Amberwolf

I'd recommend removing all vestiges of ASIO4all from the system entirely.  It's known to cause various issues in CW/CbB/Sonar/etc, because of the way it works.   It doesn't work in "low latency", it's minimum latency is whatever the WDM/KS or other driver your device has, *plus* it's own latency, plus...whatever you have to use to get it to "work" in your program. 

It isn't really an ASIO driver, it's a wrapper for whatever other non-asio audio drivers are on the system, and it does them "all", so if there's a problematic driver somewhere (even if you never use it) it could cause problems for anything using A4A. 

It could easily be causing your issue, especially if that Blue ASIO driver worked before installing ASIO4all or on a previous system that didn't have A4A on it. 

Sounds like the long-defunct Mason Jar here in the Phoenix area.

19 minutes ago, David Baay said:

Used to be Edit > Interpolate, and I'm pretty sure it's been around since Cakewalk for DOS.

EDIT: Just checked, and it had moved to the Process menu but was still 'Interpolate' in S7 which is the oldest installed version I have. 

Ah, yes, I use that too, but more often just the slection filter which is basically the first half of interpolate, because that way I have the chance to actually see what has been selected and verify that is correct, before I do anything to it.  (because it's a modal dialog I can't get "behind" it to zoom, scroll, etc., as I can rarely fit everything I want to deal with on one screen at one time with enough zoom to see whats going on). 

32 minutes ago, David Baay said:

I've encountered that situaiton with velocity-switched samples. I generally use Process > Find/Change in that case.

Unfortunately that feature wasn't created yet in the ancient version of SONAR I use.

So I use Edit - Select - by Filter and then whatever process I need to. 

But yes, it is primarily with velocity-switched samples, and some with non-sample synths that have different stuff going on with velocity for different oscillators, LFOs, modulators, or effects,

1 hour ago, PhonoBrainer said:

6) if your primary utility for it is an audio storage system, better to create a YouTube account. . . SC could die tomorrow, but YouTube seems to be as futureproof as

It's also a little closer and easier to play, with slightly higher bandwidth.

FWIW, I used to have a SC page https://soundcloud.com/user-792388526 , but after so many spam-likes I removed all the tracks and just use it as a link-page to my Bandcamp https://amberwolf.bandcamp.com/ .

(and soundclick https://www.soundclick.com/amberwolf , but that's a virtually dead site; anything I post there can get like 1 listen and suddenly be in the top 1000 of the entire site's charts :lol: .   I still use it to host experimental stuff that isn't ready to go on the bandcamp page or alternate / older versions of BC "final" songs). 

1 hour ago, Rain said:

My cat doesn't seem to care much for my music. 

 Neither does my present dog (who isn't really a dog, just a schmoo-alien in a dog costume, who occasionally goes on vacation back to her home planet and leaves the costume behind, which gets lonely and thus acts like an actual dog).  

One of the things I like about this song is the parts you've got in your song, but also the vocals, which are a type I like (along with some of Yazoo / Alison Moyet, Eurythmics / Annie Lennox, etc).   I don't have a useful voice for this kind of thing, but it would be how I would sing if I could.  

I added some other stuff I found to my first reply, but this one may be the "easiest" solution so I'll post it separately as well, to make it easier to find:

This page http://recordinghacks.com/microphones/Blue-Microphones/Yeti-Pro says there is already a 1/8-inch stereo headphone jack, which the "fact sheet" says is "Zero-latency headphone output with volume control for direct monitoring".   This has to be post-preamp (and so have the sound quality you're after), so you could use this signal directly into your regular audio recording interface, and thus have only the IF's latency to worry about.  

3 hours ago, Misha said:

The problem is latency. Clips come at about 1 seconds late.  If I do low buffer of even 256, starts crackling. 

That is a pretty long latency.  I wonder what they have the driver doing that it takes that long to process--it seems pretty unusable for a mic presumably intended to be a live audio source.   

