msmcleod

8 hours ago, sjoens said:

Just wondering how you did this as the Gain knob & TV Vel+ slider can't be selected for Write-to automation.

Right click on the gain knob, and select "Remote Control" - you can assign a CC to control it there.

You're correct however - you can't automate it using automation.

38 minutes ago, Aewon said:

Interesting MCleod. Thank you for taking the time to reply. That is a shame. So i need to look for vst2 only plugins? Can you point me in the right direction or is only vst3 now?

You've not said exactly which UAD plugin it is - so you'll need to check their website.

If it is VST3 only though, there's nothing that will make it work in Sonar X1.

Two things to bear in mind:

1. Sonar X1 pre-dates the VST3 format.
2. A few years ago, Steinberg revoked the license for releasing new VST2 plugins - essentially forcing vendors to release VST3 only versions (unless a VST2 version of the plugin was already released before the license revocation).

So if this plugin is a VST3 only plugin, it won't appear in Sonar X1.

2 minutes ago, Mr No Name said:

I have just invested in some AIR stuff on KVR site, they have package deals for both DB33 and XPAND 2 + Vacuum classic synth for $9.99 until 14th jan

I'd guess it's an updated version, might be worth an investment of 10 dollars??

https://www.kvraudio.com/marketplace/air-legends-pack-by-air-music-technology

Both are great sounding VSTi's and well worth the money.

I rarely use presets for VST's (vs VSTi's), but if I do have a particular one I like, I save it inside an FX chain. 

If you hold SHIFT when dragging from the browser to the FX bin, it automatically removes it from the FX chain with all the settings intact.

IIRC I had to re-install the VC++ redistributables after I last installed the AIR stuff - maybe not for the synths, but definitely for the effects stuff.

You may also want to re-install the iLok software. 

Whoever wrote the original installers didn't think to check for newer versions of the components, and just blasted over it with the ancient stuff in their installer.

Most likely a corrupt ctrlsurface.dat.

Before you launch Cakewalk, rename ctrlsurface.dat to ctrlsurface.old  within %APPDATA%\Cakewalk\Cakewalk Core - then start Cakewalk.

You'll most likely have to set up your control surfaces again within preferences.

There's also "Fit Improvisation".

Basic usage:

1. Add a MIDI track, and arm it
2. Start recording and tap the same key for each beat through the song until the end
3. Select the MIDI clip you just recorded, then Process->Fit Improvisation.

A tempo map will be created from what you recorded.

One other thing you could try... I've found that having lots of small MIDI clips on a track eats up way more CPU than a single MIDI clip on the track.   

If you've got lots of smallish MIDI clips, try consolidating them into one by selecting them and "Bounce to Clip(s)".  

16 hours ago, kitekrazy said:

Doesn't Linux already lead in appliances and hardware not requiring an OS?  That's what the fanboys claim. The standalone Push is a Linux box.  

Linux is an OS.  The advantages for hardware is that it's a powerful and very reliable OS that makes development much easier.  The downside is the start-up time. 

Personally, I think Linux as an operating system rocks.

However, as a DAW platform (certainly on desktop), it has these two major issues:

1. Lack of audio driver support for professional interfaces
2. Lack of native Linux VST plugins

(2) can be worked around by using a wrapper, but this increases CPU usage.

I don't know Presonus's motives behind this move, but it may be to allow some of Studio One's features in a hardware product at some point (total speculation here!!).  Korg already uses the Raspberry Pi as the basis for a lot of their hardware synths, so likewise, Presonus may be looking to re-use Studio One's technology in a linux-based hardware product at some point - maybe as part of their StudioLive series?  Having the fully-fledged DAW out there in the wild being effectively tested by users may help to iron out issues before they commit to shipping it in a hardware form.
 

9 hours ago, Starship Krupa said:

And if KONTAKT is your main sample player, might it be more efficient to use its multitimbral capabilities to minimize the number of instances? I know that this helps for some instruments but not others. SampleTank is said to use the same amount of resources whether you have 3 instances of it or use one instance with 3 different ST instruments.

