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Cubase 6.5??? Still runs like a TURD??? Audio Interfaces
Old 3rd March 2012
  #61
Gear maniac
 

Here's a one minute long sample of a Cubase 6.5 session. The most frequent CoreAudio call comes in at a whopping "0.0%" frequency. The "coreaudio2asio" call takes up 1.3% of the cpu time. Most of Cubase's time is spent processing semaphores aka scheduling and locking which BTW is the same thing Logic is doing most of the time. The Cubase engine as coded for the Mac is what is broken here, not CoreAudio.

Old 3rd March 2012
  #62
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TAFKAT's Avatar
 

Quote:
Originally Posted by Mike P View Post
Um, yes, I play instruments. 14 guitars and basses, multiple amplifiers, etc. As I said earlier, whether I'm mic'ing amps, doing ADR, vocals, or playing live MIDI tracks from Kontakt, the card is never set lower than 128 samples..
All you have listed in regards to live recording is via ADM /HDM ( ASIO Direct Monitoring/ Hardware Direct Monitoring ) , the buffer setting has no relevance in regards to RTL in this instance , you are only dealing with the AD/DA.

If you were using Guitar Amp Simulators for example and required monitoring via software thru the Amp sims, you would appreciate the need for running at latencies below 128.

Old 3rd March 2012
  #63
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zephonic's Avatar
uh oh...

...this should be fun...
Old 3rd March 2012
  #64
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Quote:
Originally Posted by TAFKAT View Post
All you have listed in regards to live recording is via ADM /HDM ( ASIO Direct Monitoring/ Hardware Direct Monitoring ) , the buffer setting has no relevance in regards to RTL in this instance , you are only dealing with the AD/DA.

If you were using Guitar Amp Simulators for example and required monitoring via software thru the Amp sims, you would appreciate the need for running at latencies below 128.

I'm not using Direct Monitoring. I'm monitoring everything through Nuendo. And when I use Kontakt to play organs, pianos, etc. in real time, same thing. I've been recording like this since 2007 without any latency or timing issues.

Typical Signal Chain: Mic>Preamp>Compressor>Apogee>RayDat>Nuendo>DA16x>Neve 8816 w/Vintech 906ca on the 2 buss

Again, this is the core of my question. I've never had any issues with timing or latency at 128 samples, nor has anyone that's ever recorded vocals or performed ADR in my studio.
Old 3rd March 2012
  #65
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Quote:
Originally Posted by robotpriest View Post
Here's a one minute long sample of a Cubase 6.5 session. The most frequent CoreAudio call comes in at a whopping "0.0%" frequency. The "coreaudio2asio" call takes up 1.3% of the cpu time. Most of Cubase's time is spent processing semaphores aka scheduling and locking which BTW is the same thing Logic is doing most of the time. The Cubase engine as coded for the Mac is what is broken here, not CoreAudio.
LOL who said anything about core audio being broken ?

Care to explain all of the other DAW hosts , I doubt they are using Steinis audio engine.
Old 3rd March 2012
  #66
Gear maniac
 

All these DAWs work in realtime and spend most of their time processing locks. If Steinberg tunes their engine to the Mac's kernel, it will tighten up performance. But if they try to write a Windows app and then stuff it in a Mac wrapper...performance will suffer.

Anyway that is what I would *guess* is happening. It happens all the time with games.
Old 3rd March 2012
  #67
Gear maniac
 

Ok, let me put forth a hypothetical answer to why other DAWs are performing better in Windows. First of all, most of the original developers are probably Windows developers. To make a Mac version they have to hire from the outside. Many of them are based outside of the USA where Macs are like white tigers (very rare). Crackerjack Mac coders are super hard to find. So they make a Windows version and then use some sort of cross-platform tools to either wrap or simultaneously develop for both platforms.

The problem is that thread locking is super sensitive to what kernel it was coded for. None of these devs wants to code each platform from the ground up, so they focus on the one which has 10x the userbase: Windows. And you end up in the situation where only the Apple product has high performance on the Mac. Which btw, is EXACTLY one of the reasons why Apple would not want Flash on the iPad. Lazy devs will always choose the crossplatform dev tools, focus on the platform with the highest installed base, and the end user will assume there is something wrong with the Apple product.

None of this really matters though because its pure speculation (although it is fun to do!). The point is just that Logic has high performance. Who cares what the problem is with the other DAWs, I just wish Cubase could pony up to Logic's level of performance.
Old 3rd March 2012
  #68
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TAFKAT's Avatar
 

Quote:
Originally Posted by robotpriest View Post
All these DAWs work in realtime and spend most of their time processing locks. If Steinberg tunes their engine to the Mac's kernel, it will tighten up performance. But if they try to write a Windows app and then stuff it in a Mac wrapper...performance will suffer..
What Wrapper ?

