CJ Jacobson

2 minutes ago, August said:

I meant a headphone amp, sorry. Do I use a headphone amp if I need more headphone outs?

Yes, if you need more headphone outs, then you need something that can give you more headphone outs

13 minutes ago, August said:

So I should use my interface in conjunction with a sound card?

Your interface is your sound card. Use one and only one. Disable your onboard sound chip and use your interface

25 minutes ago, August said:

Ok, but how do I get my 2 pairs of headphones and my monitors working?

Should I get a new interface?

If you need to use both headphones at the same time, you either need an audio interface with 2 headphone outs or an interface plus a headphone amp. as for your monitors, they connect to the main outs 1 and 2 of your interface and have nothing to do with your headphones connections.

If you just need to use one pair of headphones ,then you connect that to your headphone amp of your interface and your monitors connect to the main outs 1/2

9 minutes ago, August said:

Thanks for the reply, I don't think you understand. I use ASIO4ALL because I have my Scarlett AND my integrated sound card.

I understand completely.  You cannot use 2 different sound cards. you will have problems like clocking and other problems, as you are seeing first hand. Use one card and use the driver written for that one card

1 hour ago, glen dale said:

Does anyone use the onboard realtec sound device for playback and recording? How have you configured Cakewalk?

Onboard sound chips do not have good drivers, pre-amps, good line inputs, good circuitry and have sub-par converters. you can get a sound card made for recording for very little money. If you want to record, i would invest in one.

MOTU, Focusrite and RME are good sound card companies that make great interfaces for recording

14 hours ago, Jack Hawk said:

when I exported the finished track, after modifying the tempo, if was showing over the -.1 limit I set prior to mastering.

Is this inner sample peaking or peak dB?

After a mix is mastered. It is dithered down to 16bit and exported to 44.1kHz is most cases. When you import it back into a DAW, it will import to your import settings and most are 24 or 32 bit and what ever sample rate you have configured. This can cause your levels to fluctuate.

This is why the mastering is the last stage of music production. 

My advise is to go back to the un-mastered mix to do your tempo changes, pitch shifting and what not and then master it. 

19 hours ago, August said:

This problem only occurs within cakewalk, though. I've tried NI Guitar Rig and Reaper, and it works just fine.

Drivers settings can be adjusted in Cakewalks. 

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I used ASIO4ALL because of the lack of Realtek Drivers.

Why are you using Realtec drivers? This is a major problem. You need to use the drivers that were written or your Focusrite, not some POS onboard 5 cent sound chip.

do not use ASIO4all and do not use Realtec drivers. Use the Focusrite drivers. 

12 hours ago, August said:

A few weeks ago I noticed an issue with latency when trying to use TH3 with my guitar. There was about 100-150 ms of latency when echoing the tracks. This issue also occurs with MIDI Instrument tracks. I've tried it with XLN Addictive Keys and the SI Suite. Any help is appreciated!

If your audio interface doesn't have good enough drivers and/or the drivers are set up incorrectly for low latency, you will have this problem.

Also, if your PC and MOBO are not up to par,  it can cause latency also. 

If you have a decent audio interface and its up to date with drivers and a fairly new PC, you can get low latency and use some effects in your DAW. If you are using ASIO driver mode, select 32, 64 or 128 buffer size. The lower the better. Your input latency should be under 5 msecs and your total round trip latency should be less than 12 msecs, if you do not want to hear a noticeable delay

What VST are you installing and are you choosing custom install or the automatic install?

Most VST dll's should  be installed on your C drive. This is your main drive

On 8/5/2020 at 9:55 AM, Nick Rosaci said:

Anyone having every note on every virtual instrument hit when autosave goes to work?  I have autosave set for every five minutes, and it causes me to practically jump every five minutes.

I could turn off autosave, but that's playing with fire.

I never use Autosave. Just keybinding the letter s to save and press 'S' often. After a session, it becomes 2nd nature to press 'S' every other edit or so.