If you're likely to want to freeze/unfreeze regularly to edit parts, then I'd recommend using multiple instances of Kontakt. 

The CPU usage is around the same for a single instance/multiple instances (assuming you've got the multi-processor options set correctly within Kontakt).

The amount of time to freeze everything is around the same, however with multiple instances you have the option of unfreezing one of the instances in isolation - with a single instance, it's all or nothing.  Re-freezing that single instance in a multiple instance scenario is much quicker than re-freezing everything.

However, if you don't regularly edit parts and just want to freeze to allow you continue with the rest of the composition, then a single instance might be more convenient as it's a single click.

As an alternative to freezing, you could consider bouncing to a stereo track then archiving your original tracks.  I use this method when I need to re-record something later on in the mixing process and I need latency as low as possible.   

If you've got lots of tracks that need freezing, this method is often much quicker as it only requires one mix-down, rather than one per track (or one per synth).  Obviously when it comes to mix-down, you'll need to mute this track and freeze the individual tracks in order to get more granular control, but it can be useful while you're still editing/adding tracks to the project.
 

16 hours ago, limpet said:

 Apparently the more expensive 8/16 fader versions work better, but I`m not sure I want to risk it.

I've got the FaderPort 16, and @Jonathan Sasor has the FaderPort 8 - both work perfectly with CbB. 

The only exception is lack of access to the Synth Rack - unfortunately this feature was added after the FaderPorts were released, and because the FaderPort fakes multiple MCU button presses, it's not easy to work around (I did try, but it got messy).  IMO this isn't a big deal, as most synths have too many parameters to make editing parameters feasible with a Mackie Device in any case.

For the single fader version, you could try setting it to Universal Mode or Cubase Mode within the Mackie Control Surface dialog.

 

I've got the TC Helicon Voice Prism  and also the VoiceWorks plus.

Normally I don't use the harmony features, just the voice modelling, but it's just an extra connection.

1. First make sure Perform VK MIDI Out is checked in preferences.
2. Set the MIDI Output of your MIDI harmony control track to Perform VK MIDI Out.
3. Connect the L/R outputs of the TC Helicon to inputs 3/4 of the focusrite
4. Connect the 3/4 outputs of the Focusrite to the inputs of the TC Helicon
5. On your vocal track, insert a send to a new Aux Track - this is where your harmonies will live.  Ensure Input Echo is enabled on the Aux track.
6. On the Aux track, insert an External Insert effect within the effects bin. 
7. Set the Send of the External Insert to Focusrite Line Out 3/4 and the Return to Focusrite Line Input 3/4 - click the button under the Delay - once calculated, it should read something other than "*0ms, 0 samples".

Once this is done you've got a couple options:

1. Let the TC Helicon do the processing in real time until final mixdown.
In this scenario, just make sure input echo stays active on the Aux track.

2. Record the harmonies as audio
If you've pre-recorded your main vocal, arm ONLY the Aux track, rewind to the beginning of the project (or where your vocals start), then record. 
If you've not recorded your main vocal yet, arm BOTH the Aux track and your main vocal, then record.  

Once the vocals are recorded, I normally copy the recorded audio to a new track, then archive the Aux track.  If I need to re-record the harmonies, I unarchive, re-arm, and re-record.

3 hours ago, SteveC said:

I think that's accurate.   But it sounds like you just want to bounce a section, not freeze the entire track?

You can bounce a section/region of a soft synth, but that requires a split instrument track so that you can record the soft synth's audio track while playing back that section (in real time).  Or, you can route a single instrument to an Aux track, then record the Aux track while playing back that section. 

There is no need to have a split instrument track - just select the region on the track (or tracks), and "Bounce to Track(s)":

Checkout videos 37,38,39 & 40 of this Sonar X2 course, which covers Audio Snap.

