Bass Guitar

Troubleshooting needs to be a process of elimination as your issue could be caused by a few different things. Here's a few thoughts I would try. I said to Bypass ALL the effects as those could be the cause for different reasons. This eliminates things you might have missed like Pro channel modules. It could be a drop out that only happens at that point because something is taxing your system. Like an effect or a VST instrument. So increasing your buffers to around 520 might solve the issue. You have a new Audio interface and there's no telling if it will perform better than your old one. Example I was surprised to find my Motu interface would drop out at the same buffer setting that all my other interfaces didn't drop out. Does the Apollo have DSP processing? Turn that off if it does. Freeze the Midi instruments.

I've only had it happen in one track. If you zoomed in there was nothing odd showing. So I just split the track there and add fade outs and fade ins and it would be gone. but your saying it's across multiple tracks and each one makes the sound when soloed? Try by passing all effects.

Freeze midi track/ Drag to timeline/ Unfreeze midi track. Open the tempo track and Inspector view of it. Now turn on ripple edit/ turn off snap to grid. Drag midi track until downbeat lines up to measure grid properly. Now check that every downbeat is lining up properly. You might have to edit the the tempo track. You can also use Set Measure Beat at now to edit anything that was wrong. But for me this has always worked for most of material. The stronger the transients the better the success. You can also do this in Melodyne stand alone or from within Sonar and export a tempo map. But it needs to be audio first.

Sorry I just saw this, Apple Auto correct strikes again. I had typed Dude! I totally didn't wish for a that "tone of voice" I hate the internet!

Seems Next has almost been abandoned or something? 61 posts here compared to Sonar 1.7K is sort of pathetic isn't it? I don't think it caught on in the DAW market. It's sort of an Accessory to Sonar so we can do stem separation.

Duh! You do realize that Cakewalk has stopped being updated almost a year ago. All updates are applied to Sonar. Cakewalk is frozen in time now forever. Go to the Sonar Sub forum and post your request there.

Yes. When you drag and drop ( import) any media Cakewalk/Sonar automatically converts it to WAV at the projects sample rate. Only the bit depth is retained. Try Gold Wave. It has a batch convertor that I've used for a long, long time now without issues. It's free to use with Nag screen on start up but I bought a lifetime license 15 years ago. I still get free upgrades. https://www.goldwave.ca/

You might want to start a new topic. A lot has changed since this was posted.

Ripple Edit. If on, any edits you make will be global to all tracks. Example- Delete one measure and the tempo map, automation lanes etc, will adjust accordingly.

It depends on the keyboard manufacturer. Korg, Yamaha and Roland and many Top brands will have the replacement silicon / carbon contact strips. I found Akai doesn't supply them. My Yamaha keyboard was straight forward to do but I counted 45 screws were involved!!

You should submit the project to staff. Your Export settings look exactly like mine and I don't get that message. Next is still in development and bugs are still going to happen.

You should submit the project to staff. Your Export settings look exactly like mine and I don't get that message. Next is still in development and bugs are still going to happen.

You should submit the project to staff. Your Export settings look exactly like mine and I don't get that message. Next is still in development and bugs are still going to happen.

Turn off snap.

What might work is to use SELECT ALL ( CTRL A) before you open the export dialogue.

Next can run different audio file formats. In other words unlike Sonar, if you drop a 48 Wave file into a 44.1 project, it does not convert the audio to 44.1. It leaves it as 48. BUT, what happens is now Next will have to convert this in in real time so it uses more resources to do this. Page 56 of users manual is one place you will see this mentioned: "Match the project sample rate and audio device sample rate It is strongly recommended that you use the same sample rate for the project and audio device. Using different sample rates will cause real-time audio resampling, which may consume extra CPU. Real-time sample rate conversion ensures that projects can play back on virtually any audio device at the correct playback speed, even if the audio device doesn’t support the current sample rate. The default project sample is Auto, which adapts to the audio device sample rate." Next stem separation returns the tracks as 44.1/16 even if they were 48/24 to begin with. I guess this is to save bandwidth on the server. I've done a lot of stem separation stuff and I always record the song I want off of You Tube directly into Next. I normally always work at 48/24 but I set up Next to use 44.1 because this is all I use it for. I then run the stem Separator and as I said it's done a lot of songs for me this way. including I did a bunch of my own originals that I didn't have multi tracks for and wanted to re mix them. But the audio is not that great for those projects.

