What do you think of these latency results?

My typical day work is running with a 64 sample buffer.

I use the MOTU 896HDs with their ASIO driver.

That adds up to 178 sample total for a round trip.

Sonar automatically accounts for 108 of those samples and I manually set the extra 70 samples manually in the dialog on the other tab.

Using 64 samples for a buffer lets me monitor through SONAR... which is something I used to think I would never do but is something I've done ever since I realized that my DAW can actually do it.


My next DAW will hopefully go down to 32 or at least 48 samples.


When I gobber up my project at mix time I increase the sample buffer to 128, 256, or 512 as required and everything works great as I add all the extra dsp. The computations are still able to happen inside the buffer.





If I was running a system at 1311 samples I'd need to use some form of outboard monitor mix matrix... which is fine... that's what I used to do.

If I was running a system at 1311 samples I'd have to get use to the idea that every time I made some small mix adjustment that I'd have to wait until I heard the results. The 816 samples at Output essentially means that mixing by "using your ears"... will be a time delay experience... OR... you may not realize that and actually think you are hearing stuff you are not. You are certainly not going to turn an knob and hear a change... you have to turn the knob and then wait for the change... and try not to get too confused. I switch over to think and do mode in this case... and work slower, more methodically, and less viscerally. I think one should be aware of this... because one may actually think they hear some change way before the change is hear able... that stuff can make you crazy or even confused. :-)

I mention that... because i'm the only one that ever seems to mention it... and it seems kinda important.



best regards,
mike

I used to pretty much always be on time or early..... but now, I do tend to run "fashionably late"...



As far as audio latency... if I can't hear it, I don't worry over it. 

Michael, when I'm tracking on my Roland USB unit (UA-55),  I am able to use it on either 2 or 3 on the Driver Settings slider when tracking, which equates to 48 or 64 samples @48 kHz.
 
Thats quick enough not to be thrown by a delay when monitoring through whatever app I'm using.
 
It has to be a pretty heavy project for me to have to have a buffer at a higher setting to sort out any pops or crackles that might come as a result of the increased work load.
 
I know my interface is a newer model so the settings may well differ.
 
The basic setup if you are monitoring through the app has to be quick enough not to be detectable and interfere with your flow, otherwise you'll have to hardware monitor at the input stage.
 
Unlike Mike I've never found a slow enough time lag to cause me problems when mixing when I'm just adjusting out the buffer from my tracking set up just so it's big enough to remove any glitching.
 
So best place to start is with a basic project that would be enough just to track a guitar or a vocal and get the buffer as low as you can go then just move your driver setting to a higher number as and when a particular project requires it.
 
Mine runs nicely for most things at a 7ms measured round trip, and although I can get the latency lower if I needed to, I find I don't need to because it's a low enough figure not to cause me to notice it when I'm playing and I'm able to run a pretty well-loaded project without it glitching at the same setting.

LOL.....I'm enjoying my morning coffee and having a pretty good time using
my music PC on-line. It handles this forum perfectly. I've never been able to say that about
this place. Not one single issue.
IE9 32 bit on a 64 bit machine and it's smooth as silk.

" used to pretty much always be on time or early..... but now, I do tend to run "fashionably late"... " Herb people are
probably surprised. I don't know why but think they probably are. Just think how much stuff you never hear and don't
think about. :)

Mike...huh?


Well there sure are a lot of numbers in that screenshot. I know to check that 64 box and to set the slider
on that UA-101 to that little triangle pointer at the 6 and those two boxes.

I think that "mixing latency" that is 7.0 ms is like telling me I'm hearing the sound from the monitors as if I were
sitting 7 feet away. Much farther away and I may not be able to play in time with what I hear.

I'm pretty sure that some can get much closer...lol...or ms down to like 2. I've never seen below 7.

Thanks Jon.

Most of the time the only time I've had trouble was with my midi guitar and always
found a way to eliminate the problem.
Sometimes it was just closing the project and re-opening - for whatever reason.

Play and record I/0 are at 256 and I don't have a track record with X2 but had to change
from 128 for Warrens project....still had pops and such but I think it was vst issues...just don't
know.

What do you think of these latency results? 27.3ms 1311samples.

I'd say that's pretty bad. I'm getting a little better results with my samples set to that amount, but I can't set them exactly to match yours so it's a little different.

I have my Play/Record I/O set to 512. My sound card buffer set to 128 samples. Midi playback buffer set to 500ms. My project settings are 48kHz/24bit. I'm getting a reported RTL of 9.8ms and the funny thing is ... I don't feel it. Normally, I can't stand anything more than 5 ~ 6ms.

I've gotten in to the habit of freezing every track. I set my 'Bounce' settings to 32bit so dithering doesn't come in to play.

I have a project now with 16 audio and 2 synth tracks, all frozen, and I can track and monitor with effects a lead guitar part.

I hate to say anything because I get accused of being negative all the time ... but ... something feels different in X2. The Reported RTL doesn't seem to be as accurate as it used to be. To clarify, it seems I'm getting less actual latency with a higher Reported RTL.

Hell ... I PM'd Beepster about this ... he can confirm I'm not pulling this out of my ars*, I had one project where I had my sound card buffers set to 512 ... forgot about it ... and I had zero latency. I discovered it when I added the LP-64 EQ. My project started popping and crackling so I went in to raise my buffers and that's when I realized they were at 512.

Something is not right. I'm not saying it's Sonar, it could very well be the old drivers for my Fast Track Ultra. It sucks that they keep the Mac drivers current but have ditched the Windows drivers for years now. But anyway, I'm noticing a difference in that area in X2.

Maybe it has something to do with the way they updated the engine to support 384kHz now? The higher your sample rate, the lower your latency.

Another thing to be aware of ... they tweaked the LP-64 EQ and in my opinion broke something. I could run my projects at 128samples with no problems if I had the LP-64 EQ in there. (I actually started using it since they fixed the drop-out problem with it.) Since the A patch ... I can't use it again. I can set my buffers to 768 and it still causes my projects to pop and crack.

But who am I to say anything ... I apparently don't do my research on these things. I guess 14 hours a day, every day isn't enough.

I loved my old EMU-0202 (Edit: typo. Meant 0404) card. It had onboard DSP with guitar and vocal effects. I could ram my buffers up as high as they would go and I could record at zero latency because of the onboard effects.

I haven't been able to find a sound card that offers onboard effects for guitar like that since. My FTU has some crappy reverb, but that doesn't help when you need some overdrive or distortion to get in the groove of a take.

Those numbers look kind of odd to me, Michael.
Especially the output. I'm curious as to where all that overhead is coming from. Is this typical for a USB device, or is something else causing it?
Just for reference, here's what I normally see:




If I switch the sample rate to 48000, the numbers are proportionally lower, as one would expect.
Buffers are set to 256 in the sync page.
At 27ms, you are no doubt hearing a delay. When I cross that 20ms line, I can hear it enough to start getting annoying.

You can't say dsp, zero latenecy, and analog playback in the same sentence without glossing over the fact that state of the art AD/DA a takes about 2ms... minimum... so the closest to zero you are gonna get is actually 0.002 seconds.

My total round trip through SONAR is 0.005+/- seconds.

Knowing that my 0.005 seconds is really only 0.003 seconds longer than what everyone calls "zero" makes the situation almost 50% better than it would seem if "zero" were really zero.


all the best,
mike