It goes through various of ways of doing this:

https://youtu.be/vUzUwK8ExC0?si=s1cxzGBI8Ux8HnlL

https://youtu.be/OxtIFAjH1js?si=9fMXYnDUrTw-MaIg

https://youtu.be/YL2n5zowZd4?si=djpP1zDxgAN5xoNB

https://youtu.be/k6lVqgk6dlI?si=tQyywLLrAsy3xmSY

The complete course is here:

On 1/3/2024 at 9:46 PM, Hoop26 said:

Hello,  im a fan now for 20 yrs of cakewalk pro audio 9, do you by chance have a copy of the program or know where one can be purchased? Computer crashed and ive tried sonar, the new cakewalk bandlab,but man the CW pro audio 9 was the ****.,i has acid 2.0 and 4.0 plus 1000's of loops for drums,had all the waves plug ins for all effects,I recorded around 50-75 songs with CWPA9 and managed to save them on flashdrive, but lost my Cakewalk and acid programs, thanks for any help, email jeffhooper26@yahoo.com if cant reach in here,

CWPA9 isn't available any more, unless you can find one on eBay - note you'll need both the CD Key and the Serial Number, and probably a Windows XP 32 bit machine (or earlier OS) to install it on.

CbB can load CWPA9 projects - I converted around 50 CWPA9 projects over to CbB without issue.

For best results:

1. Create a new Basic project
2. Drag your CWPA9 project in from the Media Browser
3. Clean up any tracks as needed, then re-save.

Don't use ASIO4ALL - use the official Focusrite Scarlett driver.

I've got a 12 year old machine (Intel i7 3770), and I can run at a buffer size of 64 on most projects without any issues.  Only very occasionally will I need to increase the buffer size to 128 or 256.

2 hours ago, Dave Maffris said:

Update--everything's going pretty well, other than the fact that when starting it up, it calibrates the faders, but I get an error saying that Fader 6 will not calibrate, and that I need to contact Mackie Support. This is a legacy unit, so I'm not sure they would support me on this issue--so the fader (6) doesn't seem to be working properly, if at all, so for the time being I'm avoiding this track in my Daw, which is kind of lousy, but it's at least a workaround for now. Anyone know if there's an easy fix for this calibration issue? Other than that I find the unit is working well. Thanks.

The faders themselves are essentially a tiny fishing reel, with the fader knob attached to the line, and an optical wheel / sensor at the end of the reel. The motor pulls the line up and down and counts the number of turns of the wheel.  The calibration phase pulls the fader up and down, so it knows how many turns there are between the top and bottom positions.  If there's a mismatch in the count for top to bottom vs bottom to top, or the fader appears stiff, then calibration fails.

There's a possibility that the sensor has some dust on it, but in my experience the most common issue is a twist or tangle in the line.  The mechanism is so delicate, it's almost impossible to repair. 

The only real fix for the calibration issue is to replace the fader.  They are normally pretty hard to find, but eBay has 3 available right now:  https://www.ebay.com/itm/134804999753?hash=item1f63010e49:g:XmoAAOSwQT9lT~Hh

Replacing the fader is a pretty simple job - just unscrew / unplug the old one, and plug/screw in the new one.

Most faders don't come with the knob, so make sure you keep hold of the old one.

1 hour ago, Glenn Stanton said:

is this for live performances or recording? if recording, the advantage of 6 separate channels would be to create multiple instruments (like string section) to perform the "chord" or small runs integral to the piece, and keeping it all in sync without "overdub".  

i'd probably go with option 1 - simpler and cleaner.

i've done some recording (and some live stuff) with people using expensive guitar MIDI gear - and ultimately, when recording, the challenge is getting a clean MIDI performance (even folks with great touch) to avoid errant notes. most times, we're simply resorted to using the guitar audio and Melodyne to clean up and convert, then split as needed across tracks.

live performance - most times you're outputting audio for the audience so the synths (yes, a number of people use several chained and i guess the VSL could do that as well) to get the full set of sounds -- but it requires lots of prep to get all modules and program switching etc setup (and the DAW can help or using other programs to coordinate).

will be curious to see what other folks have experienced.

 

^^^ This

I've used both the Roland GI-10 (the GI-20's predecessor) and JamOrigin MIDI Guitar 2.

