Folks, please help with this brain-teaser: I previously posted on this thread that I got the CEntrance test working on the asio drivers for one interface, but couldn't get it to work on my Mackie 1200f. Well, finally, by changing the "Input sensitivity" on the CEntrance test window from the default of -12 to -24, I was able to take a measurement. It would only take measurements on the -24 setting. All others gave me an error message.

So I tried this with three different Buffer size/Latency settings on the Mackie's asio panel: 64, 128, and 256. I got pretty different results with each, and I'm just wondering if that's normal or if the results are supposed to be consistent, regardless of Buffer size setting? In one post, someone said your interface's setting could be set to anything, it wouldn't affect the final manual offset number. As background, my other interface's offset was 21, no matter what Buffer size I used.

So the Mackie's results were as follows (caveat below re: variable CE numbers on consecutive tests):

(CEntrance roundtrip latency) - (ASIO Reported Latency) - (Buffer Size/Latency) = Manual Offset:

539 - 256 - 64 - 219

651 - 384 - 128 = 139

939 - 640 - 256 = 43

As you can see, the offset number varies from 43 to 219! Is that "normal," i.e., if/when I change the Mackie's asio panel's buffer settings during tracking, I need to change the manual offset as well to maintain correct or close to correct timing? No big deal if it does mean that, I just want to make sure I'm not wasting my time on results that are somehow or another incorrect.

Oh, another thing that seems odd - when I run multiple consecutive tests at any buffer setting, the CEentrance rountrip latency number did NOT stay the same. So, for example, at 64 samples asio setting, the CE roundrip latency would be 510, then 493, then 539, and so on and so on. What of this? Does this mean it isn't working or that I should just take the average of several test results and use that in the equation?

Thanks very much in advance!

Thanks, Jim.

Would you mind explaining how I check the loopback in Sonar, exactly? And once I do that, then what - is the result in lieu of CEntrance's analysis or something I use with CEntrance's?

Thanks again.

Before trusting the Centrance results, I would check the loopback in recording software like Sonar

+1

If the CEntrance-reported round-trip time changes from one test to the next at the same buffer setting, then either CEntrance is mis-detecting the onset of the returning ping, or there's something wrong with the the Mackie's driver that is producing variable latency. If you do several manual loopback tests back-to-back, and you get an inconsistent offset (with ASIO reported latency compensation enabled), that would tend to point to the driver as the problem. But if you get a consistent offset in SONAR, then you can just use that value and ignore whatever problem CEntrance is having.

Please remove the "SOLVED" from the thread title and replace it with "BAND-AIDED"!

This is NOT SOLVED!

When Sonar detects a new audio or MIDI device in your system, it should run it's own automated loop-back tests at all buffer settings and at least a few system stress levels. It should save these results to a table and you should never have to worry about it!

The two most fundamental parts of music are time and frequency. It's just absolutely wrong that we can't can't depend on the timing of our DAW! Doesn't this seem obscene to anyone?

And I can't believe that after many many versions of complaining, we can't depend on the metronome to be in time. It's not like some feature isn't working correctly, this is FUNDAMENTAL to recording music!

ORIGINAL: evansmalley
I think I saw that the newest upgrade to 8 is an improved ping test.

Are you talking about the Sonar 8.3 Announcement? And I quote "The "ping" test has also been improved to consistently return sample accurate offsets without any further adjustments."

Uh. Where is the ping test now?


Well if it's a full implementation this time, I'll be very relieved. I'm to the point of buying a Mac so I can run Logic and see if that's any better. I'm happy with everything but all the timing issues in Sonar. I know it's driver related. But I keep hearing that the compensation is much better in other DAWs.


For General Background...

The "Nudge" feature was introduced in S4 because of this issue, reported here on the forum. That was a huge band-aid, but at least it had 3 nudges.

Recording Latency Adjustment introduced in S6 was close to a fix, but it still fell way short. It needs to be input/output type related AND driver type AND buffer setting related. AES and SPDIF I/O's can have different loop-back latencies than and analog I/O's on the same sound card. Some sound cards will have different latencies depending on the buffer setting, and generally have different latencies between driver types. The fact that the Nudge was introduced with 3 Nudges hints that Cake knew all this. When it was announced that the Recording Latency Adjustment would be introduced, we pointed this out to Cake and they seemed to acknoledge this.

Users simply shouldn't have to know the gory technical details about all this to get tight recordings. If it's not tight, it should be that the user can't play in time, not because they didn't know that their subharmonic nutrinizer had to be used to calibrate the dilithium crystals to localized temporal drift of their sound card -- unless it's a full moon, of course.

