John Vere

I guess the OP is talking automation. I have about 6 wave editors and none of them "re-draw" waveforms. It would be interesting to hear what that sounds like. 

And Lee are you aware that the latest version of your software is available for free? X3 is real outdated and huge improvements have been made since then. 

https://www.bandlab.com/products/cakewalk

When I've had issues like this there's been some sort of hidden data that I cannot see in Cakewalks event list. Close inspection of the track inspector might reveal something. Like try re setting the volume slider or gain and see if it moves when you start the song. Someone mentioned Ample Bass and this will often happen where it will jump to 1.6  in the GUI. I have to lower the midi track level to 64 to re set it to my pre set of 1.0. Never had to do that with other VST's.  

Tricks I've also use were to change channels ( didn't always work) or slip edit to exact first midi event  or split and delete the " empty" data before the first note. 

Instead of a Flash drive I would invest in a nice 1 or 2 TB external drive. They go on sale for around $100 and then you need not delete or compress anything. I have 4 external drives and 2 of them are only for Cakewalk back ups. I date the whole folder which will have the sub folders of each project in it. Drag and drop to the external drive and carry on. 

This seems stupid but did you re boot your system a few times after the updates? 

The focusrite Scarletts run at about 20 ms RTL at a safe buffer so I myself cannot engage the input echo without hearing a very noticeable delay.  

3 hours ago, Kenrout said:

Hi John, fell at the first hurdle, whilst looping the track, i can see the level on CW moving but getting nothing from the headphones!!

OK lets start from over and for now forget the mixer it only ( as other John has said) complicate things. 

Just hook up the Scarlett and lets get sound from Cakewalk. 

Things to check if not already done. Make sure the Scarlett is the audio device when you click on the Windows speaker icon.

Make sure in Cakewalk preferences the scarlett is the only audio device selected and you are in ASIO mode. 

Make sure the Master buss is set to output to the Scarlett and the tracks you are playing are  set to output to the Master. When you play the song the Master buss meters are moving. 

Plug headphones into the scarlett you will hear the playback.  Or plug the studio monitoring system into the rear outputs, same difference. 

Now plug in a mike or guitar and engage the direct monitor switch. You should hear both the playback and the input signal. 

I do not recommend using the Soundblaster while trying to work with Cakewalk. You have a proper audio interface , use it. Creative drivers are notoriously invasive and will cause conflicts with the Focusrite drivers. I speak from experience as I have Creative PCI sound card on my old computer. 

The only short coming of the 2i2 is the monitoring system. Most interfaces have a blend control for playback and input. They chose to only have a toggle switch so you have to lower your playback level inside the DAW which is a PITA.  Your mixer could be used to solve that issue but lets just get sound first using the interface. 

Cj- the cue mix, as it's name implies, is for monitoring only and it has no bearing on export.  I can turn my monitors right off in Cue mix and it still exports correctly. I've personally never experience any issues with export.  I don't use Cakewalk for direct MP3 because I need to master the file first in Wave Lab. Once I'm happy with the Wave file I convert using Gold Wave which is another wave editor I like. Wave Lab for some reason still wants money to unlock MP3 encoding just like Sonar used to.

Any how, I use 190 kbps as that seems just fine for where those songs are going. 

To Meld Magic- If you use a real audio interface you won't have any of those stupid settings to worry about. 

How I have always done this without issue is Ctrl A to select all. then go to -

Tracks / Select track outputs/ Then select the audio buss 

That interface is even older than my M Audio Fast track pro. On the M Audio web site they clearly state that the driver may or may not work with W 10. For me the interface works fine on one PC but will not install on another.  You need an up to date audio interface no matter which DAW you choose. Look on Sweetwater for starters if your in the States.. 

And back in the Windows XP days the M audio interfaces had the worst drivers ( so did Tascam). So for the very beginning the OP has been fighting the bad driver issue. Things have changed and USB audio interfaces all seem to work fine as long as they are newer and have proper W10 ASIO drivers. Old model drivers are sometimes updated but not recommended if you want solid stability and good performance.  

By default my track templates all have that compressor and the EQ inserted to pre channel. They are turned off.  Then it's a very quick move to solo a track, open pro channel and turn them on and tweak. If you want a quicker fix than that , you should take up a new hobbie  

In my case it was not an upgrade, just a registration and serial direct with them. I get e mails all the time about upgrades but I'm more than happy with the version I have. Mostly I like to use the chorus and compressor pedals on acoustic guitar tracks. I still prefer my real guitar pedals and amp for electric. I need to play into the sound, can't stand playing dry and my system has too much latency to use the sims live. 