Both aren't great for latency - especially in the lower notes.  I was impressed that JamOrigin performed at least as good as the GI-10. The GI-20 may be better, but physics will always get in the way with the low notes.  You do get used to it, but be prepared to alter your playing style accordingly - i.e. anticipating the notes as required.

I've never used either live, but I'd be tempted to use an old multi-timbral sound module rather than a VSTi, which would add yet more latency.

For recording, I've found it much easier to just record the guitar and use Melodyne to convert to MIDI - tweaking the EQ as required to get the best Audio to MIDI conversion.

I guess if I practiced enough with the GI-10, it might prove a better solution, but I use it far too rarely to warrant the effort.

If you view the notes in the Event List, you can edit the start time of the notes in HMSF, so you should be able to match them exactly to the video frame.

Alternatively, if you've got both the thumbnails showing and the video view open, the big time on the video view will show you the exact time in either HMFS or MBT for the thumbnail you click on in the video track.

14 hours ago, OutrageProductions said:

Try it with and without the "Disable Handshake" setting ticked in the Mackie Control preferences page (as posted above).

This shouldn't make a difference with a genuine Mackie Control, as it correctly implements the handshake.  Older versions of the Mackie firmware actually require the handshake, as it was part of the spec at the time.

For the original MCU (the one with MIDI din connectors), v2.1.2 is the latest firmware.  This works with handshake on or off.  It is possible to upgrade to v3.0.0, but you also need to upgrade the two PWM fader IC's.

15 hours ago, Dave Maffris said:

Thanks, I haven't had time to look at any instructions much, but I'll check out that link, I definitely could use a few tips I will keep looking into it, because I'm in the middle of a massive new paid recording project and trying to use this is a mix of helpful and not helpful, because the faders seem to always snap to zero (I mean inaudible, bottom of the board) when I open a project, instead of reading where the faders are set in the project, so I have to adjust them and sometimes they "stick" and I can't adjust them until or unless I remove the automation lane from thet rack temporarily. A work in progress for me.

 

This sounds suspiciously like missing MIDI messages.  There are a few potential causes of this:

1. Some MIDI messages are "unchecked" within Preferences -> MIDI -> Playback and Recording.  Pitch Wheel is definitely required for the faders, and I think Channel Aftertouch is also required for normal MCU operation.  System Exclusive is required if the handshake is enabled.

2. Wrong MIDI ports defined.  The MCU requires two way communication.  If the MCU's receiving port is wrong, it'll force the fader to its last known value.  This could also be caused by a MIDI merge feature on some more advanced interfaces - make sure any merge/thru feature is turned off for those ports.

3. A MIDI buffer overflow in the interface. 

The MCU generates a lot of MIDI traffic, and some MIDI interfaces just can't keep up with the load.   This is especially true for those USB to MIDI "cables".

Essentially what is happening is, the MIDI interface receives data from Cakewalk at USB speeds then has to send it out at 31.25Khz (orders of magnitude slower).  This requires the interface have a buffer large enough to hold the incoming MIDI messages while it sends them out at a slower speed.

Unfortunately if this is the cause, there's not a lot you can do except replace your MIDI interface.  I'm using a MIDITech MIDIFACE 8x8 with my MCU/XT/C4 ( and also two controller keyboards).  This interface has been rock solid for me.

Your VST DLL's are in C:\Program Files\VstPlugins\Toontrack.

The path you've added is C:\Program Files\toontrack.

The only path you need for it to be found is:  C:\Program Files\VstPlugins   - all subdirectories, including the toontrack subdirectory will be found automatically.

11 hours ago, sjoens said:

Very rare and maybe project specific.  Along with the uneven L&R mono track meter.

Most likely your meter settings are different between the track view and inspector view.  The meter settings can be set separately for Track View, Console View and Inspector.

Setting one to pre fader and the other to post fader will almost certainly give this result.

By default the MIDI input for the active track will be "Omni" - change the input MIDI on that track to ONLY your footpedal MIDI device.