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Audio dropouts making Cakewalk unusable.


kevmsmith81

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 As I said, it may well be due to some incidental config change. But there can also be changes in the environment that affect a feature of Cakewalk that other DAWs don't have like plugin load balancing/upsampling, 64-bit double-precision engine , etc.

Or this, which I believe is not enabled by default; possibly you did this...?:

Experimental aggressive task scheduling model

We continue to work on improving our multi-threaded engine performance and have added a new "aggressive" task scheduling model. The aggressive task scheduler utilizes more efficient task management that can result in better multi-processing and fewer thread context switches. This feature is still experimental so report back if you notice any improvements or problems when using it.

The new scheduling model is activated by setting the ThreadSchedulingModel value to 3. This can be done via Edit > Preferences > Audio - Configuration File. 

 

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Cakewalk is consistently the worst at handling audio data flow amongst all my DAWs.  It clicks and pops and overloads consistently on mid-loaded projects that do just fine in Samplitude, Sequoia, Reaper, ProTools, FL Studio and Nuendo.  To have to adjust buffers over 128 asks for intolerable latency with soft-synths and direct monitoring.  I'm not saying that this isn't something in this particular case that will be figured out, but I am saying that Cakewalk and Sonar have always been the worst at handling audio flow.  I've been with Cakewalk since the eighties.

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17 minutes ago, Jon White said:

Cakewalk is consistently the worst at handling audio data flow amongst all my DAWs.  It clicks and pops and overloads consistently on mid-loaded projects that do just fine in Samplitude, Sequoia, Reaper, ProTools, FL Studio and Nuendo.  To have to adjust buffers over 128 asks for intolerable latency with soft-synths and direct monitoring.  I'm not saying that this isn't something in this particular case that will be figured out, but I am saying that Cakewalk and Sonar have always been the worst at handling audio flow.  I've been with Cakewalk since the eighties.

I'm just glad to have an alternative I can go to. I will try the suggestions people have provided when I'm at my computer but it is frustrating to deal with this issue. 

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13 hours ago, kevmsmith81 said:

EDIT:  Just for reference, my machine is a laptop running Windows 10 with a Ryzen 5 2500U CPU and 16GB RAM

Try checking that your Windows power scheme is set to "Performance".

That is a mobile processor and the default is probably balanced or something like that. If so that would be trying to conserve power at the expense of performance.

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4 minutes ago, abacab said:

Try checking that your Windows power scheme is set to "Performance".

That is a mobile processor and the default is probably balanced or something like that. If so that would be trying to conserve power at the expense of performance.

Thanks for tip - this is the first thing I do every time I power on a new laptop for the first time, as I always have it plugged in anyway. :)

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I wonder what you guys have in common.

I presume all of you are passing in Latency Monitor 

Lots of us are working fine. I'm running with at 3.1 to 3.8 millisecond round trip without issues.

I can't help wondering if we don't need to get back to basics and do the traditional trouble shooting.

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I have an older I7 but still - I am having to freeze everything.   My latency set at  2048.  I have recently moved to Win10.  I fell like I lost some processing power in the upgrade.  I have turned off everything I could think of.   I wish we could have an "official" statement of tweaks. I wish it would have  a few levels - These will help, These will help more, and - If you are desperate for horsepower try these.

Also - we have had tons of great  bug fixes - but now it might be time to put some resources toward performance.

I can hardly play any projects with ThreadSchedulingModel value set to 3.

Thanks,

Max Arwood

 

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LatencyMon after 40 minutes of web browsing with ~30 tabs open in Chrome -  including watching Geoff's video about it -  while Cakewalk ran at 30-40% load in the background, looping the old 'These Arms' demo, which is pretty 'heavy', at 64 samples. Cakewalk Perf Monitor reported 1 buffer dropped... when I stopped playback. DPC latency typically runs under 100us on this machine if I disable WIFI. Off-the-shelf Dell i7 6700k with power management and CPU throttling disabled:

125169015_LatencyMonat40MinutesofBrowsingandLoopingTheseArmsat64Samples.thumb.png.82a6648affd67bb12eba815f69cd0ece.png

 

Edited by David Baay
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2 hours ago, Max Arwood said:

..... I wish we could have an "official" statement of tweaks. I wish it would have  a few levels - These will help, These will help more, and - If you are desperate for horsepower try these.....

