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Old 05-26-2017, 04:27 AM   #1
marcoctn
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Default studio one 3.5 low latency feature

Looks interesting.
...but.... why did I never needed something like this in reaper?
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Old 05-26-2017, 04:52 AM   #2
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I wonder if it'd make my interface work @ 64 samples. It crackles even when nothing's happening, gotta try that demo.
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Old 05-26-2017, 05:44 AM   #3
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Looks interesting.
...but.... why did I never needed something like this in reaper?
Reaper always had the ability to be set to a lower latency than S1 and keep a stable system. By a significant margin too.

So, they're catching up? Is there some feature besides the I/O buffer control ('block size' in Reaper)? What could that even be?
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Old 05-26-2017, 06:11 AM   #4
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It seems like they've tried to fix their audio engine, which is known for falling apart at low latencies.
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Old 05-26-2017, 10:07 AM   #5
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Reaper always had the ability to be set to a lower latency than S1 and keep a stable system. By a significant margin too.
True REAPER has much been better in this regard.

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Is there some feature besides the I/O buffer control ('block size' in Reaper)? What could that even be?
I think the revolutionary aspect is the ability to set each individual track to low latency monitoring and buffer size (as needed), while keep the rest of the overall project & tracks at a much higher (and much less CPU intensive) buffer size. Set project buffer size @ say 2048 for playback and then set individual tracks that need live input monitoring (realtime synths, amp sims, etc) @ say 128, 64, 32 or even 16 samples, as low as your interface can go! Wow! No other DAW that I know of has this capability.

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Old 05-26-2017, 10:30 AM   #6
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I think the revolutionary aspect is the ability to set each individual track to low latency monitoring and buffer size (as needed), while keep the rest of the overall project & tracks at a much higher (and much less CPU intensive) buffer size. Set project buffer size @ say 2048 for playback and then set individual tracks that need live input monitoring (realtime synths, amp sims, etc) @ say 128, 64, 32 or even 16 samples, as low as your interface can go! Wow! No other DAW that I know of has this capability.
If it works like that, then makes me think of tone direct monitoring from Line 6 interfaces - you can software monitor with low latency while your ASIO is set to much bigger buffer size. Line6 does it with separate standalone app but I suppose if they can do it in software (I don't believe there's anything special inside a Line6 interface's hardware) someone could integrate such system into a DAW.
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Old 05-26-2017, 10:41 AM   #7
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No other DAW that I know of has this capability.
I thought Samplitude had that since ages. Called Hybrid Engine or so. But not sure. They were a bit messy with the explanations how it works back then.

I believe Reaper is essentially using the same trick, called here anticipative FX and render-ahead.

In the end they can't reinvent the wheel.
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Old 05-26-2017, 10:49 AM   #8
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.....I suppose if they can do it in software (I don't believe there's anything special inside a Line6 interface's hardware) someone could integrate such system into a DAW.
My question would be, since low latency monitoring is such an integral aspect of daily DAW use, why has this not been coded into every DAW app already. We obsess over getting our DAW's to perform at the lowest crackle free latencies with fully loaded projects trying to live input monitor softsynths/amp sims in realtime. Now Presonus pretty much makes that irrelevant. Keep your project at high latency (512 and above at all times) and only reduce the buffer size for individual tracks as needed when doing live IM. CPU usage never becomes an issue. I recently purchased S1 3 and have not really delved too much into it. But this innovative feature has me wanting to take a deep dive into v3.5 and further explore the ARA capabilities (I have both Melodyne & Vocalign Project 3).

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Old 05-26-2017, 10:53 AM   #9
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I thought Samplitude had that since ages. Called Hybrid Engine or so. But not sure. They were a bit messy with the explanations how it works back then.

I believe Reaper is essentially using the same trick, called here anticipative FX and render-ahead.

In the end they can't reinvent the wheel.
Yes, Samplitude, Reaper, Logic and Cubase. Now Studio One, shout out to DP for their background rendering thing too.

