Misha

Hi Folks.

Is there a way to delete Take Lanes selectively (Shift or Ctrl) in bulk?

If I select the upper left corners of take lanes (see attached), right click-> delete, it only deletes a single take lane only, not all that are "selected"  Of course I can hover over actual audio clips and it will delete those in bulk.... but not the Take lanes they sit on. Is there a way to  select-> delete actual Take lanes in bulk? 

Thank you.

msmcleod,

Sorry for not replying sooner, I think my email notification is disabled on the forum.

Thank you for suggestions!

Yeah, I think " I would create per-output instrument tracks" this is the way to go. I have one project I am working on now, do not want to screw it up, will try it on next one.

The last workout also sounds reasonable.

Thanks again!

msmcleod, thank you for the reply and suggestion. Yes, you correct a single channel in this scenario. 

So, to put it to rest, there is no way to actually use a single track (the one Kontakt sits on) with multiple volume automation lanes in same track (Add Automation Lane) ,  something like Midi envelope channels,  to control volumes of individual instruments  within same single instance of Kontakt?

Hi Folks!

 Can somebody please suggest a workflow other than what I do?
Here is the scenario. Cakewalk + Kontakt+single midi chord track that outputs to Kontakt. Assuming I have 4 instruments loaded in Kontakt. What I want to do is to  automate volume of individual Kontakt instruments linearly - automation lane with nods,  not through Midi learn fader in Kontakt. I usually just open 3-4 instances of Kontakt and use volume automation in Cakewall track which  Kontakt is attached to. The idea is to use a single Kontakt instance, instead of 4 and not using  Kontakt "midi learn" volume sliders. Is that possible? If yes how?  If it is doable, is it possible to do something like a  template for future uses, the "generic" template? Meaning, regardless of what instruments are loaded to have certain number of automation lanes that correspond to Kontakt instruments from up->down. I hope I am making sense I am only interested in volume automation.

Thank you in advance!

Hi Folks, Noel.

Was there a solution found for this? 

I never had this  sharing violation issue and I have used Halion/Sonic almost in every project. Now it does not let me save, this sharing violation is popping up. I am able to do a "save as", but the next time I open Cakewalk, I would have to re save "As" again.  So far, 20+ files for single project

Kindly let me know if there is a solution.  

P.S. E-licenser latest. Done maintenance routine.

Thank you.

I feel like completely disconnected. 

When recording, sure, I press little "fx" button and can get away with 256  buffers... 

But when mixing, having a bunch of dynamic, lookahead vsts (8 or more) plus  some Kontakt/Halion stuff, I can not getaway with anything lower that 2048.  

 i7-8850H/32ram/nvme m.2/ usb 3 + thunderbolt.. Tried both usb controllers. Oh well. Thanks everyone!

I am not a heavy user  at all

But I might have instance of Halion + Kontakt + few FX... Still 2048 is usually what I have for mixing / auditioning and it works, but sometimes need to go higher.  Limit 1024 in Motu is  just ridiculous. 

Built is great btw,  preamps are nice if you are dealing with small projects.  

John, 

Thanks for the info! I got my M2 last week. As much as I would like... to like it, it is going back next week to the store.

On 41k projects MAX buffer size is 1024. You can not go higher (on 41k). This is the most ridiculous maximum buffer size I have seen in the past 20 years.  The lowest I can get away with is 2048 due to FX/VSTs  when mixing/auditioning.  I should have been more careful in reading specs. This did not even remotely ring a bell. As soon as I ran my project, first 15 seconds, I had clicks/pops and stopped audio engine. A nice young man at Motu wanted to "tweak" my computer, I kindly refused telling him that even my 10+ year old audio Yeti Pro and 2 other interfaces, for the sake of argument (with enough buffers) are pulling my project just fine on the same computer. 

Ohh well. If it is a hardware limit, which I will accept as a lame excuse that is one thing,  but if they artificially limited buffer size within software for whatever twisted reasons they might had, this will be my last Motu experience.  You had been warned.

Ok, talked to a guy at Motu. He said it should work.  Routing a standalone program audio to Cakewalk.  We shall see. 

Hmm...

It is a bit confusing.  "mixing output with hardware"  I thought that you can have it routed as internal signal mix.

There is only one way to find out. Thankfully there is a return policy  I will try it and report back.

 Всех благ,
Миша.

azslow3,

Thank you for joining. Unfortunately I do not understand German (Dutch?)

