Craig Anderton

If a stereo file uses all the available headroom, bouncing it to mono will cause clipped peaks. You need to lower the stereo file's level prior to bouncing to mono. The image on the left is prior to bouncing to mono, the image on the right is after bouncing to mono.

1 hour ago, Sal Sorice said:

I had a bad stick of RAM wreak havoc on my system about a year ago.

This ^^^  And SSDs aren't perfect either, with some better than others. What you may be seeing is the dying gasps of a storage device. Things getting worse the more you use the system may also mean the issue is thermal-related. Have you dusted off the insider of your computer and cleaned dust off the fan blades lately? Are the fans still rotating? I know, stupid questions but you never know.

It seems like Superplate may be an issue...check out this thread. One of the replies said he's experienced problems with it if the latency is too high or too low.

I can also vouch for Jim Roseberry, he's one of the best. BTW don't forget that some plugins contribute massive amounts of latency. I'm not sure this applies to your situation, but you may encounter this issue in the future. If a project suddenly seems to "slow down," open it in safe mode and don't load any plugins. If the response improves, go through your plugins and find which ones cause the most problems.

Hey, no problem, always glad to help! I'm not sure if this is a thing anymore, but when I posted a thread about it, quite a few users said disabling strategic drivers made a major improvement. Others didn't experience any improvement...you're at the mercy of the drivers. But if it works, hey, it's free and non-invasive.

I can't find the original thread about this, but here's an excerpt from The Huge Book of Sonar Tips and Tricks:

Some graphics cards install a special audio driver, typically with a name like “High Definition” or
“HD.” This handles functions that are not needed to operate a DAW (like sending audio out to a
TV), and if poorly written, can impact audio interface latency and sometimes even a DAW’s
overall stability.

Disabling this driver can improve performance in some cases, depending on the driver. In fact
when this tip was first posted in the SONAR forum, several people confirmed disabling the type
of driver specified in this tip can indeed make for a major improvement, especially with ATI
cards. There’s no harm in disabling this driver, because if you run into problems, you can always
re-enable it.

To disable this driver:

1. Open the Windows Control Panel, then open Device Manager.
2. Unfold the Sound, Video, and Game Controllers section. You should see entries for the audio
and MIDI interface (if present) you use with your DAW.
3. Right-click on any audio driver that does not relate to your interface. From the context
menu, choose Disable. You can always Enable an entry later if needed.
4. Re-boot your computer for these changes to take effect.

However, sometimes these drivers do not have an option to disable, only uninstall. Some users
have reported successfully disabling these drivers with the following procedure.

1. Open the Windows Control Panel, then open Device Manager.
2. Open the System Devices section.
3. Temporarily disable anything labeled HD Audio Host Controller (or something similar). This
may make the driver in Sound, Video, and Game Controllers disappear, but we’re not done yet.
4. Re-boot the computer.
5. Open the Windows Control Panel, then open Device Manager.
6. Unfold the Sound, Video, and Game Controllers section. You should see entries for the audio
and MIDI interface (if present) you use with Cakewalk by BandLab.
7. Right-click on any audio driver that does not relate to your interface. The driver(s) may now
include the option to disable. If so, disable by right-clicking on the driver and choosing Disable.
If not, you will not be able to disable this driver easily, but please continue through step 10.
8. Open the System Devices section.
9. Re-enable the HD Audio Host Controller devices you disabled in step 3.
10. Re-boot the computer.
11. Open the Windows Control Panel, then open Device Manager.
12. Unfold the Sound, Video, and Game Controllers section. Verify that the High Definition
driver remains disabled.

This tip is presented “as-is,” so use it at your own risk. However, because you’re disabling
functions, you should be able to return to a previous state by re-enabling anything you had
disabled previously.

Regarding latency...I had extreme problems with one particular graphics card that installed an audio driver. The card was designed for gaming, so its audio driver took priority over everything else that had anything to do with audio. It also sent audio to TVs or some such silliness. Getting rid of the graphics card driver cut latency issues by half, IIRC.

