David Baay

It's not so much that the project "forgets" as that it simply can't restore an I/O assignment for a port that's doesn't have a driver loaded in the operating system; "Connected" assumes powered up with the driver successfully loaded.

Would be good to mention which plugin for future reference.

The project should have come back up with the correct assignements after the connection was restored unless you re-saved it with the incorrect assignments. The input assignment of a new MIDI/Instrument track should remain None unless you have Alway Echo Current MIDI Track enabled in which case it will show 'Omni' when focused and switch back to None when note focused. But if you manually enable Input Echo without first assigning a port, it will default to All Inputs and stay that way; it's not possible to assign a default.

I was just conincidentally experimenting with this setting on a new laptop with a new interface (Behringer UMC404HD) a day or two ago, and encountered no issues. I also just now switched from MME to UWP MIDI with no issue.

It will work much better if the project tempo is matching the averge tempo of the clip. To achieve this, count out to 8:01, snap the Now time to that transient, and use Set Measure/Beat At Now (Shift+M) to set that measreand beat which will reset the initial tempo to the average of the clip.

I'm sure you know that if it's a synth part, you would be better off just working with the MIDI, but I gather the synth audio is just for testing. A couple of thoughts: - If the timing is basically good and it's just about tightening up the slow drift in tempo from measure to measure, I would focus on just the down beats or at most the beat markers. - After using the Threshold and/or Resolution to knock out the majority of false-positives, you can right-click the grayed out 'stubs' of disabled markers and re-enable them as necessary. If you're just doing this for downbeats, there shouldn't be that many. Similarly you can right-click and disable any markers that the Threshold didn't knock down. You can also Promote markers individually or in groups as you go so that that changes in the Threshold don't affect them.

Blue screens are generally caused by hardware/driver issues, not software. First thing I would do is run MemTest on your RAM. If it's a desktop, open it up, blow the dust out, make sure the CPU heatsink, RAM and other cards are firmly seated. If any hardware/drivers were changed (music-related or otherwise) around the time you first started having issues, consider reverting. A failing/sagging power supply can also cause blue screens, but is harder to diagnose and usually shows up as a boot failures first.

Depends on the situation and the goal. If it's already in sync with the timeline, try using the Resolution setting as well as Threshold to disable false-positive markers. If the material has a lot of sustained tones with relatively weak or 'smudged' attacks (e.g. strummed acoustic guitar) try gating a copy of the audio and using that for transient detection. EQing may also help. If it's a mix and you just want to extract the beat, you might try using a stem separation tool (like the one in Next) to extract the drums and get the beat transients from that and Apply them to the mix.

I saw your post about this in another thread today, and thought there might be more direct way to do this in Cakewalk/Sonar using the included Channel Tools plugin. I gave it a whirl on a guitar track and it seemed to do the trick: - Without converting the mono track to dual mono, add a Send to an aux track, leaving it default Postfader. - Add Channel Tools to the Aux, Invert the right channel, click the Pre button in the Delay section and enable the Link button in the Delay section. - Start playback and start turning up the delay. - Initially the right chanel will be completely nulled by the phase inversion and will fade in as you raise the delay. The stereo image will change pretty radically over a useful range of about 2-20 ms. Above that you start to hear the delay more distinctly but that can be useful up to about 50ms above which it start to become an echo. Adjust the level of the Aux track (or send level) to taste. In addition to saving time and not having to use 3rd-party plugins, the beauty of this approach is that any edits you make to the original track are automatically reflected on the Aux track without having to do anything, and the postfader send ensures the levels remain proportional. But if you want to EQ or add FX independently, you can make the send Prefader and group the track volumes or whatever works. There might also be value in unlinking the left and right delays and tweaking them individually or in playing with the Mid-Side controls and channel width/panning, You migth also consider creating a "Stereoizer" bus and sending multiple tracks to it (the same can be done with an Aux track of course).

Select the clip, go to the Clip Properties tab of the Inspector, change the Time Format to seconds and check the Length.

This assumes you're willing to consolidate all clips, lanes and comping, and destructively render all clip-level edits: slip edits, fades, mutes, stretches, looping, clip automation, and clip FX.

Yes, you can use Quick Grouping to freeze multiple tracks.

