David Baay

The forced MIDI channel setting on the track just re-writes the channel of recorded data on playback as it's sent to the output port. It would have no bearing on whether Find/Change (or any other editing operation) is successful. One of the keys to Select by Filter and Find/Change is that you have to start by selecting all or part of the clip to be processed. It's likely you didn't have the clip selected the first time.

Late to the party, but it seems highly unlikely the coding of the new build itself changed anything. More likely the installation process or some other event returned some config setting to default. My guess based on the symptoms would be that you previously had 'Remove DC Offset During Record' enabled under Preferences > Audio > Playback and Recording. Try enabling it now, and see if you get a similar result.

UseHardwareSamplePosition tells Sonar to use the position reported by the driver to determine the Now time rather than just keeping track internally. I don’t know details of the implementation but there can be conflicts

I can't reproduce this after multiple attempts with loop/punch at differet places in the timeline with different lengths (not always snapped to beats) and at various tempos. I still suspect this is related to your particular interface and driver mode; what is it? Maybe try enabling/disabling UseHardwareSamplePosition in Preferences > Audio > Config File (AUD.INI), depending on what is currently set. You should also try re-booting and/or power-cycling your interface if that isn't happening regularly. Your original post said offset was 7 measures while your last screenshot shows offset is two measure or less if I'm interpreting correctly. Is it that variable?

In the past mis-located recordings have sometimes been related to metronome count-in and/or specific interfaces. Try disabling count-in and let us know your interface and driver mode.

Like other MIDI controller types, sustain messages have a range of 0 (full up) to 127 (full down). Basic pedals just have an on/off switch that causese the keyboard to generate one message or the other when the switch opens or closes (there's no convention for which state generates 0 or 127; implementations from different manufactureres vary). A continuous pedal presents a variable resistance to the keyboard that allows it to generate intermediate values to produce partial damping in an instrument that's capable. Continuous sustain messages can complicate the editing process, and in a lot of musical contexts the difference between continuous and on/off values isn't really noticable even in an instrument that supports it. But thinning continuous sustain messages to only 0 and 127 is a non-trivial task; (I might have written a CAL to do it at some point). The easier thing to do would be to configure the keyboard to produce only on-off values if possible or get a pedal that only switches.

It's basi cally a map that does nothing except cause CW to show notes in the drum pane (with the MS buttons) instead of the regular notes pane. Incoming notes go out to specified port with no change to their note number, channel or velocity. It's possible I created my "Passthru" map at some point, but I thought one came with the default installation...?

You can also add a Passthru drum map to the track to get mute/solo buttons on the drum grid.

Plugins with large lookahead buffers can cause this.

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.