David Baay

Short answers: - Any way you slice it, this is not a job for a DAW beginner or the faint of heart or the impatient. As mentioned, Youtube and the Ref. Guide are your friends for learning the details of how these operations work. - Use Set Measure/Beat at Now (Shift+M) on every downbeat of the click track (using Tab to Transients to snap the Now Time) to align the timeline to musical measures with tempo changes that Sonar will calculate and insert fo you. If timing within measures is not tight, you can set additional beats of the click track or even all beats if you want. - Once you have the tempos dialed in, you will need to enable Audiosnap on all the audio clips, set the follow option to Autostretch and enable Follow Project. - Decreasing the tempo of the first section is a simple matter of selecting the relevant range of time in the Tempo Track and using the Tempo Offset function to lower them by a percentage. - Ramping up the tempos in the latter part will reuiqre a more manual approach as Sonar does not currently have the doeing that with existing tempos, Inless you're willing to delte all the tempo, and replace them by Inserrt Series of Tempos to create a perfectly linear increase. - Converting audio to MIDI can be done either before or after making the tempo changes, but is much easier said than done in any case and will require musical knowledge and a good ear as well as skill with CbB to get anywhere near satisfactory results. It will also require a 3rd-party plugin like Melodyne to convert polyphonic parts, and the initial results will be rough and require a lot of polishing to sound anything like the original, depending on the instrument and the material. CbB has a Drum Replacer function that will help extract basic kick, snare, toms and hi hat patterns, but if the drumming is complex, it will take a lot of work to get it right. Incidentally, Melodyne can also be used to extract a tempo map, but I prefer the result of doing it manually with Set Measure/Beat at Now.

At the time I was investigating this, I was unable to find a hi-res MIDI file to play with so I created my own by copying the velocity lane in the controllers pane of the PRV to a CC88 lane and sliding the resulting controllers 1 tick earlier. If the original MIDI perfromance was played live, the velocities will be variable so the CC88 values will as well. Without getting too far into the weeds, my conclusiion was that hi-res velocity was of no value when playing Pianoteq in particular because the inherently variable response it produces as a phycially modeled instrument swamped the much smaller level of variability due to the finer gradations of velocity. Moreover, I'm not convinced that any keyboard player on the planet has the level of control and repeatability necessary to take advantage of higher velocity resolution. There's an argument to be made that even uncontrolled/random variation at a fine level might add some "life" to a piano or synth patch that might otherwise sound mechanical due to producing the exact same response to the same velocity every time, but some instruments already address that by "round-robin" samples or physically modeled randomness as in the case of Pianoteq, and if they don't, they probably won't repond to hi-res MIDI anyway!

Yes, for the most part. You just end up with an Event List that shows a CC88 message right before every Note event. But Sonar currently ignores Note Off velocity and always sends a neutral value of 64, so it will record the CC88 controllers that a Hi-Res-capable keyboard sends right before a Note Off and show them in the Event List even though it doesn't show separate Note Off messages or allow editing them other than indirectly by changing the duration of a Note event.

Just to clariify, MIDI 2.0 Velocity offers 65,536 (16-bit) possible values with the value encoded in the the Note On/Off messages as it is in MIDI 1.0. What's commonly referred to as "Hi-Res Velocity" (what Vidal is offering) is just a MIDI 1.0 convention for using a CC88 message sent immediately before a Note On/Off message that can be interpreted by the receiving instrument as dividing the velocity value by another 128, yielding 16,384 (14-bit) possible values.

I have experimented with high-resolution velocity, using Pianoteq which can respond to it in theory with microtonal and loudness changes. Putting aside whether the variation it produces is musically meaningful, I think the main obstacle to working with the CC88 implmentation in a DAW is that the controller messages are not tied to the note events so special care needs to be taken to keep them together when editing.

Unplugging headphone/speakers from an onboard audio jack can signal the PC to unload the driver, but it sounds like you're talking about the headphone output on an ASIO interface, is that right? In that case, I would guess that the issue is that headphone output is wired for a stereo TRS (Tip/Ring/Sleeve) phone plug while your single un-powered computer speaker is probably a mono TS (no Ring) phone plug, and meant to be plugged into a "sister" speaker that has a stereo amp in it. Something like this: https://us.creative.com/p/speakers/gigaworks-t20-series-ii You might get one channel of the headpohne output from your unpowered speaker but with impedance/power mismatch, the level is probably near zero.

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.