Again, as I noted earlier what you are proposing is that all of the cross platform DAW developers are inept at developing their applications for OSX.

Well I have to say I will be very interested how Digital Performer 8 scales on Windows /ASIO , especially if it performs comparably with the established DAW hosts, as this is the first time in recent memory that a MAC DAW host has been ported to Windows.

Quote:
Originally Posted by Mike P View Post
I'm not using Direct Monitoring. I'm monitoring everything through Nuendo. And when I use Kontakt to play organs, pianos, etc. in real time, same thing. I've been recording like this since 2007 without any latency or timing issues.

Typical Signal Chain: Mic>Preamp>Compressor>Apogee>RayDat>Nuendo>DA16x>Neve 8816 w/Vintech 906ca on the 2 buss

Again, this is the core of my question. I've never had any issues with timing or latency at 128 samples, nor has anyone that's ever recorded vocals or performed ADR in my studio.
You do not use direct monitoring when recording live instruments, good grief, why not ?

There is no overhead imposed and you are dealing with substantially lower RTL, Win/Win !

When Playing instruments using Kontakt you are only dealing with DA latency not RTL. Vocals if you are monitoring with vanity FX to the talent @ 128 samples, sure , but that depends on the talent. You also have the advantage of using one of the better audio interfaces in regards to RTL.
Old 3rd March 2012
  #69
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zephonic's Avatar
The first Nuendo engine was developed for Windows exclusively and only later ported to Mac. It remains at the core of all current Steinberg products.

As for ProTools cross-platform performance, when used with its own DAE engine rather than ASIO, it is remarkably similar on both platforms.

It will be interesting to see how Digital Performer will fare cross-platform.


BTW, Vin, any chance of Kontakt 5 compatibility for the VI benchmark anytime soon?

Last edited by zephonic; 3rd March 2012 at 01:00 AM.. Reason: btw
Old 3rd March 2012
  #70
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Quote:
Originally Posted by Mike P View Post
Um, yes, I play instruments. 14 guitars and basses, multiple amplifiers, etc. As I said earlier, whether I'm mic'ing amps, doing ADR, vocals, or playing live MIDI tracks from Kontakt, the card is never set lower than 128 samples.

My website is in my signature, if for some reason you don't believe me.



That may be the case with your system but it certainly is not with mine. I cranked out more than 150 tracks last year and scored an Indie Feature film. Never once was the latency below 2.95ms at any time.



I just switched from an RME HDSP 9652 (old PCI system) to a RayDat PCIe card for my new i7 build. The drivers are identical. Both cards interface to Apogee X16 series converters.
What´s your input and output latency? If those both are 1.475ms with 128 buffer size, then you might just have the fastest DAW in the universe.

What sequencer do you use? I use Cubase and it shows both input AND output latency. You have to sum those numbers to get the complete latency.

But really, you don´t notice the latency with 128 buffer when you are playing guitar for example? I guess that´s good for you. I notice it with 32 buffer and anything more would be too annoying to play.
Old 3rd March 2012
  #71
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Quote:
Originally Posted by TAFKAT View Post
You do not use direct monitoring when recording live instruments, good grief, why not ?
My standard rock template has two instances of Lexicon Native Reverb, two instances of PSP 42 delay, Guitar Rig 4 (running the Chorus/Flanger for bass), twelve Waves SSL Channels and a few sorted CLA Waves compressors. At 128 samples, recording "Live", the ASIO meter in Nuendo is at less than 10%.

I've never experienced clips and pops nor any issue related to timing. I compose/produce music for a living and I'm often on a tight deadline. But all that said, it's supposed to be "fun". Fun for me is hearing effects when I'm tracking, not listening to some dry guitar or bass cab in an iso booth.

Old 3rd March 2012
  #72
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Quote:
Originally Posted by GearNerd View Post
But really, you don´t notice the latency with 128 buffer when you are playing guitar for example? I guess that´s good for you. I notice it with 32 buffer and anything more would be too annoying to play.
No, I do not notice the latency at 128ms, nor has anyone that's ever recorded in my room, whether it's multiple singers (including Gearslutz own, James Lugo) or Emmy-Winning actors and actresses performing ADR.

