Will.

3 hours ago, scook said:

An M.2 SSD is overkill for a system drive. 

Best choice.

3 hours ago, scook said:

A 500GB SATA SSD is plenty for a system drive. 

Definitely too small.

Plugins like Waves, Perfect and Addictive drums, Kontakt (to name a few) - all have huge factory Libraries that comes free with its installation, that requires the content to be in the same folder/Disk space/path - as the installation files.

2:) Your DAW utilizes the same space for recording as your OS drive. So for best results - it is best to have "500GB"  free space available to play with for this, so a 1TB for the OS, the DAW and to run your effects plugins is best here. It also eliminates that "Audio Dropout" nuance as well when theres enough space on your disk for speed recovery. (SSD's has its own limitation too.) If they get overworked when theres not enough space they burnout easily. 

When CPU reaches its limitations it utilizes ram and disk space too. 

I've got a fellow mixing engineer that has the same problem with UAD plugins in Cakewalk. These answers look promising - I'll pass this on to him. 

8 hours ago, Stephen Latham said:

I want to show my friends how I produce music by streaming/screen sharing through an app called discord, however the audio from my DAW is never audible to the viewers who watch my stream. I almost certain it is some setting in cakewalk, because when I play a YouTube video, they can hear the audio from the mentioned YouTube video, but are unable to hear anything from my DAW. I have also tried it with Twitch, but still no luck. How can I do a livestream with Cakewalk?

For this to work if you don't have an Audio Interface. Download the demo version of FL Studio and use it's drivers (FL Studio Asio). It ×10 better than Asio4All and the drivers available in Cakewalk. 

It's a great hack. 

Feature request for the Hardware Output channels "Mutes" to work independently from the "Link" button. 

I like to keep the Link Button Enabled to play things safe, but it beats the purpose of listening to the balance on seperate channels independently. 

Or add the Ctrl+Click to mute the channels separately alone. 

1 hour ago, Noel Borthwick said:

There is no loss of information in fact there is a lot more information than before. 

With virtual instruments Cakewalk groups all outputs into pairs. However the mono stereo listing was arbitrary because there is no guarantee that a plugin output is stereo or not. The numbers now clearly indiscrete which output is being used unlike before. Most other DAWs do the same.

For your purposes you can assume 1 means left 2 means right and 1+2 is stereo. What are you missing?

 

 

This comes back to what I have said earlier. It makes it a little confusing at first, but getting used to it. I think for a long time we were used to the Old Names. 

In fact it's good to slowly move to the language other daws uses in their routing. We just need to get that separate Mono and Stereo track insert strips on both Audio and Instruments now. 😅

3 hours ago, noynekker said:

Interesting you have it working ! My VST3 efx seem to all be in c:\program files\common files\VST3 . . . my 3.5dotnet is up to date, latest Win 10 update patched.

Then you have a serious problem on your system. No one on this forum ever complained about them. Maybe you should contact their support team. 

Fabfilter never give problems. 

Try to remove the VST 2 DLL file if you have it installed. If you would like to keep it . . . get the JBridge Software to convert it to 64bit. 

53 minutes ago, noynekker said:

Hmmm . . . when I saw "improved VST3 support" I hoped the bug regarding inserting Fabfilter Pro-Q3 VST3 had been addressed. (a reminder . . . insert FabFilter Pro-Q3 VST3 into the FX bin, on a new blank project - - -> instant Cakewalk crash) It's been a year and a half, is this on the radar ? I launched a ticket with FabFilter back when it first appeared, but they have not addressed it on their end ? Yes, the workaround is use VST2.

Been working fine here for 2 years. 

Make sure it's in the correct path folder and that your 3.5dotNet is updated. 

Why doesn't the "flatbed menu" work on plugin preset menus too? 

Can we have the "Bounce to Track" to commit the midi file to "Audio" as a shortcut on the right click menu on the clip too. 

27 minutes ago, David Baay said:

Because your original post said "So why is cakewalk stretching the note instead of moving it leaving you ending up with broken information. "

Exactly.

14 minutes ago, David Baay said:

That's precisely why Audiosnap uses stretching instead of 'moving' audio. Stretching lets you move the transient without creating overlaps. This is more important when you're working with audio that has pitched sounds sustaining across transients at a significant level. Drums, and rhythmic bass/guitar with short sustains are the only types of audio that really lend themselves well to the split-and-move approach.

