Kalle Rantaaho

Is it stereo or mono? Where does the output go - to a mono or stereo bus or directly to main out?

On 8/15/2020 at 11:12 PM, 57Gregy said:

I have never in any iteration of my Cakewalk software seen a template where all of the tracks are already armed for recording. You must have set it that way at some point and saved that project or template.
So, open the template in question, disarm all of the tracks and save it again.

Same here. I didn't understand the question at all at first. Had to read the answers to understand.

19 hours ago, John Vere said:

This is why my habit is to delete as I work.  I found that if all I did was overwrite a bad take, the bad take was saved anyway. It's not clear to me if these get delete when you close so I just delete as I go. I always use overwrite option for recording for this reason too. It's a habit from the days of tape where you had no other choice and it suits me fine. 

I do ctrl+Z after every failed take, so nothing gets saved. Of course it often  happens I keep several potential candidates to choose from later.

On 2/24/2021 at 4:06 PM, DeeringAmps said:

My guess is the mp3 is 44.1k and you’re playing it back at 48k. Check your interface settings. Rarely do I use real-time bounce when exporting, unless I use “audible” bounce cause I just want to hear the track. 
t

One always learns more here...

I thought the sample rate/tempo(speed) issue only goes with wav-audio files, and doesn't concern mp3-data files.

My suspect #1 is the mouse as well.

One way to make good unisons is bounce the track to audio to create the second (or third, fourth..) instrument, and edit the audio track with Melodyne.
Pitch a little up and down here and there, minor timing changes, and maybe even a very gentle formant change to add a little depth. That is also one way of handling phase issues.

I wonder, how much  unsaved actions do you have there when " the number of clips get to a certain point" and the issue occurs?

Would it slow you down or have other unwanted side effects, if you simply bounced to clips and saved more often? Not necessarily the the whole track length, but a nice, appropriate number of clips a time. Maybe you have tried already.

How often do you normally save while you edit? In operations like that I'd recommend saving everytime a clip is edited, at least. 

3 hours ago, martsave martin s said:

to the OP you know you can upgrade to cakewalk buy bandlab for free...?

Unless the OP is in the same situation as I am  :o/

My music PC is so old that I'd have to get a whole new PC, audio interface and all the VSTs .  I'm looking forward to it as I look forward to a costly nightmare.

I just hold my breath and hope it works one more year.  As it is, it does everything I need it to do.

18 hours ago, Colin Nicholls said:

I recommend adding your vote to this feature request

 

 

This has been discussed again and again in the past. I would welcome negative bars warmly. A small thing, but quite useful.

5 hours ago, John Vere said:

Your using ASIO 4 all?? I wouldn't expect studio quality sound from that. Are you not using an audio interface? 

Your workflow seems very strange? 

The midi file plays the vst drums. The VST instrument is in a audio track ,  There is no reason to "record" the VST.  Nobody needs to bounce instruments anymore unless your computer is 10 years old or something. When you export a song the audio output from any VST instrument is treated exactly the same as a audio track of a guitar or vocals. 

Why are you using a ridiculously high sample rate? That's asking for trouble too. 48 is all anybody really needs but if you think your fans will notice  while listening to the MP3 on $10 ear buds-- then 96 is more or less way out there. But I wouldn't attempt that on a system without a good interface and ASIO driver. Asio4all is not really ASIO, it just pretends to be. Its a wrapper for WDM mode. 

That's what I thought, too. I think the OP has somehow misunderstood the idea of  the use of EZDrummer , but I'm not sure.

On 1/16/2021 at 6:58 PM, msmcleod said:

Do this:

• Create your audio track
• Open up the take lanes (this will create one take lane)
• Drag all of the files to the take lane

This will put each clip in a separate lane within the same track.

To make it a single clip:

• Select all the lanes
• Right click one of the clips, and select "Bounce to Clip(s)"

You may want to adjust the overall clip gain level after doing this, and Bounce to Clip(s) again.

What defines the order / timeline position of these clips that are drag-dropped in one go? The order in which they've been selected?  Unless they are Broadcast waves with a time stamp. Or do you have to move them oe by one after dragging? 

Often the blank space in the beginning is left there to allow smooth MIDI playback start or for a similar reason. So one might want to leave it there for future sessions of the project.

If you simply select the desired portion of the project in the Time Ruler, then only that part gets exported - doesn't it? Or does my memory fail?

17 hours ago, bdickens said:

That would be weird. Any purcussion hit should ring for its full duration no matter what note value is drawn in. 

That's what I thought, too. The length of a cymbal MIDI-note should be irrelevant. There is a separate note for muting the cymbal, IIRC,  isn't there?

23 hours ago, Marcello said:

I thought that with mono you could not pan the guitars.

A mono track can be panned anywhere in the stereo field. Wherever you position it, you will hear the whole signal.

(the following at  least used to be true in SONAR, someone will correct if I'm giving you old info)

Pan knob on stereo tracks is actually a balance knob.
If you simply pan a stereo track, say 100 % left, what happens is, the right side gets simply muted.
If you have a piano recording with left hand parts on the left channel and right hand parts on the right, you don't hear the right hand parts at all, unless you do the panning with a channel tool VST which allows you to position each channel to your liking.

"...with everything else" meaning..? With other DAWs? MP3s? Windows Music Player?

So you do get something recorded, but only output is the problem?