All of which points back to my initial question: Why record at 32ms latency? If I were to do that, the end result would not be any different and I'd put an unnecessary strain on my computer.
Old 3rd March 2012
  #73
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TAFKAT's Avatar
 

Quote:
Originally Posted by Mike P View Post
My standard rock template has two instances of Lexicon Native Reverb, two instances of PSP 42 delay, Guitar Rig 4 (running the Chorus/Flanger for bass), twelve Waves SSL Channels and a few sorted CLA Waves compressors. At 128 samples, recording "Live", the ASIO meter in Nuendo is at less than 10%.

I've never experienced clips and pops nor any issue related to timing. I compose/produce music for a living and I'm often on a tight deadline. But all that said, it's supposed to be "fun". Fun for me is hearing effects when I'm tracking, not listening to some dry guitar or bass cab in an iso booth.

Right, well earlier it didn't sound to me like you were monitoring with FX or using Amp Sims , you were talking about Mic'ing Amps, and external record signal chains, etc , thats what I was referring to.

Anyhow, what ever works for you , if 128 is a working sweet spot , more power to ya , that doesn't negate the fact that some prefer the tighter RTL , not necessarily 032 , but I know plenty that prefer to work at 064.

And on that point , there is so much variance at what respective buffer settings actually deliver across various interfaces, you can't just quote a number and expect that to be valid across the board. It will be interesting to know what your actual RTL really is via the RME/Apogee , as the RME card will not report the AD/DA to the DAW host.

BTW: Cubendo on OSX sucks comparatively even at 128 compared to Windows, so we have come full circle and back on topic... :-)
Old 3rd March 2012
  #74
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Quote:
Originally Posted by TAFKAT View Post
Anyhow, what ever works for you , if 128 is a working sweet spot , more power to ya , that doesn't negate the fact that some prefer the tighter RTL , not necessarily 032 , but I know plenty that prefer to work at 064.

And on that point , there is so much variance at what respective buffer settings actually deliver across various interfaces, you can't just quote a number and expect that to be valid across the board. It will be interesting to know what your actual RTL really is via the RME/Apogee , as the RME card will not report the AD/DA to the DAW host.

BTW: Cubendo on OSX sucks comparatively even at 128 compared to Windows, so we have come full circle and back on topic... :-)
Yeah, everything works great at 128 and I really dig my workflow, which is why initially inquired as to why someone would chose to work at 32 or 64 samples.

I have no idea how to test the latency through the RayDat/Apogees and with my current delivery schedule, it wouldn't be until sometime in April before I could actually take time to test it.

Thanks for your time and testing, as that info has helped out friends and family members!
Old 3rd March 2012
  #75
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zephonic's Avatar
Quote:
Originally Posted by TAFKAT View Post
BTW: Cubendo on OSX sucks comparatively even at 128 compared to Windows, so we have come full circle and back on topic... :-)
While I can't compare it to Windows, Cubase 5.5.3 runs just fine @128 with an Mbox3Pro on OSX 10.6.8
In fact, I was unwittingly running it @64 for a while, and never noticed the difference. My workload doesn't require 200+ voices so perhaps that's why.

But 32 sample buffer is a no-go zone on my machine. I briefly tried that today and it's unusable.
Old 3rd March 2012
  #76
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Quote:
Originally Posted by TAFKAT View Post
Right, well earlier it didn't sound to me like you were monitoring with FX or using Amp Sims , you were talking about Mic'ing Amps, and external record signal chains, etc , thats what I was referring to.
I'm doing all of the above. I record guitar and bass cabinets while monitoring through Nuendo with effects on group channels. I'm not a fan of amp sims but GR4 has some a amazing effects that I use as send/returns as well as channel inserts.

Their Transamp (Sansamp) is awesome for vocals.
Old 3rd March 2012
  #77
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TAFKAT's Avatar
 

Quote:
Originally Posted by Mike P View Post
I have no idea how to test the latency through the RayDat/Apogees and with my current delivery schedule, it wouldn't be until sometime in April before I could actually take time to test it.
Hey Mike,

If you can spare 5 minutes , you can use the RTL Utility I helped develop.

Available Here

It will give you an accurate RTL at each specific buffer setting.

Even pinging one latency setting will give us the value of the AD/DA , I can actually work out the rest from there as long as I have the reported values in Cubendo for I/O.

The RAYDAT values are a little different to the HDSPe AES.

Old 4th March 2012
  #78
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Quote:
Originally Posted by TAFKAT View Post
Hey Mike,

If you can spare 5 minutes , you can use the RTL Utility I helped develop.

Available Here

It will give you an accurate RTL at each specific buffer setting.

