David Baay

Set now time where you want the change, and Project > Insert Meter/Key Change. The new meter persists until you enter another one to change it back. If you have long series of regularly alternating meters you can copy/paste special with repeats. Or you can use the Meter/Key view.

Are you saying the VSTis are inserted in the FX bin of the track? If so, try removing them, and inserting by dragging from the browser to the tracks pane or by Insert > Soft Synth, which will put each of them in the Synth Rack separately with audio output going to independent synth audio tracks via their Inputs rather than via the FX bin.

Yeah, I sort of figured you were looking for a more advanced routing and bussing setup than track templates could accomodate. And you would still need to move your clips from their original tracks to their template tracks. So you might as well be moving them to a new project where everything is already set up.

No, project templates can only be used to start a new project; they can't be applied after the fact in any way. But copying the tracks over to to the new project started from template should be as easy as opening both projects' track windows side by side, and dragging tracks/clips to the new project.

So long as you don't re-save the project with the clips gone, the last saved copy will still have them. And even if you did re-save, the actual audio files themselves will still be in project's audio folder, unless they were all recorded in the current session and never saved...? Check Edit > History for evidence of a Delete, Crop or other action that you can undo. Otherwise it's probably just a display issue of some sort. Try opening the last saved copy of the project withouout closing the current one, and if everything is there, you can abandon the current one or Save As with a different name for further investigation.

So just to clarify, for a VST instrument for example, you set the associated audio track/s gain so it's peaking around -18dB? Or do you lower the fader the on the synth audio track? Note that Bob said he sets the combination of instrument Output and audio track Gain to get the desired input level to the track. Ignoring live hardware inputs for the moment since they aren't affected by track gain, and this thread is about soft synths, the Instrument output level will be determined either by a fader/knob in the synth GUI, or - as I described earlier - by the MIDI track Volume. Gain is the knob at the top of the track Inspector or the fader labeled 'Gain' in and audio/synth track header (when showing Mix or All controls). Note that on a Simple Instrument track that hasn't been split into separate MIDI and audio 'Synth' tracks, you have to use the Audio tab of the track Inspector to access audio Gain. Usually when people say 'fader' without any reference, they mean the output Volume fader. What Bob and others are advocating is to get the input Gain level (what you see when an audio/synth track is armed for recording) to some nominal value like peaking at -18dB), and then use the output Volume level to control the level of that instrument in the mix. There is no fixed rule for where to set the track Volume fader to start. The more tracks you have, the lower each one has to be for the sum to stay under 0dB on the Master bus with Gain and Volume controls at unity (0dB). In a DAW, buses effectively have unlimited headroom at the input so, if you want, you can control the summed level by simply pulling the input Gain on the bus down, and not worry about the fact that somewhere in the DAW's floating-point math inner workings, the summed level is theoretically above full scale. But in order to get a good feel for how it all works, it's a best practice to pull your track output Volumes down to keep the sum peaking under ~6dB at the input to the Master bus. More food for thought. I would suggest you close your browser at this point, and go play with all the controls in Cakewalk to see what they do. ;^)

Good to hear. All's well that ends well, right?

If a VST's default output level is consistently lower or higher than you want with audio Gain and Volume controls at 0dB, start by adjusting the MIDI Volume fader. This controls the synth's Master Volume with CC7 (or channel volume of a multitimbral synth if the track has an output CHannel assigned). By default, the MIDI Volume fader is disabled as indicated by parentheses around the default level of 101 (see the MIDI tab at the bottom of the Track Inspector in the case of a Simple Instrument track). When you move it, it automatically becomes enabled, and that value will be sent to the synth to initialize its output level every time you start playback. This is especially important with hardware synths that may have their volumes left at some other level by a previous project or alteration from the front panel. Softsynth states are stored in the project so that won't happen, but it's still good practice to set that initial volume to have a reliable starting point. Conversely, if a MIDI volume control was inadvertently enabled at some point, you can double-click it to reset the value to 101, and then right click and choose 'Disable Control'. Then you can control the synth's volume from its GUI. Also, be aware that if two or more tracks point at the same synth/channel, and both have their MIDI Volume faders enabled, the highest numbered track will have the final say in what that level is, as Cakewalk process the track volume messages (and other MIDI controller messages like Pan) from top to bottom.

Key+ is working with clip playback, but not live input here. Weird because it's not a synth function; it's just the track transposing the MIDI note number. Works if you put a virtual or physical MIDI cable between the track and the synth, but not if you insert a pass-thru drum map. Somehow Cakewalk knows where the MIDI is going when is stays inside the app, and is not honoring key+ in real time. There is at least one reference to it on the old forum, and I vaguely recall running into it myself in the past. 'Pitch' is working here with both playback and live input, but it's an Arp function so the Arp has to be enabled for it to work at all.

'Screenset' is catchier. ;^) There's a module in the Control Bar. https://www.cakewalk.com/Documentation?product=SONAR&language=3&help=ControlBar.09.html

I thought the point was to do everything from the keyboard...? After D, D, using keyboard cursor keys work to scroll the PRV (or zoom with Ctrl).

Or you can just hit D, D to quickly toggle the multidock closed and back open. No?

I don't recall right offhand if X3 allowed stretching MIDI clips by dragging - I suspect not. But Cakewalk by Bandlab does, is free, and has many enhancements fixes over X3. You should get it: https://www.bandlab.com/products/cakewalk Otherwise, the old-skool methods are: Process > Fit to Time (if you know where you want it to end) or Process > Length (if you know how much longer/shorter it needs to be as a percentage of its current length)

D'oh! Fixed.