OK so I looked at your mixer and what might work is the following

You'll need to use to short patch cables and connect the Main outputs to the 1/2 inputs of the 2i2.  This is not ideal as those inputs are made for low out put mikes etc so be careful to keep the levels low while you experiment. These will send whatever you have in the mixer to the DAW. If your recording vocal and guitar at the same time you will need to pan those hard left and right. 

Then for monitoring  plug the output 1/2 of the 2i2 into the 2 track inputs. Hopefully you have a cable that can do this. It would need to be 2 RCA to 1/4" cables. There are also adaptors available but best to purchase the correct cable. 

Now by using the Control room /headphone levels you'll hear the DAW playback and what ever you have plugged into the mixer. Make sure to only engage the " 2 track to Control room" button. If you engage the 2 track to mains it will loop back to the DAW and feedback. I think you'll have to turn up the " phones/cntl room level found next to the mains volume.

That said there is not a big advantage to using the mixer as it has limited connectivity. 

Sorry I made a error which I will correct,  I said Right input for input 1 when it is the opposite. The way most interface show as inputs is using this as example.

Left - name of interface- mic input 1 =  Input 1

Right - name of interface-mic input 1 = Input 2 

Stereo - Name of interface - mic input 1 = Stereo both inputs used, this also automatically creates a stereo recording in your track. 

If you have more than 2 inputs it would carry on using only odd numbers 1, 3, 5, 7 etc. 

So 1 is for inputs 1 and 2,   3 is for inputs 3 and 4 and so on.  

In the past Scarlett 2i2 and solo's didn't use the mix control software, not clear about the 3rd gen. 

But I have the 6i6 and they work the same. I have also set a friend up with a Solo. This example we will use a guitar and vocal set up. 

Make sure you have the very latest driver.

Use ASIO only do not use any other driver mode for overdub recording or timing will be off. 

You need to plug the 1/2 outputs into your studio monitor system. If you don't have studio monitors get some.  Computer speakers will not give you accurate mixes.

In a pinch you can plug computer speakers into the headphone jack using an adaptor. 

Use the direct monitoring option other wise you will hear a small delay due to system latency.

Plug in a guitar to input 1 set it for instrument ( depending on guitars system) . Set the input gain below where it peaks. You don't want to ever see it peaking. This is the level for your recordings. Your DAW's level and volume controls have no bearing on recorded levels. 

First plug in some headphones and adjust the level to hear guitar.

Open Cakewalk and from start screen select New project and the 4 track option. 

Set track 1 input to LEFT input 1

Set DAW to record standby and observe meter to check it is not peaking, record a few minutes or a song. 

Playback what you have the sound will automatically go to your 1/2 outputs and your headphones when you use this template and you have selected the Scarlett as you only in/out in preferences.  

Now plug a mike into input 2 and set levels as above. 

Set track 2 input for RIGHT input 1 set in record standby check meters while sing loudly. 

Now  play the song  while practicing your singing. You should hear the playback in either the headphones or the monitors ( 1/2 out)  as well as your voice. If the playback is too loud with the 2i2 your stuck with turning down the master buss or the guitar track ( or creating a headphone buss) . Most interfaces have a input/computer adjustment control but the 2i2 is lacking this important feature. 

This should have everything working properly. 

You can make up for it's short comings by using a mixer but the mixer needs to be configured so you can monitor playback without it be re directed back into the recording output. Tell me which mixer your using and I could help with that too. 

Steves answer is what happens 99% of the times you will have your issue. Way down the time line there is an auto punch active and only that section can be recorded to. Otherwise I cannot think of anything else other than I hate instrument tracks so would try recording to a real midi track to see if that works. Add a midi track and set the output to the VST your using. That has worked for me for 20 years  

Back when Roland ditched us Overload gave Sonar owners a full unlock serial if you send them a screen shot of your Sonar Splat serial found in Help/ About tab. I thought that was very nice of them. So maybe it will depend on if you took that step.  

You use the word normalize which is a tool used when your not happy with a track being a low level and you want to bring the level up. But all that can do is push the PEAKS to the near limit. It won't actually make all the audio even. All's it take is one popped P in a track and there goes your peak level. 