Even pinging one latency setting will give us the value of the AD/DA , I can actually work out the rest from there as long as I have the reported values in Cubendo for I/O.

The RAYDAT values are a little different to the HDSPe AES.

Hey Vin,

Thanks for linking me to the test. It appears to be a very nice looking program and I'd love to be able to test, but I don't think my system is set up in a such a way that it'll work.

The AD has an 8 channel DSub to TRS cable, connected via ADAT to the RayDat card, so we're good there. The problem is that the DA16x is connected DSub to Dsub - Converters to Neve 8816, so I don't have an actual TRS output in which to plug into an input. Please let me know if I'm overlooking something.

For the record, at 128 samples, the Raydat says 2.948ms Input Latency and 3.696ms Output Latency for a total of 6.644ms round trip. I swear, I can't hear it, nor has anyone that's ever recorded here. It sounds like everything is in "Real Time", with no delay whatsoever.
Old 4th March 2012
  #79
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Quote:
Originally Posted by Mike P View Post
For the record, at 128 samples, the Raydat says 2.948ms Input Latency and 3.696ms Output Latency for a total of 6.644ms round trip. I swear, I can't hear it, nor has anyone that's ever recorded here. It sounds like everything is in "Real Time", with no delay whatsoever.
A-ha! Finally I got the right numbers from you. I notice 6.6ms latency easily when playing guitar, but I wouldn´t notice it singing.


Quote:
Originally Posted by Mike P View Post
No, I do not notice the latency at 128ms, nor has anyone that's ever recorded in my room, whether it's multiple singers (including Gearslutz own, James Lugo) or Emmy-Winning actors and actresses performing ADR.

All of which points back to my initial question: Why record at 32ms latency? If I were to do that, the end result would not be any different and I'd put an unnecessary strain on my computer.
So you got your answer right there. Some people do notice the latency and it affects their performance, for the worse of course. 32 buffer is an absolute MUST for me and many others when recording guitars!

The thing that pisses me off in this forum is that when some people don´t have the need for faster computers, better latency, 64-bit, etc... better technology in general. They don´t understand why some people might have that need and they put down those people by saying "everything´s great, we don´t need faster computers, 32-bit is just fine, 128 buffer is just fine, we don´t need 32 buffer latency, Beatles made records with 4-track BS etc...". Just sick of hearing it! Nothing personal against you.
Old 4th March 2012
  #80
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Quote:
Originally Posted by GearNerd View Post
A-ha! Finally I got the right numbers from you. I notice 6.6ms latency easily when playing guitar, but I wouldn´t notice it singing.
I don't notice, period, whether it's guitars, bass, MIDI, vocals or spoken word. Neither has anyone else that's ever recorded here.
Old 4th March 2012
  #81
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Quote:
Originally Posted by Mike P View Post
I don't notice, period, whether it's guitars, bass, MIDI, vocals or spoken word. Neither has anyone else that's ever recorded here.
Good for them and you!
Old 4th March 2012
  #82
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Quote:
Originally Posted by GearNerd View Post
So you got your answer right there. Some people do notice the latency and it affects their performance, for the worse of course. 32 buffer is an absolute MUST for me and many others when recording guitars!

The thing that pisses me off in this forum is that when some people don´t have the need for faster computers, better latency, 64-bit, etc... better technology in general. They don´t understand why some people might have that need and they put down those people by saying "everything´s great, we don´t need faster computers, 32-bit is just fine, 128 buffer is just fine, we don´t need 32 buffer latency, Beatles made records with 4-track BS etc...". Just sick of hearing it! Nothing personal against you.
What sound card / converters are you using? I don't know anyone that needs to have their computer at 32 samples in order to record tight tracks. The only people I know of that require that type of buffer settings post in this forum.
Old 4th March 2012
  #83
Gear addict
 

for me Cubase 6 works better on a bootcamp windows 7 than on snow leopard , I have macpro 2006...I'm so glad about that...

slow performance (32 tracks mix in 24/88: latency: 512 on mac / 126 on windows 7 same plugs)
slow zoom on OSX
slow opening mixing board

I have an RME fireface + apogee rosetta,

so everyday I work with windows, next time I will buy a PC for sure (cheaper!)
Old 4th March 2012
  #84
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Quote:
Originally Posted by Mike P View Post
What sound card / converters are you using? I don't know anyone that needs to have their computer at 32 samples in order to record tight tracks. The only people I know of that require that type of buffer settings post in this forum.
I already said Symphony I/O. Well now you know at least one person. I know plenty of others.