I found some of the same issues putting two mono EIs in series on a mono track - mainly delay compensation not working correctly but also some other undesirable side-effects. I'll send a report.

This will just get you the left channel of Mix 1 and the right channel of Mix 2. Panning does not collapse a stereo mix to mono, it just reduces the level of the opposite channel until it's gone. If you're really wanting two mono mixes, you'll also have to set the Interleave button on the two MIX buses to mono. Obviously, it's not possible to preserve the stereo image when the output is a mono channel, but the information in both channels of each mix will be preserved by switching the buses to mono.

I can confirm you can now just change the I/O assignments of an existing External Insert to use one channel of the pair, and the other channel will be freed up for use by another EI instance. However there may be issues trying to use the two mono paths in series on the same track. I encountered some issues that I would need to investigate further.

The downside if that approach is that you can end up with lots of redundant copies of .wav files that are carried forward with each version. To me, that's messier than having 20 .CWP files (that can be easily sorted by Last Modified date) in a common project folder, referencing a common audio folder.

The recording is likely being over-compensated for record latency due to the driver mis-reporting the value (in samples) to Sonar. You will need to determine the appropriate a negative offset to be entered as Manual Offset under Preferences > Audio > Sync and Caching. The best way to do this is by temporarily disabling the Use ASIO Reported Latency checkbox (zeroing any Manual Offset that might already be entered) and using a utility like CEntrance Latency Tester to measure the Actual round-trip latency via a patch cable looping an output back to an input. Then subtract the Reported value from the Actual value to get the Manual Offset (again, this will be negative in your case because the driver is over-reporting the latency). Enter that value and re-enable User ASIO Reported Latency, and you should be in business. This is a newer utility that does the same with a nicer, more flexible interface than CEntrance: https://oblique-audio.com/rtl-utility.php

I can't immediately repro that which means the Bakers will need to have a copy of your specific project to investigate.

I would guess the Edit Filter of the lane is still set to 'Audio Transients'. The beveled corner on the clip shows it hasn't been bounced.

With the track armed, you should see live input signal in the meter of the track to which you're recording with the transport stopped. If not, the Input of the track is not assigned correctly or your interface driver is misbehaving and needs to be re-initialized by power-cycling or reboot, or the interface internal routing via hardware or software mix control is fouled up. If you can see the signal in the record meter but can't hear it with Input Echo enabled, I can only think that some other track/bus is soloed.

Either don't create a new folder for each version or create the top-level folder outside of Sonar using Windows Explorer before you save the first version and navigate up that folder level when saving a new version. I generally take the first appraoch and just Save As new versions with a new file name. That way, the different versions can share some audio files that are common to all versions and also reference newly recorded/bounced files that are unique to a particular version, but all saved in a common Audio folder. But I do have higher level folders that contain, for example, all projects recorded with a particular audio-MIDI interface setup or all Improvisations started in 2025, all Bug Demo projects, all Test projects, etc.

I would tend to suspect this is what happened. The 'M' got stuck or inadvertently pressed during playback and the project was re-saved or auto-saved before it was noticed.

Yes, better. And the vocal placement is quite a bit better. Previously, with heaphones, it sounded like it was floating in the middle of my head while the instruments were in a different space, more out front. I didn't recognize it as excessive width, but that must have been it. Nice work.

Interesting. Vocal placement in the sound space is very... spacy in headphones, and it has a super-phasy and breathy sound. Nice little tune overall. Some timing weirdness with a couple pianos notes around 1:13. I'll probably have to try this out with one of my own songs for fun sometime.

Is the original track mono, and the export is stereo from the Master Bus/Main Outs/Entire Mix? I've just verified Youlean shows a loudness 3dB lower on a mono source track than on the stereo bus that it's playing/exporting through though both show the same peak level. Right offhand I'm not sure whether this is a quirk of Youlean or just the nature of loudness measurment when going from a mono track to a stereo output, but I would guess the latter. EDIT: Google says it is indeed the case that perceived and measured loudness will increase 3dB going from mono to stereo, and I verified that using a -3dB Center Pan Law (Preferences > Audio > Driver Settings) will result in Youlean showing the same LUFS on the bus as on the track.