Studio One having this is the opposite of revolutionary, it's copying something that other DAWs had 10 years ago.
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Old 05-26-2017, 10:56 AM   #10
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I believe Reaper is essentially using the same trick, called here anticipative FX and render-ahead.
Unless I totally missed something in REAPER, in regards to Anticipative FX processing, I don't believe it is the same thing. IOW, can I set the REAPER project buffer size to say 2048 and then select two individual tracks for live input monitoring at 32 samples?
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Old 05-26-2017, 10:59 AM   #11
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You're getting mixed up between the audio interface buffer size and any other buffers the software wants to use on top of that. Reaper measured the antFX and media buffer settings in milliseconds, but what you effectively have is a playback buffer of thousands of samples and a live record buffer of whatever your interface is set at.
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Old 05-26-2017, 11:16 AM   #12
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You're getting mixed up between the audio interface buffer size and any other buffers the software wants to use on top of that. Reaper measured the antFX and media buffer settings in milliseconds, but what you effectively have is a playback buffer of thousands of samples and a live record buffer of whatever your interface is set at.
Okay you have my attention. I am using an ASIO audio interface capable of 32-2048 samples) So if I want to keep my overall project buffer size with loads of tracks & FX, at say 2048 (to keep CPU usage low) and only want to live input monitor several tracks @ 32 samples to maintain a realtime feel when overdubbing, where do I select the buffer size for my IM tracks? Maybe that is where I am confused? It seems pretty easy to do in S1 3.5.

Thanks for your help
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Old 05-26-2017, 11:30 AM   #13
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Okay you have my attention. I am using an ASIO audio interface capable of 32-2048 samples) So if I want to keep my overall project buffer size with loads of tracks & FX, at say 2048 (to keep CPU usage low) and only want to live input monitor several tracks @ 32 samples to maintain a realtime feel when overdubbing, where do I select the buffer size for my IM tracks? Maybe that is where I am confused? It seems pretty easy to do in S1 3.5.

Thanks for your help
Reaper does it automatically.

In Reaper you set your audio interface buffer size to what you want it at for live monitoring/recording. If you have Anticipative FX and Media Buffering enabled (they enabled by default), everything else is running with an effective playback buffer of thousands of samples. Rec-arming a track automatically bypasses those additional buffer settings.
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Old 05-26-2017, 11:46 AM   #14
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Reaper does it automatically.

In Reaper you set your audio interface buffer size to what you want it at for live monitoring/recording. If you have Anticipative FX and Media Buffering enabled (they enabled by default), everything else is running with an effective playback buffer of thousands of samples. Rec-arming a track automatically bypasses those additional buffer settings.
Okay, so as long as I have Anticipative FX & Media Buffering enabled and my audio interface set at low ASIO latency (64 buffers or below), I only need to rec/IM arm my tracks to get low latency and low CPU on playback? Thus this setup effectively has REAPER already running at thousands of samples on playback. Where do I see/set the actual playback latency (1000's of samples)in REAPER? In S1 3.5, this playback latency is adjustable to further minimize CPU usage and or snap, crackle, pop during playback.
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Old 05-26-2017, 11:55 AM   #15
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You go to Preferences->Audio->Buffering to get to the Media Buffering and Anticipative FX settings.

You can also adjust a bunch of other stuff there, "Allow live FX multiprocessing on: X CPUs" is an interesting one.
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Old 05-26-2017, 12:01 PM   #16
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You go to Preferences->Audio->Buffering to get to the Media Buffering and Anticipative FX settings.

You can also adjust a bunch of other stuff there, "Allow live FX multiprocessing on: X CPUs" is an interesting one.
Thanks for the explanation snooks. The way Anticipative FX / Live FX processing is effectively set/utilized/optimized has always been a bit of a mystery to me. I'll need to investigate it some more.

Thanks again
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Old 05-26-2017, 12:05 PM   #17
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Fun fact (from experience):

Increasing the buffer size on your Asio Interface Driver doesn't get you much with Reaper because of that buffering tricks in play already.

Ps. Still changing it to 2048 samples because some plugins just want that.
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Old 05-26-2017, 12:57 PM   #18
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True REAPER has much been better in this regard.
I've run Reaper at 64 samples Latency using two ancient Delta AP2496 cards that I've had from all the way back in the days when I ran Cakewalk Pro Audio and GigaStudio on two dedicated Windows 98 machines.