EDIT2: This page

https://sharklatan.com/fix-microphone-blue-yeti-usb-advanced-audio-device/

has info on fixing certain issues; I don't know if any of them would fix a latency problem. 

EDIT3: YOu didn't say which driver or driver mode you're using, but if you're using ASIO4all to get this mic working with ASIO, it would probably work better to uninstall that and use the native WDM/KS or WASAPI driver it has instead (since that's what the ASIO4all is actually using anyway, and just adding ASIO4all's own latency on top of that. 

EDIT4:  Also, it may make a difference whetehr you have it plugged directly into the computer's USB port (and which port it's plugged into), vs thru a USB hub or some other device's USB plug that then feeds into the computer (like some computer keyboards have, for mice and other things to be used without using up your computer's ports). 

3 hours ago, Misha said:

Yes, driver is ancient, but I was wondering if this clip placement can be set somewhere globally to compensate for certain time, so future recording with this mic will respect that, unless I change it back.

Do you mean something that automatically shifts the clip position after recording is stopped to put them where they should have been?   

Other thoughts:

Depending on the version of software (CbB, Sonar) you have, there is a "time+" to adjust the playback time of tracks.  It doesn't affect the recording of new material, so you would basically just be syncing the existing stuff in the project to whatever is coming in on the new one.   With a delay of a whole second, that would be pretty hard to work with. 

There isn't a way to add a negative value to the incoming stuff to match the unaltered playback time of the existing project (it would have to play before the signal was actually coming into the program  ). 

3 hours ago, Misha said:

I decided to re-visit the mic, as it has a very particular sound.  What I found is that using XLR through interface sounded differently. Probably because of pre-amps. However, when I plugged it in as a USB device, I got the sound I was after.

FWIW, if there is no other solution, you could open the mic up and trace out the circuits, then find the output from the preamp (if it's a separate piece from the USB conversion; it might not be), then install a jack for that to run to your regular interface.  Not necessarily an easy task, and there is risk of damage that could leave it inoperative or even unrepairable. 

EDIT5: this page http://recordinghacks.com/microphones/Blue-Microphones/Yeti-Pro says there is already a 1/8-inch stereo headphone jack, which the "fact sheet" says is "Zero-latency headphone output with volume control for direct monitoring".   This has to be post-preamp (and so have the sound quality you're after), so you could use this signal directly into your regular audio recording interface, and thus have only the IF's latency to worry about.  

EDIT1: This video (for a different purpose) shows the disassembly of one of the Yetis (dunno if it's the same model) to give you an idea of what is involved in disassembly. 

... https://youtu.be/1yxUSUbJkwk

What specific synth is "GW synth"?   

Is it a separate piece of software you're running outside CbB?   

Or is it a synth within CbB?

What MIDI driver do you have installed on the computer?

And what software outside of CbB are you trying to route it to?

 What synth are you using to play the midi with, and how is it routed to your speakers?

Meaning:

Are you routing the midi to a hardware synth or keyboard outside your computer?

Or are you routing the midi to an internal softsynth inside CbB?

Details on how your midi is routed to the synth, and how your synth is then routed to your speakers, will help us help you figure out where the problem is and how to fix it.

A quck google found this on reddit (yeah, a bastion of truth and reality :lol: ) for an instance where someone tagged a vehicle:

https://www.reddit.com/r/AirTags/comments/17vjfac/just_found_an_airtag_hidden_in_my_truck_how_can_i/

Quote

 

Corey415

•1 yr. ago

“Every AirTag has a unique serial number, and paired AirTags are associated with an Apple ID. Apple can provide the paired account details in response to a subpoena or valid request from law enforcement. We have successfully partnered with them on cases where information we provided has been used to trace an AirTag back to the perpetrator, who was then apprehended and charged.”

https://www.apple.com/newsroom/2022/02/an-update-on-airtag-and-unwanted-tracking/

 

FWIW, it's much much safer to let it save as a copy, not over the original.