ORIGINAL: evansmalley
But all you can do for now is a good "ping test" for your own system and compensate.

BTW, there is an article in the Wiki about all this.

I personally wouldn't use CEntrance to do the test. It may not account for everything Sonar does to the signal. In any event, make sure you test your final settings in Sonar with a loop-back test after you enter the number in Sonar's "Recording Latency Adjustment". It's really great to see it line up to the sample!


Also, if you are also connected digitally to anything (AES, SPDIF) try doing loop backs with that -- it will prolly be different. For instance, I use a POD. Loopback chain: Playback from Sonar -> sound card Analog out -> POD Guitar In -> POD AES out -> sound card AES in -> to record in Sonar. (Use clean setting on POD).

Of couse that's for playing along with tracks already recorded. If you are re-amping digitally the loop-back chain would be: Playback from Sonar -> Soundcard AES -> POD AES in -> POD AES out -> Soundcard AES in -> Record in Sonar.

and so on....

It's not really practical to make delay compensation completely automatic without some kind of loopback, because SONAR can't automatically tell a given I/O combination in advance from all the possible I/O combinations out there. But some steps can be taken toward automation and management of this. A reliable automated loopback test is a great first step (as a Wizard, GUI, or both). The next step would be to enable saving of the measured compensations, as a kind of setup-preset with an associated name. That way, users could save multiple presets and call then up as needed when same or similar setups are used. As a convenience, presets could be assigned as defaults, or set to come up as defaults when certain inputs or outputs are used (helped further by having the named inputs and outputs).

some steps can be taken toward automation and management of this. A reliable automated loopback test is a great first step (as a Wizard, GUI, or both). The next step would be to enable saving of the measured compensations, as a kind of setup-preset with an associated name. That way, users could save multiple presets and call then up as needed when same or similar setups are used. As a convenience, presets could be assigned as defaults, or set to come up as defaults when certain inputs or outputs are used (helped further by having the named inputs and outputs).

yes- both you and Xav are exactly right on

I have not recently and never to that detail. I'll submit one after checking 8.3 out. Cake has been know to take our feature requests to the next level on other things. Maybe they've finally one-up'ed us on this one too?

Now I'm wondering if we need to do some sort of frequency test. I have noticed sometimes that tracks seem more out of tune than when I recorded them

ORIGINAL: Xavier

Now I'm wondering if we need to do some sort of frequency test. I have noticed sometimes that tracks seem more out of tune than when I recorded them

Are you serious, or just being sarcastic? Sorry, I can't tell here :-)

ORIGINAL: Xavier

ORIGINAL: Jose7822

ORIGINAL: Xavier
Now I'm wondering if we need to do some sort of frequency test. I have noticed sometimes that tracks seem more out of tune than when I recorded them

Are you serious, or just being sarcastic? Sorry, I can't tell here :-)

To the best of my knowledge with emoticons, a smile with a wink ;-) or is the best one to use for sarcastic joking. Is there a better one for that?

Everyone uses emoticons in different ways which is why I wasn't sure about how you meant that. I personally would've used this one for sarcastic joking. Thanks for clarifying though.

Take care!

unticked the box right beside the manual offset box - the one that says, "USE ASIO Reported LATENCY". 

I think what happened is you calculated your manual offset based on numbers obtained with Reported latency correction disabled. So you built the reported latency into your offset. The problem with this is it will change with the buffer setting, but your manual offset won't.


You need to redo the offset measurement with ASIO Reported Latency compensation enabled. Depending on your interface, you should end up with a manual offset on the order of 10s of samples. I'm guessing you're in the hundreds now...?






 

Okay - sounds good.  Now all I have to do is find my aud.ini file.  I've had no luck in locating it even with a search?  Is it still called that in version 8.03?  What am I missing here.  I want to delete it and rebuild/reprofile.  I found my aud.ini file in previously installed versions 5.0 and 4.0, but even after renaming them, I now cannot locate any other version of an aud.ini file.  Am I losing it?

Thanks...(I'll repost this if I don't hear back on this thread).

Martin B.

I'm using Home studio 7 XL,
and when I look at the advanced tab in the audio window, there is no such thing as a "reported latency" number at the bottom right hand side of the screen. Can anyone help?
Thanks.

This is a very old thread,, you need to start your self a new thread with the proper topic etc... You are also in the wrong forum. There is a seperate forum for Home studio, not that it matters much, they pretty close to the same anyhow. But the people there will now which features are included. Sonar has more stuff to it.
It also helps if you list your system specs and which OS and interface you are using.