A beat is always a quarter note in Cakewalk. So in an 8th-note meter,the point half way between two 'beat' markers is .25. I am pretty confident this goes back to the inception of the function. Here's a thread from 2013 (three years before Platinum) in which I mentioned it: http://forum.cakewalk.com/tm.aspx?m=2921804 "(note that fractional beats are entered as decimal values not ticks, so 02:480 is 02.500)."

Clean Audio searches for projects that use the audio in the specified folder. The Global Audio Data folder doesn't have a project folder as a parent, so the only option in that case is to search everywhere. If you just want to clean up one project, and get its audio into a per-project folder, the thing to do is to open the projectin Cakewalk, Save As to a new project folder, and enable the Copy All Audio checkbox. Only audio that is actually referenced in the project will get copied. To clean the Global Audio folder of all audio no longer referenced by any project, use the Cakewalk Audio Finder tool. I do all my 'cleaning' with this tool as it's more powerful and flexible in terms of telling you what's what, and giving you options. See post #5 in the below thread for a 'primer' I wrote on how I use it: http://forum.cakewalk.com/Clean-Audio-Folder-still-broken-I-see-m3023320.aspx#3023681

Set Measure/Beat at Now uses decimal beat fractions. An 8th is .500. EDIT: Has forever worked this way, incidentally.

Glad to help, as always. My professional customers should be so appreciative. ?

As Chuck suggested, you can Ctrl+Shift+A to deselect everything. Alternatively, I've found that if you hover in the very narrow band of the timeline below the MBT scale, you won't get the selection marker tool cursor popping up when nearby.

Sounds like maybe you have ASIO4All installed...? I would uninstall it since you have a good interface with native ASIO driver. If that's not it, I would have to do some Googling. EDIT: Found some posts on the old forum indicating ""Generic Low Latency ASIO Driver (1 in 1 out)" is a Cubase thing: http://forum.cakewalk.com/Does-Anybody-Know-A-Better-Way-m3040170.aspx

Right-click > Split > Split at Selection option > Enter/OK Or hold Alt with the Smart Tool to get the Split tool and drag-select the region to be split out.

Glad to help. PM me a link to anywhere I can download. MS OneDrive or other cloud file store that you might already be using would probably be best, but Soundcloud is fine if it allows direct download. Or even your friendly, neighborhood Bandlab sharing facility. ;^)

Cool. So you have your Timing Offset back to zero, and everything is good? If 10.9ms is a round trip time with ASIO buffer at 128 samples, you might have to shell out to find a USB interface that does much better. I use a MOTU PCI-based interface that runs smoothly with a 32-sample buffer, and delivers an actual measured RTL of 2.3ms. RME is the only company I know of that gets close to that with USB. 8-10ms with a 128-sample buffer is more typical (3ms each way for the I/O buffers plus 2-4 ms for A/D/A conversion and 'hidden' firmware/hardware buffers.

A few thoughts: First, I agree that there is really no better way of approaching the arranging process, except that it might be simpler to use lanes of one track for the different sections of the original guitar part, rather than different tracks. Second, you might want to consider getting the timinng locked in first so that you can then copy tempo changes when duplicating sections, rather than having to map the tempo multiple times. While Melodyne and Cakewalk's own Audiosnap tempo mapping features can be helpful. I still find that manual tempo mapping using Set Measure/Beat At Now is more flexible, more precise, and ultimately doesn't take much more time (and possibly less) vs. all the massaging of tracks that may necessary to get the 'automatic' process to deliver the desired result. And, yes, once you have a good tempo map, you can 'smooth' or completely 'flatten' the tempo varations by enabling Audiosnap Clip Follows Project in Autostretch mode on the original track(s), and then adjusting or deleting the tempo changes that were needed to make the timeline follow the audio. If you're only going to add MIDI parts, it's easier to leave the tempo variation intact and quantize the MIDI to the variable tempo, but if you're going to record audio, it can be tough to follow someone else's tempo changes, and harder to tighten up the timing after the fact without introducing undesirable audio artifacts. That caveat applies to flattening the tempo of the original recording as described, but if it ends up being only one part in a multitrack mix, the artifacts will be masked to some extent, and it will making recording more parts with good timing much easier. I've posted many times about using Set Measure/Beat At Now to align the timeline to audio (or MIDI) on the old forum.cakewalk.com, and you can Google my old username, brundlefly, with 'SM/BAN' against site:forum.cakewalk.com to find them. It's pretty straightforward but there are some extra tricks for dealing with free-form or rubato intros, pick-up beats, rhythms that deliberately hit off the main quarter-note beat, etc.. This is the real world stuff that the automatic processes don't handle well. If you're interested in trying this, the best way for me to explain the process might be to share an example of what you're starting with, and I can map it (or part of it) for you, and tell you the steps I took.

Does it happen if you play the whole thing through from the start without interruption or looping? My first guess would be some plugin has unsynced randomness in realtime that does not occur (or occurs less frequently) with offline rendering. You might try bypassing some FX to determine the culprit, and then Google for similar reports on the web at large or the vendor's forum for possible solutions. You migth also experiment with different ASIO buffer sizes, and see if maybe that's a factor. If you have a non-zero bouncebuffsizemsec= parameter value specified in AUD.INI that's different from your real-time buffer size, that might explain why the issue doesn't manifest when exporting.