Jim Roseberry

Another thing to be aware of...

Some older USB-2 audio and MIDI interfaces don't work well with USB-3 controllers.

Usually, the issue is with 3rd-party (not Intel) USB-3 controllers.

Presonus Quantum can get down to ~1ms total round-trip latency.

UFX+ can get down to ~2m total round-trip latency.

I was running an Apollo-8 (Quad) and Quad Satellite expander (both via Thunderbolt) for a good while.

Fidelity is excellent (super low noise-floor).

Stability was excellent.

My only gripe with the UA audio interfaces is that (due to their onboard DSP), round-trip latency is a little higher than something like a UFX+ or Quantum.

Of course, if you're wanting to run UAD plugins, you can play/monitor in realtime thru UAD plugins with 2ms round-trip latency (via their Unison tech).

The lowest round-trip latency you'll achieve with the Apollo series (and Arrow) is ~3.7ms.

If you go Thunderbolt, you've got to make sure all the details have been tended-to.

The combination of Asus Thunderbolt EX3 and the Apple Thunderbolt-3 to Thunderbolt-2 adapter has worked extremely well in all scenarios we've tested.  This includes many different desktop builds, laptops, etc...

Though Presonus recommends the StarTech Thunderbolt-3 to Thunderbolt-2 adapter, the Apple adapter works perfectly fine with Quantum.

5 hours ago, Starship Krupa said:

I'd prefer mine to be non-destructive.

In this day and age, Normalizing should absolutely be non-destructive.

It would make sense to tie this into that Per-Clip "Static Gain" parameter I've been lobbying for.   😁

There are many cases (when mixing), where I'll use static gain changes (per clip) to even out a vocal performance.

Yes, Clip Envelopes work... but it's slow (compared to a Static Gain/Normalize parameter).

As a test, what happens if you switch to the onboard audio?

Does the issue with crashing persist?

The reason I ask is the HD500x drivers are getting a bit long-in-the-tooth.

First thing I'd do is rule it in/out as a culprit.

For a period of time, I was using a Zen Tour audio interface (which doesn't provide a WDM stereo port for Windows Audio).

This small application can route Windows Audio to one of your audio interface's ASIO outputs.

https://www.vb-audio.com/Voicemeeter/index.htm 

If you're wanting to "connect" various applications, this may be what you're looking for:

https://www.vb-audio.com/Cable/index.htm

That was excellent all around.

Both performance and recording...

Funny enough, I kinda like this version better than the original (though I'm not a huge Chicago fan).

2 hours ago, henkejs said:

My Windows 10 Home PC says 1903 is available: "When you're ready for the update, select "Download and install now."" Can I put this off indefinitely?

If you keep your machine disconnected from the Internet.  😉

With the Pro version of Win10, you can completely shut down all Automatic Updates (including notifications).

Unfortunately, you can't do this in the Home version.

There is an "in-place" upgrade from Home to Pro... but MS gouges a bit on the price ($100).

There's no performance advantage to Pro, but we recommend it to clients because it allows more control to "rein-in" Win10.

Automatic Updates can be fully shut down... and the Group Policy Editor makes it easy to shut down Cortana, OneDrive, etc.

I've got the Montage version of SampleRobot.

Haven't had the time to use it.

Yeah, I'm sure you *can* do it... but it will be an elaborate process to get all articulations captured/programmed... with custom script/s controlling the switching.

That makes my tendons hurt just thinking about it.   😁  LOL  

IIRC, Waves Central doesn't ping the Registry for your preferred/default VST Plugins folder location.

Every time I reinstall my Waves plugins, I have to manually move the WaveShell dlls.

That said, I wouldn't want to be without their SSL Bundle, H EQ, or Scheps Omni Channel.

If you ever need a de-esser to solve problems (string squeaks, sibilance, excessive cymbals), the pair in Scheps Omni Channel are amazing.

On 6/27/2019 at 1:09 AM, Max Arwood said:

Yes. You can make nice samples libraries with either of these. I like samplit because of its ease of use (and lower price) I have been trying to sample my vl70m-turbo. Neither program is really getting it right. Samplit set up quicker.  Sample robot is more tweakable. Msmcleod made me buy sample robot (just kidding lol!!). Still not happy with the sample set from this device  with either sampler.  Matt Traumam At Patchman is the king of this Stuff. You should listen to some of his Kontakt samples.  Wind pack 1 is great!

Hey Max,

Isn't the vl70m a physical-modeling synth?

If so, it's going to be really hard to sample that... so it accurately reproduces all nuances.

If you're working with straight up audio production (especially with today's delivery often being "digital"), it doesn't matter nearly as much as 20 years ago.

Sample-rate conversion is significantly better than it was all those years ago.

If your'e working with video, it's more convenient to stick with 48k.

The latest project I'm mixing was done at 96k.

The previous project I was mixing was at 44.1k.

While I do believe there are audible advantages to using high sample-rates, there are many other factors that have larger impact on  final quality.