What is the sound you're trying to hear? Something you imported into CbB or something you recorded yourself?

Do you see activity and waveforms on the tracks, but nothing's audible?

What kind of a setup do you have? Mobo soundcard or an external one?  How are things connected?

5 hours ago, John Vere said:

You keep saying Sonar? this is important because a lot has changed since Sonar. Like the workspace and customizing views etc. Instead of the developers spending 5 minutes, I think it's best to just spend time exploring the GUI until you know everything there is to know about it. Yes we all want software to work the way we think it should, but that is a rarity. I used to hate a lot of the way Cakewalk ( sonar) worked but when you actually learn all there is to know  about the GUI--( impossible) --you might just find it's a lot closer to what you wanted. You just have to know where to look. 

 

Exactly. And knowing what version of SONAR it is would be equally important. The OPs posts don't show any hints of him knowing where we are now (?).

Otherwise, I assume, he would somehow  refer to using old software or something.

22 hours ago, Kurre said:

I don't like this deleting of subtones and overtones. The end result is then a false representation. (Even if you can't hear it.)

And to delete anything lower than 150 or 250 hz?

"From wikipedia:  The voiced speech of a typical adult male will have a fundamental frequency from 85 to 180 Hz, and that of a typical adult female from 165 to 255 Hz. Thus, the fundamental frequency of most speech falls below the bottom of the voice frequency band as defined above."

Female Singer Type: SOPRANO/C4 - C6 (261-1047 Hz)   MEZZO-SOPRANO/A3 - A5 (220 - 880 Hz)   ALTO/G3 - G5 (196 - 784 Hz)

CONTRALTO/F3 - F5 (174 - 698 Hz)

Male Singer Type: COUNTERTENOR/E3 - E5 (164 - 659 Hz)   TENOR/C3 - C5 (130 - 523 Hz)   BARITONE/G2 - G4 (98 - 392 Hz)

BASS/E2 - E4 (82 - 329 Hz)

Explain yourself, please.

I've never recorded a female voice, but I assume I wouldn't cut it any harder than a male voice (?). Cutting  at around 50 Hz cleans the accidental proximity booms of my baritone voice, I don't want to cut anything more than just problem frequencies which add to volume but can't be heard. But as mentioned, I've never faced those of a female voice. I'm an amateur, not so high goals. Such tricks often  help you out if you have lacking monitoring equipment or environment.

Trying to fit together something like piano, bass guitar or contra bass,  and a male vocal  (maybe a cello,too) makes one really sweat with what to roll off and what to emphasize. All the most loudness creating low frequencies more or less overlap. I remember how I struggled with my first rock song which included a piano. I ended up wiping out practically all left hand piano parts, cause I couldn't make it work with the bass guitar. 

Vocals are very vulnerable in tracking, small changes in mic distance, face direction, sibilants, etc. can make a big difference.

Not knowing what type of vocals we're talking about, what I would try is volume automation on the vocal track pulling the loudest peaks a little down, then EQ cut
the low end  somewhere around 50 Hz (depending how low your voice is, is there strong proximity effect etc.).

After that try again, with the backing tracks a little lower, and maybe  gently compressing the vocals.

A very common thing to do is also pulling the backing tracks a bit down only when there is vocal on top, but not drums and bass.

3 hours ago, lapasoa said:

Don't complicate your life.

Mute all tracks except the tracks you want to mix together.

Choose Bounce to track.

Get the normal procedure checking mono without touching anything else. Check OK and Cakewalk create  automatically a new track.

After that remember to delete or archive the original tracks you wanted to mix together

Exactly. It has never even popped in my mind to think any further: Select the tracks and bounce to track. Ten seconds and done.

Please refresh my memory.

I remember reading here (or in the old forum) several times, that you should not export/master to 0 dB level if you're going to later convert the file to mp3, but rather export to -0,3 dB max. This being due to slight level raise in the conversion, and possible digital distortion/clipping in the mp3-file.

I expressed this view on another forum and I've been "challenged" to give some background to what I wrote.

Is my memory failing? 

18 hours ago, EdStauff said:

I found a workaround!

1.  Add a new short temporary note that ends at the same time as the Very Long Note. 
2.  Select both notes, and then you can drag their ends to lengthen or shorten them.  
3.  Delete the temporary note.

Not a perfect solution, but it beats the currently available alternatives.

-- Ed

Creative thinking!

Channel name? Is that a typo?  Do you mean Project name?  Is it possible there's a weird symbol somewhere you're not looking?

[...how to best restore a 44.1 24bit recorded session from tape.]

I understood this so that it's a digital tape recording, but maybe I got it wrong ? There would not be tape hiss or unwanted frequencies (due to tape) on a digital tape (?)

So the question is how to transfer an analog tape recording to 44,1/24 ?

I've used  CW/SONAR since early 90's. I don't recall ever reading, hearing or seeing of something being erased in track view.

Neither have I ever needed to have Write on in order to make Read work. 

AFAIK to get rid of a part of a clip has always included either using Split/Delete (or Mute) or Slip editing.

6 hours ago, Billy86 said:

Is that sheet music something CbB produced from your song? It would be cool to see sheet music produced from a song. 

Isn't everything you do with MIDI always presented as sheet music as well? Audio is another story.

I've printed sheet music from SONAR already ages ago. Although the staff functions were not very good.