But, I do not think you are correct about only one pair of loopback ports in M2.  Loopback seems to be a separate thing from physical outputs.  And as I understood from this video, it has 2 pairs of loopback outputs (even on M2) .  

https://cdn-data.motu.com/django_prod/techlink-attachments/156163/2020-08-24 11-17-09.mp4 

I know there are some software solutions that exist for multi client ASIO,  but overall this seems to be a perfect package for the price, if it works like I want it  of course  

Hey Hey, don't derail my topic

I know all the good stuff that is said about RME. I already have a good interface that I use to record vocals etc. I am keeping that.  The MOTU M2 I want mainly because it has that loopback feature and it just seems it just might do what I want from it.  Hopefully without Windows  drama.

InstrEd, Thank you.

My probable choice was  M2

Hopefully it will do this sort of routing thing.  Seems to be quite a piece of gear at that price point.

These guys have M2s in stock (for now) :

https://www.perfectcircuit.com/ 

Tonemangler, 

thank you for your input.  I done some reading and from what you are describing, it seems doable. There are couple of issues I see that I forgot to mention and your post "reminded" me of one. I intend to use this particular loopback interface with headphones only...  You are running your setup, directly monitoring external source (youtube) through dedicated 1&2 outputs, but  headphones I think would give me whole mix. Which can be problematic because as I understand the recording channel in Cakewalk has to be muted so it does not create feedback.  

My goal is to route the audio from non VST, standalone software to a track(s) in Cakewalk for auditioning and recording in Cakewalk. Sounds simple with the loopback, but it might get tricky.

Tecknot,

I assume interface with loopback function will have a proprietary driver, which most likely be Asio. If I am thinking about this in right perspective, it is internal audio routing. I do not think it is a combination of Was/Asio...

Anybody used these interfaces with audio "loopback"  with Cakewalk + other sound producing, non vst software?

Thank you.

Hi Folks.

Can somebody who actually used/knows about this reply. I am looking at a couple interfaces with "loopback" feature. To my understanding they are something like multi client Asio drivers?

Here is what I want to accomplish. I have a standalone  software that has internal synth (produces sound+midi) The midi part is relatively straight forward.  Loopmidi in/out of Cakewalk, done. The audio part what concerns me. Would I be able to monitor and record audio from another software to Cakewalk with interface that has a loopback feature?  Thank you in advance!

John,

I do not think I did... answered my own question.  I believe Noel once suggested that I try a different machine to replicate the problem. 

The issue with that is I can not transplant whole project because another computer does not have all the licences and it would take a long time to download huge libraries.. But I did try something. I opened a 12 audio track project on another, far less powerful computer. Loaded it with 2 dozen of plugins... mostly with ones that have "look ahead feature" What I got is some random stuttering / clicks and very rare Audio Engine stops, it even pulled everything on 256 buffer, but nothing remotely close what I get on a much newer and faster machine running 1024 or 2048 buffers! There was no distortion of sound on slower computer after "stopped engine", it played as new after I hit that play button. It seems there is most likely nothing wrong with Apogee One.  It is my main computer + Apogee + Cakewalk  equation that does not work right.

I am just trying to narrow things down. This is what I get.
1)There is some piece of audio software code that is interfering with Apogee + Cakewalk on my faster PC
2)Unrelated non-audio hardware / software  that is crippling Apogee

3)USB port or internal hub has issues particularly affecting Apogee  

4)Cockroach died inside machine, bridging some circuits

5)Aliens

I cleaned a lot of bloatware from computer, turned off throthling and most of stuff generally recomended for tunning pc for music.  There is one thing that I should probably try. If it is Usb port/ internal hub related, to get a different cable to plug it to thunderbolt port as it  has a different controler to see if it resolves the issue.  But on the other hand, same port does just fine with another interface.  Really like that Apogee, hope to make it work properly somehow.

Hi folks.

Hope that somebody can shine a light on the issue for me...  Ok so we are clear, I know what is "Stopped audio engine" in Cakewalk.... Problem has a related twist that I can not solve.

I have Apogee One (Asio) that I use as for mobile mixing. On heavy vst project it often overloads, even on max 2040 buffer size(gets me that "stopped audio engine") But the twist is that sound after that becomes distorted and does not return to normal. Unless I change the buffer on interface software to anything else... Meaning it would become normal again (without distortion) even if I change it to lower setting. Apogee is pretty much useless, suggesting some very generic solutions. My issue is that in theory after the "stopped engine" it should HEAL itself without touching buffer settings... As it happens with my main interface. If I get "stopped engine" it would resolve on the next press of Play button.

My assumption is that Apogee driver needs to reset itself after boo boo without manual therapy... Or maybe Cakewalk does not send correct info to this particular interface for some reason?  I am not tech advanced, I do not know what happens on the background of driver and interaction with Cakewalk at time of "stopped engine" and when you resume, by pressing play button.