 

Don't know why you're repeating what I've said. 

53 minutes ago, Noel Borthwick said:

Routing is (and should be) a decision the user makes not the program.

100% Agree with you. 

But there's also room for software intelligence. 

28 minutes ago, Xel Ohh said:

‘Duplicate Selected Clip(s) as Linked Clips’ command

To duplicate the selected clip(s) as linked clips, press CTRL+SHIFT+D.

I dont understand what is this suppose to do... I just seems like the same duplicate that was already in place. Am I missing something here?

Bro, I'm with you on this one too. 😂

6 minutes ago, Noel Borthwick said:

IMO it would be more confusing and unexpected for I/O connections to change based on interleave. Interleave affects signal flow not connectivity in the engine.

 

Just asking: Shouldn't it be that way? Having the I/O to read as "Mono/Stereo-in" | or | "Mono/Stereo-out?"

What defines the strips I/O as "Mono or Stereo?" 

Maybe I'm just confusing myself with this. 😂 

2 minutes ago, Noel Borthwick said:

No because the output has nothing to do with the bus interleave setting. All that does is force the  effects track to mono or stereo. It's up to you to route there bus output to whatever you want. I.e routing and interleave are not related in any way

 

 

I Understand.

Although it would make more sense for the input to follow the interleave as well.

It's not a big deal - just a naming/visual thing. 

Just thought some might find it confusing to read. 

2 hours ago, Noel Borthwick said:

Are you sending to a hardware out, bus or aux? Please post a screenshot since its not clear what the routing is.

Hi Noel.

Thanks for responding. No, I'm sending to a normal Aux track.

I'm just asking, shouldn't the input display follow the interleave as well. 

Example: "M: Reverb" to indicate that it's in fact in Mono on the input too. 

Always appreciate new features and changes. 

Awesome work guys. Been at it for 4hours with no issues to this update. 

Just a Question: When changing the interleave to MONO on a send aux - shouldn't the Input Names Reads/Change to Mono too? Right now it say's L+R Filter instead of following the interleave on the channel and reading as MONO FILTER (M: Filter) when the interleave on the "FX Aux channel" has been changed. 

1 hour ago, murat k. said:

It's the same as Set Timecode At Now command but for measures.

It can be done in Cakewalk already. It's just displayed differently. 

12 hours ago, Mike Z said:

 

Its just a time ruler format change and not as what/how you describe in your original post. 

FWIW: Cakewalk has had this for years. 

46 minutes ago, murat k. said:

I think the same way. It could be customizable. Also we should be able to have negative bars.

Cubase have that. But it's changing from the Project Setup. It should be simple as one command connected to the Now Time. 

I mean let's say your Now Time Indicator is at measure 4. You apply the command "Set Measure Beginning at Now Time" then measure 4 becomes 1, measure 3 becomes 0 , and measure 2 becomes -1. Just like that.

You're referring to "TIMECODE-STAMPS." 

20 minutes ago, Base 57 said:

From Wikipedia

"In the 24-hour time notation, the day begins at midnight, 00:00, and the last minute of the day begins at 23:59. Where convenient, the notation 24:00 may also be used to refer to midnight at the end of a given date[5] — that is, 24:00 of one day is the same time as 00:00 of the following day."

😂 I'm just gona laugh at that. 😂 

You're sending back information i'm giving you in answer here. 

LAST TIME: Zero is One | and | One is Zero and that's why the count starts at 1.

So i'm gonna say this "EXACTLY" a 24hr clock only starts at 01:00 

Here's the thing. We've all worked an 8hrs shift before right? But out of that 8hrs you get 1hrs lunch so basically you're working only 7hrs right? 

So from the start you go on lunch until the end of your lunch - thats "1" hour. not 0 hrs right? 

Now to come back to you "Wikipedia" research. There's 24hrs in a day correct! But the day ends on iets 23hr with the minutes and seconds making up for it. Now heres the question: what happens to that 24th hr?

Answer: That's that "LUNCH BREAK" in your 8hr shift. So now on the world clock that count starts at 01:00 again. 

