Short answer because it's late:

No echo problem unless you do both direct and input monitoring. I only do one or the other, usually input monitoring. I used 100ms in my testing to make the effect more obvious without having to count samples, but I think it might be even more valid at lower latencies where the performer is not consciously aware of compensating.

Interesting stuff, in any case.

You have both done my head in. I have read it twice but I'll go over everything again.

I am going to try to summarize this in a few key points.

1) My 80ms recording offset should be kept as is?
2) I never use hardware direct monitoring as i ALWAYS need to monitor through an effect (a zero latency reverb of course,, so the monitoring will be done via sonar's software mixer). The vocaist should theoretically just try to sing as well as he can with what he is hearing to get in the best time he can get.
3) The WDM latency MIGHT have hidden buffers we dont know about. i.e. 128 sample buffer in asio is 336 round trip in the audiophile case as confirmed by ceentrance. But there is no way to know what the REAL roundtrip of the same buffer size (128) is with WDM, right? It could be double, I mean, we don't know, cause there is no way to test it, other than feel. I can't really feel a difference so I am guessing it's the same.
4) Even if i was going to bypass input monitoring i don't even know HOW to do direct monitoring with WDM.. is this an asio specific feature anyway?
5) And to Jose when you wrote "In ASIO mode, most but often not all of the A/D conversion time is included in the "ASIO Reported Latency" that Sonar uses, which is why you usually need some additional offset to get perfect compensation"
Ok this is weird. The asio reported latency is 116 and i still have to adjust it a further 130 ish to get it to line up. In WDM 80 does the trick!!

Anyway, maybe I will just not worry about it, record some music LOL. The only thing that concerns me is determining the true roundtrip latency of wdm. The guy that wrote ceentrance should have added that option!

ttoz said:

You have both done my head in. I have read it twice.

ditto that!
So please help me understand this if you can.
Just a note here, I always monitor from my HARDWARE mixer, never from Sonar, I can't deal with the delay.

Sonar plays the existing song file (BTW, I ALWAYS bounce a stereo "tracking" mix and archive everything else, hold over from the ProAudio days I guess).

This audio is delayed by the trip through the AD converter, AND any buffers that Sonar & the interface card are using.
I play along to this track, AS I HEAR IT, and this perfromance is delayed by the DA converter, AND again, any buffers that Sonar & the interface card are using.

Are you with me so far? Do I GET the basic concept?

Now, I have added an 82 sample compensation in the Sonar Audio control panel (as per the tests outlined) and my over dub should line up with my existing tracks, right?
Provided of course, that I am able to play in time, etc, etc, etc.
Do I still "GET IT"?

SO, WHY does the TOTAL round trip even matter? If Sonar handles that part, and all I have to do is compensate for the AD/DA, I'm good to go, RIGHT?

I would think the total round trip WOULD affect the performance that is being monitored IN Sonar, Right?
In that case you can't hear the over dub until it clears the AD converter AND any buffers; hidden or not, set in latency, etc. (Exactly why I DON'T monitor the performance in Sonar, ALWAYS in the outboard mixer).

Do I get it??

Thanks,
Tom

ORIGINAL: ttoz

You have both done my head in. I have read it twice but I'll go over everything again.

I am going to try to summarize this in a few key points.

1) My 80ms recording offset should be kept as is?

Yes, keep it as is.

2) I never use hardware direct monitoring as i ALWAYS need to monitor through an effect (a zero latency reverb of course,, so the monitoring will be done via sonar's software mixer). The vocaist should theoretically just try to sing as well as he can with what he is hearing to get in the best time he can get.

Right.

3) The WDM latency MIGHT have hidden buffers we dont know about. i.e. 128 sample buffer in asio is 336 round trip in the audiophile case as confirmed by ceentrance. But there is no way to know what the REAL roundtrip of the same buffer size (128) is with WDM, right? It could be double, I mean, we don't know, cause there is no way to test it, other than feel. I can't really feel a difference so I am guessing it's the same.

In your case, Sonar does a very good job compensating for latency in WDM mode because all you're having to adjust is the 80 samples from the converters. So that right there tells me that Sonar is indeed compensating for latency in WDM mode as there's no way your roundtrip latency is only 80 samples at 128. But it's really hard to know if there's anything else being compensated for because of the way different drivers reports latency to Sonar.

4) Even if i was going to bypass input monitoring i don't even know HOW to do direct monitoring with WDM.. is this an asio specific feature anyway?

AFAIK, yes.

5) And to Jose when you wrote "In ASIO mode, most but often not all of the A/D conversion time is included in the "ASIO Reported Latency" that Sonar uses, which is why you usually need some additional offset to get perfect compensation"
Ok this is weird. The asio reported latency is 116 and i still have to adjust it a further 130 ish to get it to line up. In WDM 80 does the trick

Like I said above, it just depends on how each driver was written. Some manufacturers have better WDM drivers while other have better ASIO ones. I just happen to know what RME drivers do because they tell you exactly how they work in the manual (which isn't usually the case with other manufacturers).


HTH

ORIGINAL: DeeringAmps

ttoz said:

You have both done my head in. I have read it twice.

ditto that!
So please help me understand this if you can.
Just a note here, I always monitor from my HARDWARE mixer, never from Sonar, I can't deal with the delay.

Sonar plays the existing song file (BTW, I ALWAYS bounce a stereo "tracking" mix and archive everything else, hold over from the ProAudio days I guess).