Two other interfaces from another brands restart normally after overload, but not the Apogee One. (Drivers /firmware updated, latest Cakewalk)

I am not asking for a workaround. A simple workaround would be to freeze / bounce tracks, use fewer vsts etc... That is understandable. What I am trying to figure out is not "healing back" is a bug in Apogee driver or it has to do particularly with Cakewalk given symptoms above.

Thank you for your time. 

Kevin,

Well, this is how I see it:  Treat "track folder" as a track.  If Track has 5 take lanes, 3 of which are muted... Muting/un-muting whole track will not alter "muted" status of those muted take lanes, they will remain muted... unless you decide to un-mute them manually of course. Basically, concept of request is to be able to mute Track folder output only, not touching particular mute settings in everything that is going on inside that track folder.

So symmetry with solo (in "track folder") would be identical if treating an individual track that is soloed.  In track, it would play only active content, which is  not muted. 

Hi folks & bakers.

Not sure if there is a proper workaround. If there is, request is for automating it. If not, here goes:

1)Freeze whole "Track Folder" with all the content. 

2) Able to Mute / Unmute whole folder -> NOT the takes or separate tracks in that folder. Meaning if there are muted track(s) placed inside the track folder, it will "un-mute" if the folder is unmuted, creating cacophony.... "UNDO" does not work for that action.  Request:   if originally,  muted tracks are placed in folder or muted manually by expanding folder tree, they should remain "muted" if folder is un-muted.  Looking for a similar behavior to "take lanes".  If a "take " is muted by a right click->mute , it will remain muted if the track is muted or un-muted. 

Thank you for considering it!

Bob,

Exactly..., that is what I thought: "That 2nd icon lets you know that the audio track is assigned to pick up its input from a soft synth.  "

But what happens, after I record that signal from "Soft Synth" and delete that initial source "Soft Synth" , it  still resides in the project as full instance on that Audio track I recorded.

How do I know this:

1) As soon as I click on that "changed icon" with little midi plug on newly recorded track, full instance of Kontakt with all previously loaded instruments in it pops up in a matter of 2 seconds. It can not do that if it was loaded from scratch or even as a "saved" program

2) If the project is saved and closed, when opening it, it loads Kontakt, even when that initial "instance" was deleted.

3) When I have a few of these "recorded audio tracks" from Kontakt output, Cakewalk becomes glitchy with Audio Engine stop 1 toast.

4) If I freeze those "audio" tracks Everything is smooth.

Bob, actually I want to understand too and was hoping somebody would shine a light on this

See attached. 

Bob,

Speaking of workflow, the easiest in my view to capture audio from Kontakt (or other synth) is to solo Kontakt channel,  create audio track point input to Kontakt output, press record

It works, I just do not understand why the Audio track becomes a heavy "Synth" track.

I played with it yesterday a little and...

If I freeze this recorded Synth/Audio track and duplicate it with events, it duplicates as a normal Audio track. Go figure    

As far as I know, there is only one VST SF2 which does several things that most free ones (and paid) do not and gets updated is:

VstSynthfont. It is 64 bit, Responds to GM Program/bank changes, allows for large, multi file SF2 files. Full 16 channel mixer. Has many other features for SF2 playback.  It is priced very fairly too.

-------------------------------------------------

The most stable free VST   is:

BassMidi VSTi 1.4 by Falconsoft. A bit limited, but works well. Development stopped on that and as far as I know it will not be getting any future updates.

I think I figured it out after trying about 3 dozen things    Seems like it is my workflow flaw (rhymes too)

This is what I done to earn constant "Stopped Audio Engine 1" toasts:

1) Insert Instrument->Kontakt. When I was satisfied with how it played midi file  ->

2)I Inserted Audio track and set it's input to Stereo output from Kontakt, to record signal as Audio to this track. (see attached) 

3)When I recorded my "Audio" tracks from Kontakt, I deleted the initial  source "Kontakt instrument",  to cut unnecessary weight of the project.

But it seems that Each recorded Audio track (with input from Kontakt) somehow  became as instance of Kontakt.  As soon as I froze these Kontakt recorded "audio" tracks, glitches with "stopped audio engine 1"  went away and everything plays fine.

-------------------------

Little more patience with me, please.  I used a vintage thinking on this.  Lets say you have a CD that you want to copy to tape. Once you set input of tape deck to CD out,  after you done recording, the tape remains a tape, it does not become a CD just because you used a signal from CD deck to record it.

So now that I know the cause, what should be my workflow to record audio track from Kontakt, so it remains standard audio track and not what I got...?

I thank you in advance.

Misha.