So between 1am and 2am thats One Measure. Your 1 metronome measure counts (Or 4 beats count.)  So on the clock 00:00 to 01:00 will be your 24th Measure and whole note count. ("96 Beats on a 1/4 Note.) 

22 minutes ago, Base 57 said:

When the Olympics start, watch any timed race. The clock starts at 0.00, not 1.00. Even the absolute time counter in Cakewalk starts at 00.00.00.00.

 

As I've said before! Thats because "0" is 1 and "1" is "0"

a 24hr clock ends at 00:00 and starts at 1:00. Where the last count of 12 is at 00:59 "AKA" the start of "1."

The fact that you make an example of an Olympic clock is disappointing, because although it DISPLAY "00:00:00" the count starts at "1" 

That is why "0" represents the number "10" on the abacus. Because the count just starts back at 1 again - which then gives you "11." and every time as you go higher 21; 31 etc. 

I can go on all day with this to why it will never work. 

Unless you want to read the music score wrong. 

49 minutes ago, Mike Z said:

Exactly. A musical score starts at measure 1, not measure 2.  Yet every DAW forces songs to start at measure 2 or later to allow early notes to be recorded.

Yes. But it's time based - as in reality, "real-life." Meaning 0 is 1 | and | 1 is 0  - so this tells you that between the "0" and 1 is a negative count. It's the law of time. We don't start to count at "0" we start by 1. Your house clock starts at 1:00 not at 12:00 because 12 ends at the beginning of 1:00 O'Clock (exactly like your metronome counts.) 

So you see why "0" can't be added to the score as this will add a negative clock count? That is why they teach you this in music schools (Not that i've been to one.)

So all this comes down to a count-in at the end of the day. It is what you're basically asking, but not in so many words. 

FWIW: The metronome count-in does not only applies for recording it's playback. 

That's why your loop points keep your time and song length in order. 

And like I said: I totally understand what your asking. But do you see the complications?

Yes, we all want a way to have that silent free count to make up for starting times for rendering purposes, but this why it cant be done. That's why the only way is to draw it in yourself once production is finished. 

Isn't there an easier way with the arranger? 

7 hours ago, Clovis Ramsay said:

Once the song is ready for mixdown, you still keep that dead space in the beginning so that no audio gets accidentally clipped off 

We just drag out/in silent spaces and add in fades to prevent clipping. 

11 hours ago, Mike Z said:

I've been thinking about this for awhile now, haven't seen it on any DAW yet. I would like my projects to start at measure 0 instead of the usual measure 1. The reason is that I always like to have a bit of silence between hitting the Record button and starting to play. Sometimes there's a grace note or pickup note before the bar, other times it's just an eager note that hits a little early. I don't like cleaning those  up because it can add to the organic feel of songs. If all notes hit at exactly the same time it can sound too mechanical imho.

Normally, I wait until measure 2 to start recording. That means all the measures in the song are advanced by 1 (eg. a 16 bar phrase starts at measure 2 and ends at bar 17).  

I can't imagine this would be difficult to implement; it probably should be an "opt-in" feature since many people will want to stick with the way it's always been. Just a check box in Preferences to start the Time Ruler at Measure 0 instead of 1.

I realize this is a minor issue, I could live without it (have for a long time now) but it would help me stay organized a bit.

 

What you want to do then - is to change your metronome count-in to 1 measure or 4 - beats in the drop down menu seen below. This will give you a pause of 1 measure (Bar) count before the actual recording starts. 

It is a global tradition in music and video for the timeline to start at 1. This has been the law since the beginning of music. 

Reason for this is: Every count starts at 1 and return back to 1 count. Whether that's in triplets or dotted - and its just mathematical correct for 4 measures (Bars) to end at the 4th measure before you get the 5th measure (Bar.) This rule cant be changed as it agrees with "time" in general. 

Meaning: The 1 you see in the timeline  (irrespective of what clock method or counts you use) those beats in between a measure - counts down the seconds. Again: Time in general. Thats why a 1/2 note is 1 second and a whole note count/measure/bar is 2 seconds long. 

It's the principle and law of time itself to have it immediately start at 1. 

I totally get what you're saying and asking - I get it and understand it. That's why a count-down | or | count-in, are so important in music applications on recording.

Who knows maybe it is possible to make a visual aspect of it - like a blank count-in area on the time ruler. I wouldn't know. 

Cheers ✌