In January I upgraded my DAW from 32 bit to 64 bit, and with a clean install of Win7-64 and the latest drivers for the old Delta AP2496 cards, I still run all my projects from start to finish using ASIO set to 64 samples. I never get pops or crackles, and I never ever freeze or bounce any tracks. My last project had 107 effects on 26 tracks, and plays smooth as butter. Many of the effects were things like PSP Vintage Warmers, Lexicon MPX reverbs, and others that use a bit of CPU juice.

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Old 05-26-2017, 01:09 PM   #19
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So can I safely stop being impressed with the OP, because what they've done is just standard stuff? Load off my mind.
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Old 05-26-2017, 01:12 PM   #20
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So can I safely stop being impressed with the OP, because what they've done is just standard stuff? Load off my mind.
Basically yes.

It's a great change for Studio One users, but compared to systems in place in DP, Logic, Cubase, PT, Reaper and others... it's nothing ground breaking.
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Old 05-26-2017, 02:27 PM   #21
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So can I safely stop being impressed with the OP, because what they've done is just standard stuff? Load off my mind.
The feature as described sounds legit actually. Multiple I/O buffers for record armed tracks vs. the rest.

Low latency is of course a moot point unless you are running live sound or monitoring live performance inputs.

Sounds like a way to get deep into a studio tracking project and then have the ability to track a live midi instrument (played through a live plugin) without having to render stems to track to in order to dial your system latency back to 'live sound mode'.

Lots of people doing just that. Studio projects needing processing headroom that rules out live sound low latency but also tracking midi instruments played live with instrument plugins.


I don't believe Reaper has THAT ability. (Process the existing tracks ahead of time within the first block size interval. This is the point in time to be in sync with now here. Now the 2nd block size (set low) is your interval to handle the live tracks with the buffered 'everything else'.)

But Reaper can let you hit significantly lower latency with the same processing load than S1. This will do nothing to improve S1 there. It would just give you the above scenario as long as the system was capable of running at whatever that low latency mark was in the first place.

All this is only relevant if you are running live sound or doing live VSTi performing.
It probably should be mentioned again that if you are setting your system for low latency when you are only mixing already recorded tracks, you are significantly reducing your performance for literally no benefit. (Would be like driving in first gear on the highway or something.)

Last edited by serr; 05-26-2017 at 02:35 PM.
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Old 05-26-2017, 03:45 PM   #22
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so that seems to answer my question.
I'm always on 256 (block blabla) as long as I don't put a plugin that introduces latency in my armed track I'm always ready to go, even if on the other tracks I have tons of plugins with huge latency. (That wasn't the case at all on Ableton, for example)


Quote:
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Reaper does it automatically.

In Reaper you set your audio interface buffer size to what you want it at for live monitoring/recording. If you have Anticipative FX and Media Buffering enabled (they enabled by default), everything else is running with an effective playback buffer of thousands of samples. Rec-arming a track automatically bypasses those additional buffer settings.
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Old 05-26-2017, 04:15 PM   #23
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Sounds like a way to get deep into a studio tracking project and then have the ability to track a live midi instrument (played through a live plugin) without having to render stems to track to in order to dial your system latency back to 'live sound mode'.
I record my midi drums, usually as the very last track in Reaper, and I play them on a set of V-Drums that are triggering Superior Drummer 2. By the time I'm tracking the drums, I always have at least 60-70 plugins running, and sometimes more like 100. I never change the ASIO latency on my system from 64 samples for any reason, or ever render anything to stems.

A typical song will have 24+ tracks, most of which are audio, with 3 to 5 plugins on each. Midi V-Drums / Superior Drummer gets recorded last, because I wait until the song has been completely arranged, and mostly mixed before I lay down a one shot take of drums for it.

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Old 05-26-2017, 04:31 PM   #24
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I record my midi drums, usually as the very last track in Reaper, and I play them on a set of V-Drums that are triggering Superior Drummer 2. By the time I'm tracking the drums, I always have at least 60-70 plugins running, and sometimes more like 100. I never change the ASIO latency on my system from 64 samples for any reason, or ever render anything to stems.

A typical song will have 24+ tracks, most of which are audio, with 3 to 5 plugins on each. Midi V-Drums / Superior Drummer gets recorded last, because I wait until the song has been completely arranged, and mostly mixed before I lay down a one shot take of drums for it.