Myself I would treat each track separately and use the PC76 U in pro channels to adjust the input gain until you see the compressor working. If a track is weak you can certainly normalize it but using gain is none destructive.  I would also EQ each track.  Basically solo each track until it sounds as best as it can.   Then send the 20 tracks to a bus for further processing, reverb etc. 

Yes definitely no reason to be using 8.5 anymore.

Another solution which is how I transfer projects to other DAW's is simply save as a midi file and then export the audio as individual tracks. ( stems in Pro tools speak) I open the project as a midi file in the other daw and then just drag and drop the audio to new tracks. Its a bit more work but works great. You seem to understand the prep protocol for the audio but for others reading this I'll mention that you need to make sure all your audio tracks start at zero and exporting a track will create a wave file that runs from start to end with no parts missing. 

I seem to remember the OMF method being unreliable. 

I have always used the first synth option. That way I have a track for the VST and other tracks for the midi that is assigned to the synth. Example I might break a piano into 3 tracks for editing and keeping parts organized. drums I often have 3 or even 5 tracks. That way I can assign parts to different VST's.  I will use a track folder if I need to tidy up. There's really no big difference, it's all a matter of how you want things laid out on the track view. I like to have the VST track on top of all the corisponding Midi tracks and I colour code them the same too. 

Traditionally in mixing you only HAD one reverb to work with and that might have been a Plate in the basement. People are sort of used to that and also in the real world if a band is playing in a nice auditorium or hall, then that is the space ( reverb)  around the music. We close our eyes and we imagine we are in that space when listening.  So that said it depends on if you are trying to create a " real space" or an "unreal space" .   To me it jumbles up the music with a lot of different spaces and (depending on the music) can make it tiresome to listen to.   

So generally I use only one reverb on a buss set at 100% and then send my sub busses or individual tracks to that. This way the "band" is in that same space.  

The only other place there will be a different reverb is on a guitar amp sim because that's a classic sound. And then some synths often need a little help. But I'm very cautious about using too many "spaces" in a mix. 

What is the soft synth?  Is it a 32 bit?  ,, a freebie? 

Hey Lars.. I never monitor at the back end of Cakewalk as for me that is a good 20ms or so. I have always used direct monitoring via my interface and have no issues with timing.  My Yamaha digital drums tend to put notes a bit ahead of the beat but I generally will be quantize anything midi so it is not an issue.  

Only time I don't quantize is a piano part played by a real piano player. And with the drums I sometimes like the way the snare sounds just a hair ahead like that, so I leave it alone. 

I'm all about music having a groove and a feel so am very picky about timing. I play the parts pretty dang close to where they belong and if my software messes with that I'm not happy. 

If your ASIO driver report 4978 latency something is really f***d here.

Usually control panel for driver is having a setting, like 128 samples or whatever.

Not necessarily. That might be normal for that interface at a higher buffer setting.  That is the sum total of samples. 128 is a medium low setting,  I run my system at 256 and my total reported roundtrip is 1042 samples. 

My Tascam set at 256 buffer=  input  324  + output 1087 total reported RTL 

if set to 512 buffer it almost doubles = input 580 + output 1625 = 2205 reported RTL 

  On many low end interfaces there can also be hidden buffers. 

But none of that matters because if the driver is reporting the RTL correctly then Cakewalk adjust playback so that what you hear is put down in the correct place on the timeline. You can perform a loopback test to check your system is reporting correctly. use this tool  https://oblique-audio.com/rtl-utility.php

That said midi does not use your ASIO driver, It uses MME. There is no adjustment for midi timing offset as far as I know. The audio is involved with overdubs because your playing along with audio. 

Thanks Craig, The M Audio is not important and I was amazed it worked great with the one computer. I wasn't expecting it to work at all.  I got it for free from a old friend who was tossing it out.

On the support web site they warn you that it might not work.  but I was impressed they showed support up to that point ( W8) 

Funny thing about the Fast Track pro, it was my very first USB interface way back in I guess about 2006. Back then it had real iffy drivers and would cut out on me for no reason. It did this one night while I was doing sound for a dance show. Very embarrassing.. I got mad and bought the Tascam us1641 which  was dependable under windows XP but not W7 when I moved on.  Eventually both companies got their drivers working smoothly.  I was real surprised when Tascam updated the driver for the us1641 in 2014 and then again in 2017. To bad they never updated the drivers for the UH 7000.