It´s really annoying to start playing guitar and notice latency, even though 6ms is not a lot, but it´s still noticeable. Thank god there is 32 buffer!
Old 4th March 2012
  #85
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Quote:
Originally Posted by GearNerd View Post
I already said Symphony I/O. Well now you know at least one person. I know plenty of others.

It´s really annoying to start playing guitar and notice latency, even though 6ms is not a lot, but it´s still noticeable. Thank god there is 32 buffer!
Ah, that makes sense. The Apogee drivers are vey poorly written, IME. I tried the Symphony system years ago on a Mac G5 and lowest latency I could get at the time in Nuendo was like 12ms.

I'd bet that if you were using a different interface, like an RME for instance, you wouldn't need to be at 32 samples to be latency-free.

RME drivers are second to none.
Old 5th March 2012
  #86
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Quote:
Originally Posted by Mike P View Post
Ah, that makes sense. The Apogee drivers are vey poorly written, IME. I tried the Symphony system years ago on a Mac G5 and lowest latency I could get at the time in Nuendo was like 12ms.

I'd bet that if you were using a different interface, like an RME for instance, you wouldn't need to be at 32 samples to be latency-free.

RME drivers are second to none.
Yes, tell me about it! Apogee drivers are just horrible. The 32 buffer will give me some crackles and pops on my 12-core Mac Pro, if the project is more than 20-30 tracks, which is really PITA. But the latency I get is 1.7ms + 1.7ms, which is pretty nice.

Never buying Apogee again! Going RME or Metric Halo hopefully soon.
Old 5th March 2012
  #87
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Quote:
Originally Posted by GearNerd View Post
Yes, tell me about it! Apogee drivers are just horrible. The 32 buffer will give me some crackles and pops on my 12-core Mac Pro, if the project is more than 20-30 tracks, which is really PITA. But the latency I get is 1.7ms + 1.7ms, which is pretty nice.

Never buying Apogee again! Going RME or Metric Halo hopefully soon.
Out of curiosity, I just checked and at 64 samples, I'm at 3.7ms RTL with the RayDat on a new i7 2600k/Asus Pro Gen 3 using Win7 64.

I'm testing a song with 24 audio tracks, eight Send/Return effects, eight VST channels for Superior and probably 20 Waves SSL & CLA comps. The VST meter is at about 25% during playback. No pops or clicks.

Bummer, Dude.
Old 5th March 2012
  #88
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steveschizoid's Avatar
Should probably vist the Cubase forums for this...but have any of you battled with the somewhat annoying issues having to do with variaudio in Cubase 6?

(These include cpu spikes, discovering that the variaudio has been turned off for no apparent reason, and, finally, yesterday I had a file become corrupt and had to redo a few hours of work yesterday. I had it scan a track, noticed the cpu begin spiking, so I went ahead and fixed the pitch issue, then bounced the file - but the cpu spikes remained! The only other changes I'd made were just volume automations, no extra plugins. Long story short, I ended up having to redo the rest of the work, then fix the pitch in a renamed project file, bounce the result, then import it back into the working file.)

I use Windows 7.

Any word on whether thses issues have been adressed in the update?
Old 5th March 2012
  #89
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TAFKAT's Avatar
 

Quote:
Originally Posted by GearNerd View Post
Yes, tell me about it! Apogee drivers are just horrible. The 32 buffer will give me some crackles and pops on my 12-core Mac Pro, if the project is more than 20-30 tracks, which is really PITA. But the latency I get is 1.7ms + 1.7ms, which is pretty nice.

Never buying Apogee again! Going RME or Metric Halo hopefully soon.
It has to be noted that its doesn't matter what the brand is on the box, it inevitably comes down to the controller/drivers , Apogee use and rely on an OEM 3rd party like many other manufacturers.

RME are one of the few that don't, not sure about MH.
Old 5th March 2012
  #90
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Quote:
Originally Posted by TAFKAT View Post
If that is true , something seriously squirrelly with your Win7 installation... :-)


Cross Platform DAW Performance - Part I thru to VI

Overall performance of Studio One on OSX is even worse than Cubendo :-(

I have to agree. I was running S1 this weekend in a studio, because I needed to track some drums and the S1 kept crapping out on me. Granted my MacBook Pro is about 5 years old now, but I never had an issue with Logic. The only reason I used S1 is because I already had the session files open in S1 and didn't have enough time to get them into Logic.

I only had about 12 audio tracks in total, no effects and S1 couldn't hang. Hell I get better performance out of Live. Not impressed with S1 so far.
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