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Sure. If the projects stay small and there's no reason to increase your I/O buffer for more processing headroom, then done and done.

I'm thinking of the example where there are more like 300 tracks and just as many plugins and you need that 1024 sample block size to open up more headroom. Or perhaps someone using an older spec computer and hitting the back wall sooner.

I don't use midi instruments for anything. Just mics in front of sound sources. Live sound for me is running a live band with mics on a stage in front of a crowd. But there are lots of midi instrument players out there these days. I see where they're going with this. It's not going to help their bottom latency abilities one bit but it will offer a workaround for anyone in-between.
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Old 05-26-2017, 04:45 PM   #25
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Sure. If the projects stay small and there's no reason to increase your I/O buffer for more processing headroom, then done and done.

I'm thinking of the example where there are more like 300 tracks and just as many plugins and you need that 1024 sample block size to open up more headroom. Or perhaps someone using an older spec computer and hitting the back wall sooner.

I don't use midi instruments for anything. Just mics in front of sound sources. Live sound for me is running a live band with mics on a stage in front of a crowd. But there are lots of midi instrument players out there these days. I see where they're going with this. It's not going to help their bottom latency abilities one bit but it will offer a workaround for anyone in-between.
I suppose if you were recording or composing for symphony orchestra, you might run into issues like that. I generally have one instrument per track, and I couldn't name 300 different instruments, even if I got into the odd named percussion ones. I cut my teeth on 1" 8-track and 2" 24-track tape, so I still produce music from that angle, only using Reaper instead of Ampex. :-)
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Old 05-26-2017, 05:41 PM   #26
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Is there some feature besides the I/O buffer control ?
Looks like it to me, yeah. It looks more like their VSL hardware thing. My personal guess (and we're obviously all guessing) is that it's closer to what Pyramix does than Logic, Reaper, Samp. The low latency monitor paths seem to be created on the fly as needed.

No clue if it's better or worse than anything else but as long as people who use it who need to run at low latency can do that I guess it kinda doesn't matter.

One guy explained it to me this way, it's three software layers, native low latency monitoring (green z), hardware low latency monitoring (blue z), and playback processing. For the first one, you can't run 2ms round trip if a plugin in the monitor path is 11ms. That would defy physics and you can't anticipate or compensate for a live input, so the native low latency monitor path there is literally capped, it can only use audio plugins with latencies up to 3ms, in the NLLM monitor path.

Sorry for the edits but yeah, it seems to be a bit of a different design overall. And yeah, the playback buffer seems to be responsible for the cpu gain or stopping the cpu spiking people always complained about at low latency, while the other input thing is (I suppose) just more ensuring low latency from in to out in all cases at 16 samples or whatever you can run at.

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Old 05-26-2017, 06:37 PM   #27
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The playback buffer settings are shown in their video, along with the device buffer settings.
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Old 05-26-2017, 06:49 PM   #28
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Yeah, but there's a little more going on there than just that, in the way that it works overall, for better or worse. If you muck around around with it a little you'll eventually see what I mean.
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Old 05-26-2017, 09:44 PM   #29
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Coincidentally, I finally wiped Studio One Pro off my machine for good yesterday.

Dont suppose it is worth anything at all - v2.65 - so I put it down to a good learning experience.

Some nice ideas but in the end the poor performance and set of workflow blinker it forced me to wear was enough to bring me back to Reaper with occasional Sonar use again.
FWIW I did the same with Harrison Mixbus.
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Old 05-27-2017, 01:34 AM   #30
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I observed that reaper's performance tremendous on low buffer settings .
In fact when i pushed the buffer to a higher size, it did not make much of a difference.
We used multiple instances of DIVA and REPRO on cubase and Reaper to compare,
On my system, what REAPER could do on 128, CUBASE needed 512 or 1024 .

However when we pushed up the instances say from 8 to 20 of each plug ,
the pops and crackles did not go away in REAPER even after pushing the buffer at 2048 but CUBASE got stable on the higher buffer.

