Jim Roseberry

The reason for the "audible glitches": The CPU is overheating... and the machine throttles down the speed (to avoid burning up the CPU). The reduced CPU speed causes the audible glitches.

On a modern build, Hyper-Threading should be enabled.

With off-the-shelf systems, the BIOS often doesn't expose all possible parameters. This is to keep less savvy users from fouling up their computers. The downside is that you don't have access to these parameters.

I've raved about the Waves Scheps Omni Channel before... but just had to do so again. I've been working on some vocal tracks for a client's mix... where there are many plosives. Usually, I can isolate the plosive (as a separate clip) and use a High-Pass filter to remove it. In a few cases, the plosive is still objectionable (even after using a high-pass filter). I'd been using the Scheps Omni Channel's De-Esser a lot lately... to solve numerous issues (guitar string squeaks, standard de-essing, taming cymbals, etc). So... I figured I'd give the Omni Channel De-Esser (there's actually two of them) a shot at eliminating these problematic plosives. I set the frequency at 325Hz and engaged the first De-Esser. Problem solved. It was that easy. Ironically, I haven't been using the Sheps Omni Channel for standard mixing duty... but for specific problems it's been indispensable. The two "De-Essers" are capable of *far* more than standard de-essing. If you're dealing with similar issues, check it out.

FWIW, The issue is the chipset on the Firewire controller. Over the past 25-30 years, I've seen many similar cases... and in almost all... the Firewire controller wasn't using a TI chipset. With a desktop, the solution would be to swap the Firewire controller for one that has a Texas Instruments chipset. With a laptop, you live with what it is (can't be swapped out). The issue has nothing to do with the MOTU or Sonar.

Hi Pete, The latest Win10 install discs are version 1903. Installing on socket 1151/Z390 based machines (numerous different motherboards), with all available patches (thru Windows Update procedure), high DPC Latency persists. Installing from an older Win10 disc, there are no DPC Latency issues with v1803 or v1809. I've downloaded the KB4505903 fix from your link above... and will test to see if that resolves the issue. We appreciate your presence here on the forums!

Hi Noel, I have not filed a feedback hub report. I'll do so ASAP.

FWIW, Don't confuse DPC Latency with Audio Latency (two completely different things). You need low/consistent DPC Latency to effectively work at low audio latency settings.

All popular DAW applications feature Automatic Plugin Delay Compensation (Automatic PDC). If a latent plugin is inserted *anywhere* in a project, all other audio is delayed by that amount (to maintain sample-accurate sync). There are two ways to work around the issue: While tracking, avoid latent plugins CbB (Sonar) has a global Automatic PDC bypass feature (enable this while tracking) Latency has but two sources (unless the DAW application itself uses additional buffering): Audio Interface Latent Plugins Some DAW applications use extra buffering (which adds additional latency). Ableton Live uses additional buffering (which can't be disabled). Cubase has "ASIO Guard" (which can be disabled) StudioOne has "Dropout Protection" (allows setting the additional buffer size from 128-samples up to 2048-samples - the green Z button bypasses the extra buffer)

Though not directly related to your issue, Win10 build 1903 has been problematic on many current build machines. High DPC Latency...

If you have a Yamaha keyboard (Montage, ModX), it can function as both a MIDI controller and audio interface. Yamaha owns Steinberg... and users their audio interface/driver technology. When you load the drivers for your Yamaha keyboard, the "Yamaha Steinberg USB Driver is installed". FWIW, I've seen a similar issue when a Windows update was (first) released. In the short-term, had to roll-back to the previous Windows build (everything then worked fine). Long-term, the issue was resolved.

Any meta-data embedded with the kick sample? Does it happen with any kick sample... or just a particular sample/library? If you're talking loops, they're being time stretched/compressed to match the current project's tempo.

I'd try deleting the Aud.ini file (stores information about audio settings)... then re-open CbB and see if the issue persists.

If it were TI, it would mention it in Device Manager.

Yeah, it's most likely a Via or Ricoh chipset Firewire controller.

In the meantime, if you go to Preferences>Audio Driver>ASIO Panel... and change the ASIO buffer size (to another value and then back to what it was), this may clear the distortion without having to close CbB (Sonar).

Are you using a TI chipset (Texas Instrument) Firewire controller? In almost every case regarding a problem with a Firewire audio interface, the end-user isn't using a TI chipset Firewire controller.

