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  1. Most likely the two different software implementations are different in how they define %. Rather than a standard acoustical definition, it represents a reproducible setting for the particular control. That said, your use of reverb on a bus with sends from different instruments at different values is one reasonable approach for saving CPU by using a single reverb. You may want to verify the sends are post fader (it's the default) if you want to preserve the mix value as you adjust an instrument fader.
  2. If you are using MIDI long notes and expecting to start playback in the middle of any of those notes, that is almost guaranteed not to align with the rest of your score. The reason is that the DAW only has two choices when starting playback in the middle of any note: 1. start the note at the beginning or 2. don't play the note. In case 1, some kinds of synth voices can't possibly align when started at just any place in the score because they are being started at the "wrong" time. If you were to freeze those notes into their audio equivalent, you'd then be able to start playing anywhere. If you're not talking about long MIDI notes, then never mind. It must be something else.
  3. In real life, true stereo is represented by relative signal delay between left and right and also reflections from the environment, maybe more than volume differences. One can also argue for differences in frequency response. The relative signal delay depends on the distance between microphones or ears. Reflections and frequency response also depend on microphone orientation and ear physics. Microphone orientation can certainly also affect relative volume as well as frequency response. You could try this on a signal recorded in mono: put the straight signal on left and the same signal delayed by a fraction of a millisecond on the right. With the same amplitude on both sides, the signal moves to the left. Experimenting with the delay amount can be fun. When considering how to reposition instruments (even when recorded in stereo), why not consider relative delay? One reason why not is listening (or mixing) stereo to mono can wreck the whole sound if only delay is used to represent stereo. A "true stereo" panner would have a lot of work to do.
  4. Any tempo changes near 9 of the first one? Tempo-based effects?
  5. One way: any controller can be modified using the event inspector in the control area. After selecting a range, that inspector can change values by n% or by increment +n or -n. By % would flatten or exaggerate the dynamics, while + or - would change the over all level, keeping the same dynamics.
  6. What a pain. The Nord should have just shifted the notes before sending them. As it is, the track you record from the Nord would probably play back correctly on the Nord, but not on other synths. Also, most likely, the octave shift CC will not be sent when switching to a program preset that has a programmed octave shift. Again, not friendly to other synths. It's possible a CAL script could run through a track and replace all the CC 29s with the appropriate octave shifting. Hopefully the CC event has an included value to indicate which direction to shift. And beware sustain (CC-64) messages and also notes held across a CC-29. But the shift in a patch change probably won't be seen and may confuse your entire strategy. Also, what happens when you send MIDI notes to the NORD outside of the configured range? Would a track restructured this way still play back correctly? CAL is a custom scripting language built in to Cakewalk. If you have a programming background you could do it. There are also a few people on this forum that are so skilled that they might volunteer to do it for you.
  7. Occasionally I notice late buffers show up when the song is just over, not while it is playing. No artifacts are heard... Also, my guess is the hidden "safety buffer" in some interfaces overcomes artifacts for slightly late buffers.
  8. Useful info here! But just to clear up my understanding about scheduling VSTi's? Normally there's just one VSTi serving any audio (synth) track. That is to say, the "synth rack" for that track only has one item in it, not a stack. This is clearly different from the case where an FX rack with multiple audio effects serves a single track (or bus). My understanding of plugin load balancing is that each audio effect in such a rack could get a thread simultaneously. That per track concept of load servicing does not even apply to synths. Your last paragraph implies a slightly different concept for synths; it doesn't sound like there is anything preventing running simultaneous threads on individual synths in the synth rack (it's global). In this sense, doesn't it make sense to say there is load balancing among the synths in the synth rack (and there probably always has been)? In this view, for a synth instance that has multiple parts, a single thread has to serve all the parts, even if the parts are directed to separate audio tracks (i.e. for the majority of VSTis, not supporting multithreading). Are there situations where Cakewalk would do multithreading to a VSTi that supports it? That would clear up most of my question (post 5?) about "overhead" of stacking synths. RAM is another part of the equation. For a VSTi that can support multiple ins & outs, how much difference in RAM does it take for multiple instances each with a single voice vs. a single instance with multiple voices. The code part should be shared. For a sampler, if samples can be shared between instances, I imagine there is relatively not a whole lot of difference. But how about for a synth like TTS-1? Does it wind up using a lot more memory for multiple instances?
  9. I'm interested to know how mildly and how substantially overhead might be increased by stacking a mono-timbral synth. Also, I wonder if a multi-timbral synth can run multi-threaded.
  10. I had the problem of my Korg M1 (64-bit VST2) opening properties all blank/white in an existing project. I tried Korg Wavestation (also 64-bit VST2), and the same thing happened. I use the Intel onboard 630, no MSI, no gaming software, only one monitor. The project is fine in Sonar latest Platinum. I worried! The project was last saved by Sonar X2. I resaved it from Platinum, and then those VSTi's worked OK in Cakewalk. Lucky me to have Platinum still around.
  11. I guess my notions of mute and solo come from mixing equipment, where they apply to any signal on a channel. The "soloing tracks" article you posted doesn't say whether it applies to MIDI or audio, and it doesn't refer to the input echo control, or make a distinction between recorded or live signal. The MIDI echo article doesn't refer to mute or solo. So really, the behavior is not spelled out, so it could be anything. It's potentially confusing in its description, and maybe as well in its implementation, since recorded and live signal are merged somewhere along the signal chain, but it's not spelled out for MIDI. I did notice, though, that the behavior for audio tracks is different from MIDI. In particular, solo on one audio track shuts off the incoming live signal on a separate, echo-enabled, track. (simple: project with two empty audio tracks, both echo enabled; listen to audio input through one of them, then solo the other; the audio stops) That is how I expected MIDI to work, too. Here's Cakewalk's audio signal flow. The little tan M and green S operators come well after recorded material merges with live input, so it's pretty clear how audio should work, and it does seem to work that way. I haven't seen the MIDI equivalent signal flow diagram. Sorry I was so late to respond. By the way, I don't use "Always Echo Current MIDI Track". But that possibility just brings up the notion that the echoed "Current MIDI track" with a mute, or some other track with solo, would cause the same behavior as I observed for a conventionally input-echoed track. Bill B.
  12. You could write for Stephen Wright. Actually, here is a Stephen Wright line: "Last night I played a blank tape at full blast. The mime next door went nuts." Your line is funnier.
  13. support@cakewalk.com, for reporting, but no tracking that I know of. It sounds like a bug. Can it be isolated to the playback of just one track or does there need to be a stream from multiple tracks? Generally, people find that those initializing MIDI messages need to be spaced out in time in front of all the notes, otherwise the first notes are all skewed depending on how long the synths take to engage program changes, not just sysex messages. The same might be said for controls & program changes on the same track, namely leaving time after a patch change before the next control. (I come from a world of dino-synths, so maybe today's synths process patch changes or sysex instantly.) Also, MIDI is pretty slow, on the order of just over 3 bytes per ms. That's 1.6 bytes per Sonar's tick at 120 bpm. So putting a sysex in there is guaranteed to delay some of the other messages "on the same tick". Still, order is certainly important.
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