I don't know what to infer from this or if i goofed up on some setting , though the verdict seemed that Reaper was efficient on low buffer but CUBASE could push the performance on high settings.
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Old 05-27-2017, 03:02 AM   #31
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If you muck around around with it a little you'll eventually see what I mean.
Ditto. I might politely suggest that you look at the ASIO SDK to see what Direct Monitoring/VSL is, to avoid confusing VSL and playback buffers.
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Old 05-27-2017, 04:28 AM   #32
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Ditto. I might politely suggest that you look at the ASIO SDK to see what Direct Monitoring/VSL is, to avoid confusing VSL and playback buffers.
I was comparing it's switchable monitoring to the DM switchable monitoring, not the playback buffers.

The playback buffers have nothing to do with low latency monitoring except for allowing a bit more cpu headroom. The low latency monitoring there switches on and off, like ASIO DM. It works much the same as if for instance I was in the previous version running at 2048 samples and switched on hardware direct monitoring, except it also uses software plugins.

Sorry if my intent there wasn't clear.
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Old 05-27-2017, 04:32 AM   #33
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Scene of Action said something about S1's low latency being crap that didn't work properly... There's a video of him switching to Reaper and being blown away.
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Old 05-27-2017, 04:44 AM   #34
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What a surprise to me to learn how Reaper handles low latency. I never knew I could set at a lower latency and then Reaper would increase buffers as necessary, plus you can have a separate track with effects during recording ... I always used outboard effects for monitoring purposes only, like reverb during a vocal recording

Experimenting with this, however, I found a strange behaviour. I already knew that arming for recording will increase RT cpu usage, even when the armed track is muted. Here is another important one along similar lines. If you have your recording track inside a folder it will increase RT usage even if is the track not routed through the folder(with inserts) at all. Once you move it outside the folder RT usage falls.

I would have liked to have a recording track inside the folder where it is supposed to finally end up. Not possible. You have to set up the recording track on its own with plugins that don't use a lot of cpu, then after recording move the result to the appropriate folder carrying the heaver plugins.

Maybe be not a big deal, but you have to be aware of this behavior and do the work-around.
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Old 05-27-2017, 04:52 AM   #35
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@Lawrence: Because you mentioned it responding to serr asking if there was some feature other than I/O buffer going on with the latest update, saying that it's more like Pyramix and that "(t)he low latency monitor paths seem to be created on the fly as needed." Mentioning Pyramix and direct monitoring in response to that is a red herring, deliberate or otherwise.

Switching back and forward between playback and device buffers is exactly what other DAWs do. On the fly. You're trying to imply that S1 has done something special, instead of just copying Steinberg as they always do.
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Old 05-27-2017, 05:00 AM   #36
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I never implied any such thing. I only said it's "different" (the monitoring part) and explained why I thought so, with the 3ms cap on audio plugins being monitored being one of the differences. I never said it was "special", or even implied that was better. I've never used Logic or Steinberg ASIO guard so... maybe they work the same. Is there a ceiling on audio plugs being monitored there?

The other difference is that core management in audio setup is gone now... the Pyramix comparison... they might be hijacking a couple of cores for the low latency part like Pyramix does, no idea, but you can no longer turns cores off.

At any rate it doesn't matter. It's just software. Thanks Snooks.

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Old 05-27-2017, 05:20 AM   #37
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Only they obviously won't be, since that would require either always having only half (if you are taking 2 of 4) of processing power available all the time, or only when a track is rec armed. Both situations are not realistic since S1 has been slated for years for bad CPU usage, they are not going to cut it in half or have projects fall apart when you hit rec arm.

Re the 3ms limitation in their implementation, is that a total limit? Does having two rec armed tracks with 3ms of PDC on one result in the 2nd track's monitoring being delayed by 3ms too?
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Old 05-27-2017, 05:25 AM   #38
Lawrence
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The 3ms ceiling is a limit for what audio plugins will be active and heard during low latency monitoring. Anything with latency over that gets bypassed in low latency mode.
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Old 05-27-2017, 05:29 AM   #39
snooks
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Okay thanks, so does the PDC on one live monitored track affect all other live monitored tracks?
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Old 05-27-2017, 05:36 AM   #40
Lawrence
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No idea, but I don't think so... if you mean latency. I'm pretty you can't do PDC on live inputs.

Last edited by Lawrence; 05-27-2017 at 05:44 AM.
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