David Baay

I have often wished some plugins would have automatic loudness compensation in place of a manual make-up gain. I haven't checked the plugins you mentioned, but it seems like it would require a pretty big lookahead buffer to do the job properly - meaning a lot of plugin delay - but at the mixing stage, that shouldn't be a problem.

Hardware or software synths? If soft, which one, and is it inserted in the Synth Rack or in the track's FX bin? If it's in the FX bin, try moving it to the Synth Rack. If hardware, is it connected by MIDI DIN to an external interface or by USB MIDI built into the synth? Are there any FX plugins in the project with very high plugin latency that triggers Plugin Delay Compensation? Cakewalk/Bandlab's LP plugins induce a lot of DPC at med/high precision settings that can cause MIDI buffering issues - usually dropped note ons, but I suppose dropped note offs or controllers would also be possible. MIDI FX plugins can also potentially cause problems, even if they're on some other track. Also, just to be clear, are we talking about real-time performance by input-echoed MIDI or playback of recorded/imported MIDI? The Prepare Using Buffer only applies to playback.

A lot of users find this to be unexpected/unwelcome behavior, but it's working as intended. Paste Special is the how you access paste options, and changing an option persists for all subsequent pastes until you change the option back with Paste Special. Personally I find more often than not that I want the changed paste options to persist for at least one more paste so, in the long run, it saves time.

This has been my main use of Rewire. Notion had other issues that prevented my continuing to use it with any frequency, but I did have the Rewire functionality working realtively smoothly within its limitations (e.g. Notion's playback cursor not following changes in CbB Now Time when transport isn't running).

This typically happens when you enable Input Echo before selecting a specific input, and Cakewalk defaults to All Inputs - Omni (i.e. all channels of all inputs). Once you've selected a specific input, it shouldn't change.

Cool. Glad to help. 'Online' means real-time processing during playback. 'Offline' is what's used when you Bounce to Clip(s) to destructively apply the stretching , creating a new audio file. In general, the offline algorithms are more sophisticated, and will sound better. But in some cases, Groove Clip yields the most transparent results.

Have not double-checked, but usually if you have multiple clips selected, any change to properties will affect all of them. If they start out with different values, the property show '(multi)'.

It just occured to me that a recent change to AS is that the default stretching algorithm is now Elastique instead Groove Clip. In my experimentation with this algorithm when it was first released, I found that it could produce weird timing in some cases. Try switching to Groove Clip or one of the Izotope algorithms.

What changes did you make using Audiosnap?If you're using it to have the overall clip tempo follow changes in the project tempo, make sure the Follow Project mode is 'Autostrech'. (drop-down to the right of the option in the Audiosnap palette) If it's anything else, transient markers will get snapped to the grid with a probable bad result unless they've been carefully 'massaged' into the correct positions for that purpose.

For future reference (when mixdown is not hanging), the easy way to bounce audio clips together while preserving the original separate clips (with or without active clip/region FX), and keeping everything them in the same track where you might already have special output routing, sends and FX in place is to use Flatten Comp.

Most often clip selection failure is due to having the track's Edit Filter set to something other than clips, but that wouldn't change with a restart. Not sure I understand the bounce to clips issue. You're saying the new clips is created on top of the old clip instead of replacing it?

Yes I understood that. I was responding to chris.r's comment that the alternative required a "multitude" of mouse clicks. Really it only requires one more on top of dragging the instrument into the project, and then a couple more to delete the abandoned MIDI track. If you were to drag in all the instruments first, in the corresponding order, it would only take one click to drag all the MIDI clips to those instrument tracks, and two clicks to delete all the abandoned MIDI tracks - an average of 0.188 extra clicks per instrument for a 16-track MIDI file. ;^)

It's kind of a superfluous option. For the most part, CbB does not allow same-lane overlaps, but there are some bugs that will let it happen, and you'll see more of them if you don' keep that option checked.

I think you must be thinking of a different DAW or different situation in CbB. I just double-checked, thinking something might have changed since I last tried it, but there was no prompt about the length. The project opened with no complaints, but the audio is truncated as described. The clip runs 4 measures, but audio is a flat line for about the last five 16ths of the last measure (44.1/48 * 4:01:000 > 3:03:672).

For future reference, the Gain control in an audio track controls the input level to the track from playback of an existing, recorded or imported audio clip, but has no effect on live input signal as noted.

There's one problem with converting the sample rate of audio files outside of CbB that was pointed out to me when I suggested this solution in the past. The length of the audio clip in samples is stored in the project, so when it reads in data from a higher sample rate file, the clip will be truncated to the original number of samples, and there's no way to get the clip to reference the entire file, short of importing it at the new rate.

It's not the same compensation as for external audio input, but there is a bug that causes soft synth audio to be laid down early by exactly one ASIO buffer when the synth is driven by existing MIDI in the project. (Note that the opposite problem occurs when recording a soft synth driven by live, real-time MIDI input, but is expected because the audio isn't compensated for MIDI latency or the fact that the audio you're playing along with is delayed by output latency). Workarounds are: - Set a very small ASIO buffer (e.g. 32 samples), and ignore the error. - Freeze or Bounce to Track instead of recording. - Put a send on the Synth track to an Aux track, and record on the Aux track which won't have the compensation mis-applied.

Hmm, while you can achieve the same typically in more than one way in CakeLab and having to go through multitude of mouse clicks of course, for the workflow this seems to be a great feature request indeed . While I agree this could be a convenient feature (though not at all 'baffling' to me that it isn't already), I wouldn't call dragging the MIDI clip to a new Instrument track and deleting the unneeded MIDI track a 'multitude of mouse clicks'.

Shift+A shows automation lanes, and will create a Volume lane by default as a convenience since that's the most often used type of automation. If you want something else, you can change the type using the drop-down in the lane header or by right-clicking the envelope, and choosing Assign. If you don't want any automaiton on the lane, don't Shift+A to show automation lanes in the first place.

How are the instruments assigned to outputs within the synth? I'm not familiar with Acoustica, but TTS-1 has 4 outputs. If all instruments are assigned to output 1 (the default), it wont be possible to mute that output if some track driving another instrument using the same output is soloed; the solo will override the mute, and keep the output active. Ideally each instrument in a multi-timbral synth should be assigned to a dedicated output with a separate Synth track (i.e. an audio track with synth output assigned as input) hosting each output. This also allows you to to apply different automation, FX and Send destinations/levels to each instrument.

Late to the party. Very cool, Noel. Listening to the performances now; as we '70s kid would say... Eeeeexcelent, Man! Cheers, Dave

Yes, as demonstrated in OP's other thread on the subject. Linked here for the benefit of future searchers:

I suggest you capture a copy of AUD.INI before and after re-configuring after successful launch with default AUD.INI, and see what all has changed. Then try reverting changes one at a time and relaunching to find the culprit.

Audionsnap has Extract and Apply Groove functions that work with both audio and MIDI. But I still don't think this is going to get the OP what he's looking for in terms of a 'flowing intro'.

That's why I prefer to zoom on the Now time. Since I usually want to zoom on or near the start of a note, snapping the Now time to the nearest beat and/or tabbing to the transient, and zooming gets me that spot centered in the tracks pane without having to take care where my cursor is. And if the exact spot I want goes a little off screen at high zoom levels, it's a simple matter to scroll left or right.