Your system seems like it should be decent enough to do what you, but if you've had Windows update, who knows whether Microsoft was trying to be "helpful" and install new graphic drivers or whatever. I don't have to deal with this foolishness anymore because I bought a custom PC Audio Labs optimized for music, but if I were you I'd see if you could find a local PC integrator who knows about these things and can find that one weird little thing that's screwing up your system.

If you're using an older operating system, perhaps you're run into the Windows MIDI port limit issue?

2 hours ago, msmcleod said:

Mackie Control Universal  /  XT / C4
Presonus Faderport 16 
Yamaha 01X
Behringer BCF2000
Korg nanoKONTROL Studio
Korg nanoKONTROL 2

You're not alone, I've tried all of them with Sonar as well (except for the 01X) and they worked for me.

I don't foresee any specific seismic changes in controller compatibility until MIDI 2.0 becomes mainstream.

On 11/27/2023 at 4:36 PM, John Vere said:

They are managing this business in a business like manner which we should all support because this helps secure the future of the DAW we have chosen to stay with. 

+1. If you use something that makes your life easier, encourages productivity, and is fun to use, I think whoever produced that product should be rewarded in some way.

It wasn't an error, it was offering advice. Personally, I wouldn't buy a Gateway computer either - especially considering that if you go to their website, all the computers are listed as "available soon."

8 hours ago, msmcleod said:

When I've played around with this,  I've found the best results are when you combine the separated stems with the original, using a separated stem to boost/reinforce the original part. Keeping the separated stem at a fairly low level is enough to give that part a volume boost over the original without any artefacts becoming obvious.

That makes a lot of sense. I wonder if that's what iZotope is doing with their master rebalance feature. I've used it to emphasize vocals/drums/bass from vinyl cut in the 60s.

Have you tried processing the stems when you combine them with the original part?

I think it's only fair to give a definitive answer, so here it is: stay tuned

To put things in perspective, "you only get one chance to make a first impression." I'm sure the Bakers are doing everything they can to make sure the the new Sonar's coming-out party is done correctly from a technical, marketing, and pricing standpoint. It's not a trivial effort, to be sure. 

If you have problems with Addictive Drums, remember that if there are changes in your computer, it needs to be de-authorized from the "old" computer and re-authorized.

On 11/8/2023 at 10:57 PM, Byron Dickens said:

Yet nobody complains about Logic being Mac only....

In 2008, people complained like scorned paramours when the Windows version of Logic was killed (and all development ceased) after Apple acquired eMagic. To add insult to injury, the last version (5.5.1) had finally reached a level of stability that had been lacking in previous versions. Adding fuel to the anger was that Windows Logic users with a boatload of projects had no option other than switching to a Mac if they wanted to take advantage of future development. 

Believe me, the Windows users who felt unceremoniously dumped complained a lot about Logic being Mac-only.

Another fine point is that true peak readings are approximations. Different programs will give different results. They'll be close, but not exact.

1 hour ago, John Vere said:

Truth is I had been normalizing all my wave files to -0.2 db  using Wave Lab for a long time. This was in the CD and DAT days.  After I learned how “wrong” this was I took a close listen to those files and never really heard anything wrong?

I don't have a definitive answer, but if the material was heavily limited or maximized, then normalizing to -0.2 dB was basically normalizing a square wave. If two samples are on a straight horizontal line, then there isn't the potential to create something that arcs over 0.0 upon reconstruction. The main reason a lot of people normalized to a value under 0 with CDs was because if there was a certain number of successive samples that hit zero, CD manufacturers assumed it resulted from clipping, and would reject the master. Normalizing to -0.1 wouldn't let that happen.

Transcoding to MP3 is different, because of the data omission process. It's tough to describe the distortion, except I'd say it's fuzzy instead of spikey, and very low level.

1 hour ago, John Vere said:

I can't hear any distortion so possibly it depends on the convertor and the sample /bit rates.