Max Arwood

 

Great idea.  I'm not a deep geek when it comes to my PC so I don't feel confident going under the hood, even though I'm tempted when I see posts on forums setting out various tweaks.  For example here's one on the Studio One forum which tempted me but I left things alone in case I upset something by accident:  https://forums.presonus.com/viewtopic.php?f=151&t=37827

Noel, about Max's suggestion?

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What should the Recording Timing Master and Playback Timing Master be set to? I have Rec set to ASIO input 1 and PB set to ASIO Output 1. I use a Focusrite Scarlett 18i20 and Octopre. The 18i20 and Octopre are connected via ADAT. The Octopre is the master and the 18i20 is clocked to the Octopre. So, I'm wondering if my Rec and PB Timing need to be set to the ADAT inputs and outputs? Or, does any of this make any difference at all?    

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8 hours ago, Rod L. Short said:

The Octopre is the master and the 18i20 is clocked to the Octopre. So, I'm wondering if my Rec and PB Timing need to be set to the ADAT inputs and outputs? Or, does any of this make any difference at all? 

The Octopre is a standalone pre-amp with no direct connection to the DAW, right? In that case you can only set Timing Masters to channels of the slaved 18i20. That's how I run my interface when slaving to an Alesis QSR  (because the Alesis didn't like being a slave) . Works fine. The only limitation is that Cakewalk can't force a clock rate change when you load a project that includes audio recorded at a different clock rate.

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1 hour ago, kevmsmith81 said:

I have now done the test.   Apparently my system should be suitable for handling real time audio and other tasks without dropouts.

How 'suitable'? IIRC, Latencymon labels anything under 1000-1500us (1-1.5ms) as 'suitable', though that's not ideal for a DAW that's running a buffer on the same order of magnitude. That said, moderately-high-but-stable (say 400-500) is better than usually-low-but-spiking (150 with spikes to 1000). And you need to let it run for a while to see if anything is causing spikes.

Did you try the Config File  reset?

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I totally understand the advice about system tuning being given, as that is fundamental to good audio performance, and essential to any DAW.

But I am a bit cautious to rush to blame the hardware in this case, because the OP stated in the first post that it was only happening with Cakewalk, and not the other DAW he tested with.

Could something actually have changed in Cakewalk recently to affect performance? Just asking for a friend, so please don't flame me ... ;)

 

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For my part, there is no rushing to judgement. I'm just suggesting logical troubleshooting steps, including making changes to the Cakewalk configuration.

The current release is working too well for too many users to conclude that the update has an inherent issue that's independent of the environment it's running in or the configuration of Cakewalk or project files. 

Of course the ultimate troubleshooting step would be to roll back the update, but that isn't possible with the current installation model unless you've prepared in advance by backing up or using scook's utility for managing versions.

I believe there were some changes to WASAPI driver mode related to sharing drivers, but the OP is using ASIO.

I do see more dropouts on my laptop, but I attribute that to using WASAPI shared mode on a low-spec machine with years of accumulated software bloat (dual-core company Lenovo). And it's no worse in this release than it's ever been. It's fine if I only use it for playback with the buffer at 10-20ms, but down at 3ms where you would want it for real-time performance of soft synths or input monitoring , it's dropout city. But I can see the DPC spikes and CPU hits from other processes that cause it, and if I go out of my way to do a clean boot ith WIFI and Bluetooth disabled, and switch to WASAPI Exclusive mode, it's quite stable at 3ms.

This reminds me... there was a problem with WASAPI driver mode at 44.1 kHz in a release late last year that I never encountered because I run at 48kHz. There's no indication that anything like that has resurfaced, but it might be worth changing sample rates (in a project with no recorded audio) as another general troubleshooting step, regardless of what driver mode you use.

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