I had a Samsung SSD drive that was only one year old cause problems. Everything was SLOW! The drive was only at 60% capacity. But apparently it's all about how many read and write cycles the have had. Audio and Video is a big problem for SSD drives. I replaced it and it was like getting a new computer. I cleaned out a lot of un wanted stuff from the original bad drive before I cloned it.

100% agree. They are good meters. I'd even say better than most other Daw's. They have a peak and hold reading, use it. If you want to use a meter plug in then the You Lean Meter will tell you all you need to know. The paid version is worth it because you can drag and drop your tracks into the GUI and get a accurate reading in seconds. LUFS are just as important as peaks. https://youlean.co/youlean-loudness-meter/ If I used -18 on my tracks the overall mix would probably be way to quiet. But I realize many people advise working that way. In my world each instrument gets a different peak and I have a system that seems to work for me 90% of the time. Example my targets for bass and drums is -8 to -6 db . Because they are midi it is easy to maintain this level using velocity and setting of the instruments. Then Keyboards and rhythm guitars might be - 10 to -12. Vocal and lead guitar depending on music style, get as high as -3db. They are brick walled on to catch random peaks. I might use Melodyne to clean those tracks up and set a close level. Or a volume envelope. . Of course EQ is a big part of levels and LUFS. But I start with these ball parks on each track. Then I use Busses to mix. My resulting mixes are around -1.0 and -14Lufs. I put a brick wall on the master and I'm just about there.

My only advice would be to: Turn off the Ozone Limiter. The following is just a very, very simple approach and is not at all what a real mastering engineer would do. But it's all a beginner needs to know to get started. Use the You Lean Loudness meter placing it last in the signal path of the Master output FX bin. Then I would simply use the gain on the Master bus to set it so the You Lean is showing a ballpark target. You should open the Pro Channel EQ and the fly out and see if you have too much low end energy. That's usually where the problems are. There might be a few peaks so now you turn on Ozone to fine tune things. I would never trust it to do what can easily be done manually first. The only limiter I found works flawlessly is the Loud Max https://loudmax.blogspot.com/ Free too. I've been messing with Sonars Concrete Limiter but haven't really gotten a solid way to use it. The Loud Max is dead simple. Set it at -1.0 db and forget is. The You Lean might still show a reading of -0.8 but that's because it works at the sample level. Don't trust anything else. If you get the full version it is worth it's weight in gold. Drag and drop any media files for instant readings. Make sure and turn off the automatic drag and drop normalizing it seems they have that as the default in the new version.

It shouldn't is you are using an ASIO interface with the mike connected to it. You would need Input echo on to talk live. That's were OBS is easier to use. It has a multi channel mixer that will use your live mike as well as play from any daw or music app if your interface supports Loopback. I do a lot of this. But yes Loopback is a must have for podcasting or any live stuff.

I can only guess without a screenshot that shows the track header and routing. A console view shot ?

I've always used Entire Mix.

For streaming that always works fine with Sonar. OBS studio- Free and used by millions of people for podcasting and screen capture. https://obsproject.com/ You need to make sure you don't have "exclusive control of this device " on. Sonobus - Free. https://sonobus.net/ You just insert it as a plug in on Sonars Master bus and set it up to send over your network. Anybody in the world on the internet can listen with only a little latency.

There is no theme editor for Sonar. The only theme with a white background is Mercury. This is the thing that sucks the most for me about Sonar. I need the white background or I'm squinting at the PRV screen and then the rest looks terrible if I change to Mercury. I managed to tone it down a little and have gotten used to it.

I personally have no use for it. I use either an audio interface of which I own about 6 things that are classified as such, and all have good ASIO drivers. Or I simply use WASAPI shared. @Starship Krupa Ya, that was possibly it. I always leave that unchecked because otherwise I can't work in multiple apps or play a You tube. I have no need of WASAPI exclusive, I'm not recording. @Milton Sica My findings of a 9 ms latency is actually very good. Read what I said carefully. All systems have latency. A generic ASIO driver cannot be expected to calculate that. It needs to report the Round Trip latency to the Daw.