If something bad happens during the save, and it's saving over the original, you lose *ALL* of your work (unless you are constantly backing it up to another file somewhere else).  

Unless you have a really tiny harddisk, it is not going to be using much space for the project file itself (since the save doesn't resave the audio files that take up all the real space--those are already saved in your audio folder(s) and are just streamed off the drive as needed). 

I'd also like an ability to constrain drawing values of (assorted things); for me something like holding a modifier key while dragging would do the job 99% of the time, as long as doing that doesn't alter my selection (so I can perform the action on just selected notes when necessary). 

FWIW, there are certain sounds in certain synths that respond using different samples or different sound variations when the velocity is at specific value transitions; in these cases I have to set the specific velocity that triggers that if I didn't manage it while attempting to play in some notes where I miraculously got the timing I wanted (that I'd never be able to repeat). 

 For these cases, I'd generally do something like what the OP is doing, except that I'd probably select by a filter like a velocity range and cut and paste those notes to separate track or clips in another lane, etc., and then alter just the velocity of those notes by just setting them to the specific velocity, or by adding or subtracting enough velocity to trigger the sound I want. 

Mostly this is done for percussion sounds, but sometimes it's a synth track (most often Z3TA2+) used for some mostly-repetitive pattern (that usually used to be an arp pattern that was then edited in the track as individual notes). 

A bit OT....a repair shop I once worked at did contract service on some heavy duty large and fast diablo dot-matrix (or maybe teletype?) printers for various places.   To make sure us new employees paid proper respect and caution to the equipment, the lead tech / supervisor showed and told us what happens when an idiot wearing a tie leaned over into an operating printer....   

At some point in the past, someone had done that, and had their neck broken.   The demonstration given was convincing enough I think it could actually have happened. 

If it's the actual fader motor, there are replacement parts out there for some types.  This is a link for behringer part, not peavey but maybe they also carry those:

https://www.sweetwater.com/store/detail/MOTORFADER--behringer-mf60t-motorized-faders-set-of-5-for-motor-controllers

Remember also that the motors are driven by electronics on the board itself, and those can also fail, though it's almost always the mechanical parts that fail first. 

Most of the motors in things like this are simple brushed motors, so the brushes can wear out, leaving the motor working poorly or not at all.  Brushes are actually replaceable in a lot of motors if you are willing to take one apart to do that, but they aren't usually available as parts, so you have to make one yourself (the easiest way is to use a brush that *is* available that's close in size and shape but bigger, and whittle it down; some RC hobby forums have threads on how to do this). 

There are a number of different virtual audio cables out there, some very easy to use, and some very complex. 

https://www.google.com/search?q=virtual+audio+cable

If your other DAW is a separate computer that you can't just run audio cables to, some support offboard (networked) computer audio routing, most don't. 

As long as everything is on the same computer, then if you are using the regular Windows type drivers (WDM-KS or WASAPI) there are several possible setups you can use, so that SONAR can have your speakers' hardware out as it's main out, and a separate send or bus with the virtual audio cable as the output, so you don't have to keep switching back and forth in your configurations.  

But--if you are using ASIO (where you can only have one audio driver enabled at a time), then AFAIK there is only one that can be used in a way that lets you hear SONAR while piping the audio around (it would also let you hear the other program, and you can also route audio over networks), and it's the most complex:

https://give.academy/ODeusASIOLinkProPatcher/

You'd want to read it's documentation completely before setting it up; it's not super comfusing but it has a "style" to the way it's UI is laid out that takes some getting used to. 

9 hours ago, Sock Monkey said:

If you read the other answers you would see you only repeated what was already stated. 
And it was  clear exactly what the OP was doing and now they understand why. Solved. 

Neither one of those is the case, as I am not telepathic and cannot tell what they did from the insufficient information provided.   

That was why I asked the questions, and then posted the explanation of the reason for the questions.  