On 8/5/2020 at 6:39 AM, Clare Sudbery said:

According to what I see on the screen in the piano roll view, and also according to the event view, they both start at the same time on the same note. But when I play them, one of them starts a split second behind the other and they are not playing simultaneously. I can't find anything in the documentation that would explain why this is happening?

Always, I means always start every project on measure 2, beat 1. Not measure 1, beat1

When you freeze a track, everything you do to it will be erases when you unfreeze a track. You can try it out to confirm this.

On 8/5/2020 at 7:28 PM, Jake Sharar said:

So I got everything installed alright, got a few plugins that sound great, everything's great except-----

every time i hit the play button, it plays for .8 of a second and just stops...... idk how to fix it or what it could possibly be. 

If its dropping out, then try raising your audio interface's drivers. This is done in the preference menu.

Those types f plugins are CPU intensive and mainly meant to be used for mastering.

You may need to raise your audio interfaces buffers (ASIO) to help your PC with just an i5 cope with those kinds of plugins

1 hour ago, Sunraw said:

After bouncing edited audio the project plays back at full volume so the master can't be turned down or muted. I find it starts happening after loading Melodyne but also after restarting the program and trying to bounce a take without a  Melodyne region.

Are you playing the bounced audio that you bounced through your master bus, through your master bus again? This means you are running it through the same gain stages as your bounced it and that should not be done.

19 hours ago, Morten Saether said:

We appreciate the fact that different users have different preferences. The next release will have an option to center the Now Time when zooming with the keyboard.

Thank you and this is why I've been using Cakewalk for over 20 years

On 6/16/2020 at 11:05 PM, Colin Nicholls said:

Um, I have to admit to not liking the new zoom behavior either. I didn't realize it was "broken" in all the previous versions of Cakewalk/SONAR.

@Noel Borthwick, any chance this can be made a user-configurable option?

+10,000,000,000,000.06

Bump 2

bump

You would think it would make sense when you zoom with a control surface that it will center ware the timeline is.  I would think that would be obvious, but its not!! The latest updates took this option away. Why?? 

When you zoom, don't you want to see ware you are zooming?  Who's decision was it to not have it centered with the timeline?

14 hours ago, brandon said:

When i tried to record a 2nd audio track in a new audio track using the same settings (i.e. 1st input and 1st output) from the Focusrite  there was just silence. No wav file was recorded.  Is this a feature or a bug?

Have you selected the "1st input" as you call it, for the new track you want to record in? 

edit

On 5/20/2020 at 4:33 PM, Frank DeFede said:

 

On 5/20/2020 at 4:33 PM, Frank DeFede said:

I thought I had 22 songs recorded, mixed, and ready to release, but I am concerned about the output level.  For some reason, I have gotten into the habit of pushing the level as high as possible without clipping using a Brick Wall compressor.  The mixes sound good, but I brought in  reference tracks of other commercial mixes like Bill Joel, Maroon 5, country artist, etc. and they all seem to be much lower than my mixes.  

I think I may be pushing it too hard.  What is the general acceptable level on the meters?  Most of my mixes are all the way to the top, but without clipping.  Any suggestions are most welcome.  I hate to do it, but I think I may have to remix everything again.

Thanks

 

Frank, a good target to go by is to be between -12 to -14 LUFS and leave about -0.5 to -0.3 Below 0dB for conversion errors.

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I usually leave -3 dB headroom for conversion to mp3 and other formats.  Conversion can redline a 0 dB track.

Alan, 3dB is a little bit of overkill. 0.5 to 0.3dB on your final master  is sufficient for any kind of conversion errors. If you have a track at -3dB PEAK and you convert to another format and your signal reaches 0dB as you stated, then that conversion program is a POS

You need to go into the Preference Menu and select 'Share drivers with other programs"

Adjust your ASIO buffer until it stops doing it.  There are no one setting for all with buffers.