I'm glad you brought this up! Having transcoding distortion from too high peaks is not like the distortion you're used to hearing. What made me super aware of this was when I was mixing down a song that was going to be transcoded with AAC for YouTube. There was this strange kind of very low-level "background fuzz" that made me think perhaps there were undersampled instruments, artifacts from not oversampling amp sims, that sort of thing. I spent way too much time trying to isolate the tracks that were causing the "problems." But the problem was that I had not paid enough attention to leaving sufficient True Peak headroom. Once I did, the weird background fuzz went away completely.

I have only two things to add to the excellent advice in the previous two posts. Regarding True Peak, for material with an LUFS reading above -14.0, I'd recommend -2.0 for the True Peak max to avoid distortion when transcoding to MP3.

On a related subject, I use Loudness metering to balance preset levels in amp sims and synths/samplers. This doesn't matter so much for recording, but it puts you in the ballpark for live performance.

38 minutes ago, ptheisen said:

Just to add to this, other types of plugins can also be affected by gain changes, such as amp and console emulators, those that intentionally add distortion. Usually for those types of plugins, the more input gain they receive, the more distortion they add to the dry audio.

Yes, and one of the cool aspects is you can ramp up the level over a passage to increase the amount of drive and add intensity.

42 minutes ago, Glenn Stanton said:

as a note: 32-bit float cannot be "digitally" clipped - true, but you can definitely overload the ADC devices prior to that... so the idea that people won't get distortion isn't true

You realize you have no future in marketing, right? 🤣

I know this thread is nominally about bit depth and not sample rates, but since sample rates keep getting mentioned, y'all might enjoy this article that busts 12 sample rate myths. It even includes audio examples for those who wonder if you really can hear a difference.

I'm guessing that once Sonar becomes a product that costs money, it will be perceived as something that's here to stay. I'm sure that between the experiences with Roland, Gibson, and then a free version, there were some doubts about longevity

As far as getting press is concerned, althoughMusicTech (which is part of the group of companies that includes BandLab) and Sound on Sound aren't interested in Sonar articles, Sweetwater Publishing is open to making a Sonar tips book part of their catalog. They were even willing to do that before the new Sonar was announced, so clearly their motive wasn't to sell copies of Cakewalk by BandLab   At this point I'm just waiting for "New Sonar" to come out so the screenshots and feature set will be current. I think once a book is out, it will (hopefully) contribute at least something to the program's credibility.

I switched from 44.1 to 48 kHz recently not because I wanted to or felt it would improve the sound, but because Dolby Atmos sessions need to be at 48 or 96 kHz. Rather than switch back and forth, I figured I'd cave to the gods of "Immersive Is Really Going to Get Traction This Time, Really It Will, It Won't Be Like Every Other Surround Fail Since the 70s." We'll see about that.  

So now I do everything at 48 kHz. I guess that means seamless transfers to DATs, LOL.

But to get back to something that may be significant...the difference may not be as much about the medium, but the workflow. Tape required a certain way of working that's different from working with a DAW.

One person said he liked tape because rewinding gave him a chance to gather his thoughts before doing another take. I said "why don't you just wait a few seconds after hitting stop before hitting record?" He said it wasn't the same thing. As silly as that sounds on the surface, I understand there's a difference between a partnership with the tape where you're dependent on it doing something, versus just sitting around. There were many other workflow differences...with limited track counts, you couldn't do premixing...punching was more common than comping...you could tweak the EQ and bias for a particular sound...that sort of thing.

However, also consider that in big studios with 24-track 2", there was often a degree of camaraderie with the engineers, musicians working in other parts of the studio, etc. The social aspect may have had more influence on the music than the tape's distortion.

1 hour ago, John Vere said:

This has turned into a Gear Space thread.  

Sorry, but it doesn't exceed that threshold until actual name-calling and fistfights break out.

...and let's not forget rubber pinch roller and belt deterioration, azimuth alignment, head lapping, demagnetization, flaking oxide over time, having to redo bias and EQ adjustment every time you changed to a different reel of tape, and those hellishly expensive alignment tapes. And of course, replacing the capstan motor when you took variable speed down too far...not that I ever did that at Record Plant...it wasn't me. Really.