This audio is delayed by the trip through the AD converter, AND any buffers that Sonar & the interface card are using.
I play along to this track, AS I HEAR IT, and this perfromance is delayed by the DA converter, AND again, any buffers that Sonar & the interface card are using.

Are you with me so far? Do I GET the basic concept?

So far so good.

Now, I have added an 82 sample compensation in the Sonar Audio control panel (as per the tests outlined) and my over dub should line up with my existing tracks, right?
Provided of course, that I am able to play in time, etc, etc, etc.
Do I still "GET IT"?

Correct!

SO, WHY does the TOTAL round trip even matter? If Sonar handles that part, and all I have to do is compensate for the AD/DA, I'm good to go, RIGHT?

I would think the total round trip WOULD affect the performance that is being monitored IN Sonar, Right?
In that case you can't hear the over dub until it clears the AD converter AND any buffers; hidden or not, set in latency, etc. (Exactly why I DON'T monitor the performance in Sonar, ALWAYS in the outboard mixer).

Do I get it??

The total rountrip latency matters because it tells you exactly what you're system latency is which then helps you figure out the offset you need. It all depends on how your drivers report latency to Sonar, so some will be more effective than others and the only way to really know this is through a roundtrip latency test. There might be hidden buffers included that you didn't even know about so just compensating for the AD/DA might not be enough.

Even if you monitor directly you still get the playback latency. The music is stored in your HDD in digital form, so it has to go through the playback buffers, any added hidden buffers (if applicable) and also the D/A conversion latency. The problem comes when you monitor through Sonar while recording because you're now adding the second half of the roundtrip which is the latency from the A/D converters and the record input buffers. IOW, when you monitor directly you only hear a one way trip on playback with only the converters latency on the monitored instrument while when you monitor through Sonar you hear a one way trip on playback but a roundtrip on the monitored instrument (since it goes in and out). Does this answer your questions?

The problem comes when you monitor through Sonar while recording because you're now adding the second half of the roundtrip which is the latency from the A/D converters and the record input buffers. IOW, when you monitor directly you only hear a one way trip on playback with only the converters latency on the monitored instrument while when you monitor through Sonar you hear a one way trip on playback but a roundtrip on the monitored instrument (since it goes in and out).

OK...I see.
Not applicable to my setup.
Thanks.

ORIGINAL: brundlefly

You have both done my head in. I have read it twice but I'll go over everything again.

My head was hurting, too, so I went for a moto ride to clear out the cobwebs.

Yeah I know what you mean. It would be so much easier to explain things with pictures or, better yet, in person.


Take care!

ORIGINAL: vanceen

I just went through this with my Delta 1010. I got a 72 sample offset. Now it lines up perfectly, whatever the latency setting.

Just for anyone else who may try this with a Delta 1010, I found that I had to invert the phase of the input to get the waves to match up. Otherwise, the phase gets reversed on the "second" recorded track, and comparing the timings is a bit trickier.

Another thing I found interesting is that the samples, even when lined up perfectly, show small but definite differences. I guess this shouldn't be surprising, as the D/A - A/D process can't be perfect. I wonder if the differences would be smaller with a Lynx or an RME?

I just tested my 8 year old Delta 1010 - 70 sample offset (1.6ms @ 44.1k). Thanks for the tip on the phase reversal.

My problem is that im recording midi drums from battery 3 and the timing sounds pretty on time when recording but when i playback it is way earlier/faster than everything else in the song. even some hits are so far off it sounds like a baby was druming lol. how do i fix that? ive tried changing the latency but that just makes the timing of me pressing the key and hearing the sound a bit delayed therefore completely throwing me off. im using a wdm driver and i have sufficient space and power on my pc any suggestions

my settings in options>>>audio are: Buffers in playback que=2
Buffer size=40.0 msec
then it says efective latency at 44kHz/stereo $0.0 msec and my sample rate is at teh default 44100

are these common settings or is the latency too low i mean its at teh owest i can get it without getting clicks and pops or dropouts but for some reason its n time when recording but on playbac its slightly ahead and some of the notes are way out of timing and trust me its not that im a bad drummer lol its happenind =g with all instruments and im using moslty plugins and vst. nad the funny thig is my audio is perfect i just did a session yesterday and it came out with no problems. HELP PLEASSEE LOLOL

im sorry lol the latency was 40 msec lol i didnt even notice that anyway. i completely forgot to tell you what i am using and what sound card i have. im using sonar 7.02 i have a soundblaster live and an m audio 88 keystation es controller. so checvk this out i had on my windows volume mixer muted and i unmuted it and recorded some drums (the recording process never had any delays or problems) and i played it back and it was exactly the way i recorded it could that have been the problem. or am i halucinating because ive nbeen trying to work this out for about 8 hours now. it seems like its working now i mean the drums are on point and they miss in certain spots just because im not the best drummer but it actuall sounds humaine and not a mechanical quantise gone wrong sound lol.

The manual offset is stored in AUD.INI as InputLatencyOffset. If your AUD.INI got reset/replaced somehow, the offset would probably default to 0. Never seen mine change spontaneously.

I don't think so. I changed to WDM mode and back to ASIO to see if that would do it, and the manual offset was restored to its original value, even though Sonar ran the wave profiler when I switched to WDM.

this is very frustrating.

Are you saying this is happening regularly that SONAR is losing the offset you enter?

Ah, I see. What was the setting? Mine's only 22 samples; wouldn't be a big deal if I recorded without it. MIDI delay and jitter dwarf the manual offset on my system.