well past the end of the triggering cell in Step Sequencer. Even turning the duration down to zero had no effect, which is expected since a drum synth pays no attention to the note-off as I said.

A hardware drum machine? Whether hardware or software, a drum synth should play the full length of a one-shot sample regardless of when the sequencer sends a note off. If you trigger the sample from the drum synth's interface, does it play out? Any chance the note number of the kick sample (or the synth channel hosting it) is part of a choke group with some other sample being played soon after it?

Thanks for the reminder, Noel. I keep forgetting that feature exists.

The 'Internal' clock source setting in Cakewalk means the PC system clock, and is only applicable to pure MIDI projects driving direct-monitored, external hardware synths. Any audio in the project (including the audio metronome) will automatically force the clock source back to Audio, meaning Cakewalk is using the sample clock from the interface drivers specified as Playback and Record Timing Masters in Preverences > Audio > Drivers. So the first thing to do is make sure those are referencing the correct interface drivers. The Hotfix mostly affected systems using WASAPI driver mode, but you should definitely install that in any case as Steve suggested. Sometime the quickest way to get the audio setup into a happy state when you've been tweaking things is to rename AUD.INI, and let Cakewalk build a new one with default settings. If that fixes the problem, you can compare the renamed one to the new one to see what changed. C:\Users\<username>\AppData\Roaming\Cakewalk\Cakewalk Core Sonitus FX are DX plugins whereas the ones you're having problems with will likely all be VST plugins, so there's a key difference there, but I can't say I've heard of issue affecting only VST plugins before.

Steinberg added a feature to the ASIO 2.0 spec called Hardware Direct Monitoring that allows DAW software to enable direct monitoring through interfaces that support it, but it was never implemented in SONAR/Cakewalk. As I recall, CTO, Noel Borthwick, concluded that the spec wasn't well-defined enough and/or it wasn't widely enough supported by hardware to make it worthwhile. I'm not sure how widely supported it is today. You could probably set up remote control of direct monitoring via another piece of hardware (e.g. a keyboard controller or control surface), but there's no mechanism to add a control to the UI in Cakewalk other than Studioware panels as you mentioned.

Clip fades are pre-everything - applied before the audio hits the track input. But automation has optional curve types. So use Volume automation, right-click the attenuation segment, and choose 'Fast Curve' (i.e. a logarithmic curve that initially drops faster).