David Baay

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.

I recalled this issue having come up in the past and found mentions on the old forum going back to at least 2012. Here's one from 2017 before the Gibson shutdown: http://forum.cakewalk.com/SONAR-Track-Soloing-Remote-Control-not-working-m3561832.aspx If it was working more recently, it may have regressed, but possibly there's some other contributing factor that can make it fail. The first thread I saw suggested that it could be intermittent and work for a time after re-learning.

That's interesting. I was actually trying to reproduce the issue earlier, coincidentally using Sonitus Reverb, and could not. How big is that transient in your screenshot? Looks way too big to be from "a tiny bit" of Sonitus Delay, but maybe the track scale is zoomed..?

What it doesn't show is that if you do a loopback recording of the MIDI, the re-recorded notes will actually be 10 ticks late in the 192PPQ timeline as expected. So bringing an old project into Sonar isn't really a problem unless you notice that the displayed offset has changed, and change it back to what is was on CbB. Then you'll actually get the wrong musical result. Bottom line: There is an issue to be fixed here, but it's a display issue.

That's just because the track isn't scaled properly in Sonar to show all the notes.

Try leaving the .wav extension off. Sonar will add it automatically. I haven't yet checked the thread that Amberwolf linked, but I don't get that when overwriting previous exports.

I believe the issue here is that the Always Stream..." option will only flush the buffers at the end of live playback. Fast/offline bouncing may not allow that to happen so there's still a non-empty buffer of audio waiting to be streamed or rendered to file when you next the start playback of do another bounce. I thought this had actually been addressed quite a while ago and haven't noticed it myself recently so wonder if it might be related to misbehavior of specific FX plugins that are not getting flushed...? Whatever the cause, one way to handle this on tracks is to freeze the track with some time included for the FX tails to play out in the Freeze Options. For bus FX, having a volume envelope that runs to the end of the FX tail (or overrides it) will do the trick. Since so much modern pop/rock music ends with a fade-out this usually happens automatically if the mastering and final fades are being done in Sonar, but if you're exporting stems or raw album tracks for mastering by a third party, that might not be the case.

That's not exactly what I'm seeing or saying. Sonar doesn't change the offset, musically, it just always references the entered/displayed value to 960PPQ. If the project originates in CbB, the displayed value will increase from whatever was entered by a factor of 960/x, where x is the PPQ setting of the CbB project. The actual absolute delay that results from that at any given tempo will be the same. If the project originates in Sonar, and you change the project clock, you have to continue to enter tick-based Time Offsets based on 960PPQ,

There is confusion here becase he shows swithing timebases for the Offset, but the thing to note is that the 10-tick offset in CbB becomes 50 ticks in Sonar. What was not mentioned is that the project clock is 192PPQ. When you open the 192PPQ project saved by CbB in Sonar, the offset value is re-referenced to 960PPQ and changes from 10/192 to 50/960. If you actually re-record the MIDI via loopback with no transmission delay in Sonar, events will correctly land 10 ticks later in the 192PPQ timeline, but theTime+ field is incorrectly showing 50 ticks. Sonar is not honoring the 192PPQ project setting when setting Time+ in ticks. A reported back in FogBugz days that this issue also applies to setting quantize and snap values in ticks when the project clock is something other than 960. To get the right musical interval, you need to keep thinking in terms of 960PPQ.