• Front-end gear
• Mic placement
• Engineer's experience
• Quality of instruments
• Caliber of the player (a great player will sound good on nearly anything)
• Song arrangement
• Etc

Make sure you have a backup if you're going to install v1903.

Even with the latest patches, we've seen DPC Latency issues with current generation hardware.

54 minutes ago, azslow3 said:

One of the tricks I was not aware until recently: https://forums.guru3d.com/threads/windows-line-based-vs-message-signaled-based-interrupts.378044/

If you don't know what you're doing, be careful when switching!

Some devices don't work (at all) with MSI... and can cause your machine to BSOD.

Other devices (like RME audio interfaces) will appear fine... until you try to use it (and realize it's not actually working).  

Video cards and most of the core hardware work fine with MSI

Audio interfaces (dedicated and onboard) and Firewire controllers often don't work with MSI.

41 minutes ago, azslow3 said:

As I wrote, if I start Latency Monitor without "preparation", the picture is bad. But I can easily keep the buffer 64 without any glitches when I need that (RME+REAPER).

The reason you see this is due to CPU throttling.  

When the machine is under heavier load, DPC Latency is lower... because clock-speed is higher.

If the machine has major issues with DPC Latency, putting it under heavy load won't solve the issue.

Power-management necessary to achieve long battery-life is diametrically opposed to configuration for maximum performance.

That's the heart of the issue.

If you look at any of the top performing custom built laptops, notice that they're *all* using Clevo shells.

They're expensive, bulky, and have extremely short battery-life.

Physics... of such a tight space make it a complicated balance.

If you're strictly talking off-the-shelf laptops, Apple (OSX on the MBPs) does a better job juggling the above than any off-the-shelf PC laptop.

Custom Clevo's smoke the MBPs... but that's another topic.

Is everything powered from a single outlet?

If not, I'd try that before unplugging all audio.

At the risk of repeating:

If the machine suffers from high DPC Latency, there is no working around it.

It has to be solved at the source.

No audio interface, no setting core affinity, etc.

Read the details here:

https://docs.microsoft.com/en-us/windows-hardware/drivers/kernel/introduction-to-threaded-dpcs

With many cases... especially when it's ACPI (power-management) related, there is no solution.

The only solution is to replace said hardware (in the case of a Laptop, NUC, All-In-One) isn't possible.

Getting the personalities "right"... and keeping them all focused on a single direction is the hardest part.

I've been really lucky with bandmates the past 20+ years.

I've played in probably half a dozen bands in that time... and I'm still friends with most them.

It's not easy to leave good friends behind... but to move forward, sometimes that's necessary.

We always try to make it about "the song", "appealing to the audience", trying to strengthen our performance, and trying to increase our draw.

That usually keeps things going in the right direction.

I've been adamant on keeping the band a 4-piece.  Any more than that... and it gets complicated with schedules, etc.

We always hire commercial sound.

Crossing that threshold made the experience (to me) a lot more enjoyable.

We're a cover band in our 30s-50s... so the "rock-star" dream has sailed.

We play for fun... and to cross off "bucket-list" items.

The band is paid well... but we're not trying to live off playing.

Behringer has absorbed numerous great companies over the past 10+ years (Midas, Kark Teknik, TC Electronic).

Some of their latest gear is very good.

If you're a fan of the original 1176 and LA-2A compressors, you should check out the 76-KT and KT-2A.

They're inexpensive... and sound excellent.

The 76-KT works wonders on drums.  

ACPI.sys is power-management related... and DPC Latency issues with that culprit often can't be resolved.

Unfortunately, it's not as simple as assigning your DAW software to "unaffected" cores.

https://docs.microsoft.com/en-us/windows-hardware/drivers/kernel/introduction-to-threaded-dpcs

Laptops that are particularly good for DAW purposes are expensive, large/bulky, and have short battery-life.

IOW, The general-purpose user would hate it.  😉

There is no work-around for extremely high DPC Latency.

It has to be addressed at the source.

FWIW, Disabling TurboBoost won't solve DPC Latency issues.

If you've updated the motherboard's firmware, updated all drivers, and updated Win10... you're out of options.

When dealing with a general-purpose machine, you have to keep the target end-user in mind.

The typical user of that laptop would never know it suffers from a several millisecond "hiccup" in data flow.

For someone wanting to use it for DAW purposes, that small hiccup in data flow results in an audio glitch or dropout.

Lets say you want to run your audio interface at a 64-sample ASIO buffer size (44.1k).

The machine has 1.5ms to process/fill the next ASIO buffer. 

If a driver is monopolizing the CPU, the next buffer won't be processed/filled in time (resulting in a glitch/dropout).

The typical user cares far more about extended battery life than ultimate performance.

The power-management necessary for extended battery-life is diametrically-opposed to what we want in a high-performance DAW.

With Laptops, power-management is the source of many DPC Latency issues.

+1 on the importance of proper backup!

It's never a problem... until it's a major problem.  😉

FWIW, Your DAW's CPU will not affect the sound.