Effects count - processor speed is the largest factor Track count - disk speed is the largest factor 1. Faster bouncing/freezing of tracks would be mostly down to CPU speed. 2. Faster loading of projects would be affected by disk speed and CPU speed. If you're dealing with large sample-libraries (that load particularly slow), you can put those on a M.2 Ultra SSD (sustain ~3500Mb/Sec). 3. Ability to run more tracks and plugins (as you can probably guess) is down to faster disk-speed and faster CPU. As a point of reference: Conventional HD sustains ~200Mb/Sec SATA SSD sustains ~540MB/Sec M.2 Ultra SSD (using 4 PCIe lanes) sustains ~3500Mb/Sec When it comes to CPU speed: Not all processes in a DAW can be multi-threaded. Playing/monitoring in realtime thru an AmpSim plugin at 96k using a 32-sample ASIO buffer size... isn't something that lends itself to being heavily multi-threaded Some plugins/instruments like UVI Falcon only use a single core This is why clock-speed is the single most critical factor when choosing a CPU for DAW purposes. Note that CPU core performance doesn't scale 1:1 IOW, Doubling the number of CPU cores doesn't double performance. Having more cores is beneficial... but not at the expense of significant clock-speed. This is why Xeon CPUs are typically a poor choice for DAW purposes. They have more cores... but often significantly slower clock-speed (resulting in significant performance hit). In a perfect scenario, you want highest available clock-speed... and the most cores you can get.

It's one simple thing that can make a HUGE difference in the clarity of your mixes.

One use of EQ before compression is to use a high-pass filter (to roll out unwanted low frequencies). Doing this prior to compression can ensure the compressor doesn't respond to those (unwanted) low-end frequencies.

Unfortunately, Steinberg installs the Generic Low Latency ASIO driver (by default). This driver can bump your desired ASIO driver out of position. IOW, If the proper RME ASIO driver was selected in CbB, it may now be "bumped" out of place (leaving the Generic ASIO driver as the selected driver). In the locations scook mentioned above, delete the Generic ASIO driver entry: HKEY_LOCAL_MACHINE\SOFTWARE\ASIO HKEY_LOCAL_MACHINE\SOFTWARE\WOW6432Node\ASIO Once deleted, no application can see/select the Generic ASIO driver. This works for ANY ASIO driver you don't want to use. ie: Many hardware guitar processors, drum modules, and synths now function as an ASIO audio interface. If you know you never want to use your Helix, AxeFX, GT-1000, TD-50, Montage, etc... as an audio interface, this is a quick/easy solution.

Latency has but two sources: Audio Interface Latent Plugins If sounds like you're trying to monitor via software (vs. the audio interface's onboard hardware based mixing which is close to zero latency). If that's the case, you're dealing with round-trip latency. Round-trip latency is the sum of the following: ASIO input buffer ASIO output buffer The driver's (often hidden) "Safety" (sometimes called "Streaming") buffer Latency of the A/D and D/A converters To reduce Round-Trip Latency: Set your ASIO buffer size as small as possible If the audio interface allows you to change the Safety Buffer size, set it as small as possible You can also reduce round-trip latency by using higher sample-rates Note that reducing Round-Trip Latency comes are the expense of higher CPU use. Regarding Latent Plugins: All popular DAW applications feature Automatic Plugin Delay Compensation. If you have a latent plugin inserted ANYWHERE in the project, Automatic Plugin Delay Compensation (Automatic PDC) delays all other audio to maintain sample-accurate sync. To work around this issue: Avoid latent plugins while tracking Some DAW applications feature a global PDC bypass feature (enable this when tracking)

You can remove short spots of digital and analog clipping. What you can't do is take a constantly/heavily saturated recording (like distorted guitar) and remove it. Samplitude Pro X Suite comes with declipping capabilities. There's also Izotope Rx.

My advice... If you want a smoother ride, stay on the backside of the update wave. If you ride the crest, sometimes you're gonna wipe-out. (cue that tom intro)

When playing virtual-instruments, you're dealing with one-way (Playback) Latency. Playback latency is (roughly) half the total round-trip latency. Certainly workable (decent) with the Apollo... The RME Fireface UFX+ can achieve lower Round-Trip and Playback Latency. The Presonus Quantum can achieve even lower Round-Trip and Playback Latency than the UFX+. If lowest possible latency when using 3rd-party EFX/Instruments is a paramount concern, Apollo isn't the interface you want. Apollo's forte' is fidelity... and being able to run UAD plugins at ~2ms round-trip latency.