ORIGINAL: Jose7822

ORIGINAL: Treefight


Did the bakers really make - or try to make - CEntrance unecessary in S8? I really enjoyed and benefited from your instructions on manually offsetting. Has anyone actually checked to see if S8 is really sample-accurate or whatever the term is so that you don't need to test and manually adjust?

Thanks.

At least that's what the bakers said when the full official list of Sonar 8 came out. Yet, the question still remains. Has anyone who owns Sonar 8 tested if it was sample accurate? On a similar note, the work done to the ASIO driver model in Sonar 8 did improve the External Insert Plug, though it's still not perfect according to some of our forum members. So, anyways, who here has the answer to our question? It will be much appreciated.

Take care!

I've had 8 loaded for 2 days... I specifically looked at my CEntrance results and the Options/Audio latency adjustments. I have to make the exact same manual corrections for both S7 and S8. I see no change in how SONAR is dealing with this.

I'm on the road for a few days and will repeat the test when I get back.

best regards,
mike

ORIGINAL: mike_mccue


I've had 8 loaded for 2 days... I specifically looked at my CEntrance results and the Options/Audio latency adjustments. I have to make the exact same manual corrections for both S7 and S8. I see no change in how SONAR is dealing with this.

I'm on the road for a few days and will repeat the test when I get back.

best regards,
mike

Well that's not good .

Another question, Jose - in the post where you explain how to do the CEntrance test, you say that the buffer size in Sonar doesn't matter for purposes of the test.

I got different results, however, with Sonar 8 and a Presonus Firestudio Tube on Asio.

At 512 Presonus buffer, the offset using the CEntrance method is 89.

At 96, the offset is 107.

Very close in any event, but curious about the reasons behind this. Also, I could not do the test in the default CEntrance input sensitivity of -12. I have to use -18 or -24, and when alternating, I also get very small variations in the result - one sample difference most of the time, but if I keep pressing measure, it may drop 7-8 samples. I know it's still tiny, but just wondering if this means the test isn't sound on my rig for some reason or these are normal fluctuations.

I'll do it the hard way just in case.

Thanks.

ORIGINAL: Jose7822

ORIGINAL: brammer

Hey, I just want to TRY it at lower than 5.8

It might "tighten up" my tracks

Then again.... maybe I just haven't got the "game" to play tighter than it is.........

Wow, that would suck. It would mean I really AM that bad :-(

Brammer

Don't blame yourself just yet. Even though Sonar 6 introduced ASIO Latency Compensation, the compensation is not total. IOW, you still need to adjust it in order to make it perfect (sample accurate). This has been supposedly fixed in Sonar 8 whwre no manual offset is needed unless you add to the chain (at least that's what I'm assuming). If you're working on either Sonar 6 or 7 then try this latency test out to make your system sample accurate:

http://forum.cakewalk.com/fb.asp?m=1302373 (Post# 20)


HTH

Guitarists, on the other hand, normally stand at least 3 feet from their amps (the ones who still have any hearing left anyway) and usually farther than that. And they don't seem to have any problem with delays, and would therefore be likely to assume short delays are inconsequential. But it's more about the nature of the instrument and the parts they play.

When I play live, I go wireless and occasionally walk around the club. I can say without question that I suck just as bad when I am 30 feet from the stage as I do when I'm 4 feet from my amp.

Bob , how do you deal with latency?

I do input monitoring in my studio. With my current machine, I'm able to do everything at 2.9mS. This is more than fine for me. I know it's unacceptable for some, but maybe it's the years of playing live with horribly useless monitors that makes me so tolerant

ORIGINAL: Bob Damiano [Cakewalk]

Guitarists, on the other hand, normally stand at least 3 feet from their amps (the ones who still have any hearing left anyway) and usually farther than that. And they don't seem to have any problem with delays, and would therefore be likely to assume short delays are inconsequential. But it's more about the nature of the instrument and the parts they play.

When I play live, I go wireless and occasionally walk around the club. I can say without question that I suck just as bad when I am 30 feet from the stage as I do when I'm 4 feet from my amp.

Hi Bob, that's exactly what I was thinking when I read Bitflipper's analysis of who might be more or less sensitive to latency. I have a special 40 foot audience chord that I used to plugin for dancing with the crowd.... Mikey don't use nO wireless :-) (well for work yes... but for me no.)

That and the fact that many of us learned stage craft while using slap back off the rear wall for monitors has made me rather tolerant of latency.

I can see how other types of instrumentalists might have very different needs.


Reading your post made me smile... thanks!

best regards,
mike

ORIGINAL: Treefight

Another question, Jose - in the post where you explain how to do the CEntrance test, you say that the buffer size in Sonar doesn't matter for purposes of the test.

I got different results, however, with Sonar 8 and a Presonus Firestudio Tube on Asio.

At 512 Presonus buffer, the offset using the CEntrance method is 89.

At 96, the offset is 107.

Very close in any event, but curious about the reasons behind this. Also, I could not do the test in the default CEntrance input sensitivity of -12. I have to use -18 or -24, and when alternating, I also get very small variations in the result - one sample difference most of the time, but if I keep pressing measure, it may drop 7-8 samples. I know it's still tiny, but just wondering if this means the test isn't sound on my rig for some reason or these are normal fluctuations.

I'll do it the hard way just in case.

Thanks.

Hey Treefight!

In my limited experience with this, I have found the offset to be consistent at different buffer sizes. But, to me, it seems very strange that 1) you're getting different offset results and 2) the ASIO Latency Tool wont let you change the Input Sensitivity to the -12 option. I believe there's something going on with your setup that's making CEntrance not get accurate results. From what I remember, whenever I tested the latency offset of my setup using CEntrance I would get consistent measurements all the time (even at different buffer sizes). As soon as I can, I will conduct a test to confirm my results. I'll post then.

Take care!

all this buffer number talk is making my head spin...............

SO IT IS A CHALLENGe FOR THe CAKEWALKERS to GET IT AS RIGHT AS STEINBERG! THE MIDI TIMING AND THE AUDIO LATENCY CORRECTIONS OF CUBASE ROCK !!! WHY CANT CAKEWALK DO THAT????????????????????????????????????????????????????????????

ORIGINAL: madratter

It should not be necessary to "get a real keyboard with audio out. " Many of us use VSTs like True Piano and record with them. Somewhere, you have latency being introduced. Possibly you aren't setting the latency for your interface properly.

Another possibility is you have added effects to your VST such as Perfect Space. That can and does introduce quite a bit of latency at the default values perfect space uses.

I just read an an article somewhere that with USB-interfaces roundtrip latency around 6-8 ms is "normal" today.
I think mine is around 15-20 (??). Here on the forum I've read many posts sayinmg they're latency is set to 2-3 ms.

montezuma


Is the sample rate related to latency at all? My sample rate is 256...not really sure I understand sample rate...people often talk about lowering the sample rate and being impressed that an interface can operate glitch free on low sample rates...where I would have thought that the higher that figure the better...but...I dunno

I think you're confusing sample rate and hardware buffers.  256 is not a sample rate type of number.  most soundcards won't record lower than 44.1kHz sampling rate.  but hardware buffers are usually in increments of 32, 64, 128, 256, 512, 1024. 
 
higher sampling rate DOES lower latency, but increases resource usage (CPU, and Hard drive mainly)
 
hardware buffers increase or decrease of course directly affect latency.  the higher the number the higher the latency.

Whilst this does seem like a Sonar / Soundcard config issue I just wanted to throw in that if you have other plugs in your project, like perfect space, Sonar's PDC will add further latency to your monitoring path.

Might be worth being aware of at least.
Child

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