If you were communicating with them outside the forum to get the missing information, or just managed to correctly guess how they set it up, or something else entirely, well, I wasn't able to do that.   

In the future I'll just remember to not answer this poster's questions, as I already do for a number of others.  

Apologies for the interruption. 

If you post the details of what you did, it will help others that come along with the same issue later.

On 3/27/2022 at 8:37 AM, User 905133 said:

Not sure if archive.org (the waybackmachine people) can link it into their site captures.  They are a local phone call for you, aren't they?  If you find out they can and are willing to do something like that, let me know.  I have a ton of manuals and related docs for a particular manufacturer that have dead links on every site grab I have seen.  Since I am an old man, and I doubt my kids would want to maintain my hosted repository, I have wondered if they could use the docs to fix all the broken links.

You can upload things directly to Archive.org to preserve them, if that's something you have time for.  

I don't know if they ever changed the policy, but a long long time back before they had that option, I offered some files I had from defunct forums and sites to fill in missing stuff, and they said they could not alter any archived sites, but they didn't explain further, and I didn't get replies to further inquiries.  

The bitcrushers I have used appear to be removing assorted frequencies, along with in some cases / settings changing the bit depth.

In those cases, you can't directly "undo" the changes the BC made--the data is "gone".  There are almost certainly maths that can interpolate the data that *is* there to create new data that "fits" the existing data, as can be done with images...but it isn't the original data, so it may not sound even remotely the same as the original did, depending on what was missing.

I expect that some form of "LLM" AI where it's model is composed of bajillions of sounds it can use to compare to the BC'd sound, and where it has maths to try to figure out how the BC was done (which gives it more info on what's missing and why), would be able to do a decent job, if slow. 

There are probably simpler ways but those are two I can think of that have already been done wiht other "missing data" situations to help restore damaged (accidentally or deliberately) data (especially images).  

.

I don't know that this will do what you want, but it was suggested in a search for anti-bitcrushing plugins as something that might help

Apulsoft's ApShaper https://www.apulsoft.ch/apshaper/

Never heard of it before this, so dunno how it might do that, but it has a demo version....

,

53 minutes ago, tdehan said:

As I mentioned.... this issue is solved.  Thanks

Yes, and I acknowledged that before, but you posted a question, so I answered it. 

7 minutes ago, tdehan said:

I thought I did!  I said it as ttS=1... that's MIDI right?

Well, it's much more complicated than that, whcih is why the info is important to know how to help.

TTS 1 is a synth.  

That means it takes input from MIDI tracks, and it has an audio output. (several, actually, per another user's post above)

So you can set up a "multitrack" drum part in several ways.

You can feed a single instance of TTS1, that has just one audio track output, from a single MIDI track that has lanes for each drum's notes.

You can feed a single instance of TTS1, that has just one audio track output, from any number of multiple MIDI tracks;

--each of those tracks could be sending on the *same* midi channel, so that all of them are sending ot the same instrument (drums, etc) in TTS1.

--or each of them could be sending to *different* midi channels, so that they are sending ot different instruments in TTS1

--or some combination of those two

You can feed a single instance of TTS1, that has multiple audio track outputs with separate instruments on different audio outs, from any number of multiple MIDI tracks (see the variations above again for that, multiplied by the mulitple audio outs)

You can feed separate instances of TTS1, each of which has just one audio track output, from a single MIDI track.

You can feed separate instances of TTS1, each of which has just one audio track output, from any number of multiple MIDI tracks.

(again, complicated by all those --ways to set those up, listed above).

(there are further variations, but you get the idea)

This "complicatedness" makes it possible to do a lot of different things if you don't mind the complexity, but doing it the "simple" way limits the things you can do while making it much easier to setup (and remember how it's "wired").

So some of the advice in the thread assumes you're doing it one of those ways, some of it assumes another, but all of it has to make assumptions because we don't know which way you have it setup.