Old 03-22-2007, 09:54 AM   #1
charles.monteiro
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Default calibrating for latency

hi, so what's the state of affairs now with latency? by this I mean now in the world of Core 2 Duos, plenty of ram etc. Are we now in a situation that latency is basically neglible to the point that it has no impact on timing. More concretely, if recording an audio signal i.e. a guitar is the audio that gets captured perfectly in relation to a monitored audio track? Even more concretely I have drums on track 1 - BFD at that i.e. midi, I'm recording guitar on track 2 which undoubtedly will have some effects, will the captured recording be a truested snapshot of the performance? Put it this way are now analog and digital basically on an even keel with regards to faithfully reproducing a performance. Again, by that I mean on the latest hardware Core 2 Duo 2.66Mhz, 2 gig ram, 10k rpm drives and a basically very new DAC i.e. an Edirol FA-101.

My issue is whether a slighly lagging of time is possible by the system , being able to determine when its happening if it is and being able to appropriately correct it.

thanks in advance,
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Old 03-23-2007, 08:55 AM   #2
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ok, C'mon guys, somebody? Or did I word this too confusingly?
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Old 03-23-2007, 09:40 AM   #3
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Originally Posted by charles.monteiro View Post
ok, C'mon guys, somebody? Or did I word this too confusingly?
I tested my system's latency by sending a previously recorded track out (output #1), then patched it back into the sound card via an input (input #3), armed a track to record the #3 input, and then recorded several minutes worth of the original track. I then compared the two tracks WAV files and they were spot on, down to the highest level of precision that I could get in Reaper. I have Reaper set to auto-adjust for latency when recording.

How's that?

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Old 03-23-2007, 09:55 AM   #4
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I just had a situation where I went from an Athlon 2400 (2G), with 1.5G RAM to a Athlon 64 X2 5000+ with 2 G RAM.

I thought I would see a huge improvement in latency. I didn't. In fact it changed very little.

So whatever latency I'm seeing, I have to assume it's in my interface.

I can correct for it in Reaper, and everything lines up perfectly.

The extra horsepower, CPU and RAM wise, allows me to run lots of tracks and effects.
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Old 03-23-2007, 10:17 AM   #5
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Quote:
Originally Posted by mr. moon View Post
I tested my system's latency by sending a previously recorded track out (output #1), then patched it back into the sound card via an input (input #3), armed a track to record the #3 input, and then recorded several minutes worth of the original track. I then compared the two tracks WAV files and they were spot on, down to the highest level of precision that I could get in Reaper. I have Reaper set to auto-adjust for latency when recording.

How's that?

-mr moon
fantastic and I got to learn about "auto-adjust" to boot,
also it seems that our boxes are very similar in setup i.e. cpu , even the Corsair, are those the Dominators, right? the only diff is that I have an Edirol FA-101 which is still standard Firewire i.e. 1394 and you have the new Firewire 800 which is apparently twice as fast. I am also on Win XP Pro 32 bit, I'm a bit scared of going into 64, don't think Audition works there but Reaper certainly does

thanks again, didn't think about testing for latency in the way you described.
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Old 03-23-2007, 10:19 AM   #6
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Quote:
Originally Posted by jimst57 View Post
I just had a situation where I went from an Athlon 2400 (2G), with 1.5G RAM to a Athlon 64 X2 5000+ with 2 G RAM.

I thought I would see a huge improvement in latency. I didn't. In fact it changed very little.

So whatever latency I'm seeing, I have to assume it's in my interface.

I can correct for it in Reaper, and everything lines up perfectly.

The extra horsepower, CPU and RAM wise, allows me to run lots of tracks and effects.
I guess by correcting you mean using the "auto-adjust" in Reaper i.e. like Mr. Moon does? Sorry for being lazy, I guess I would find that somewhere under preferences. I'll look when I get home.

thank you very much for the feedback.
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Old 03-23-2007, 10:33 AM   #7
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Originally Posted by charles.monteiro View Post
I guess by correcting you mean using the "auto-adjust" in Reaper i.e. like Mr. Moon does? Sorry for being lazy, I guess I would find that somewhere under preferences. I'll look when I get home.

thank you very much for the feedback.

It's under preferences,audio,recording. There is a check box for "Auto-Adjust" or you can uncheck it and enter a value manually.

I have have to do the manual method because with my interface, a Digidesign MBox2, the auto-adjust does work correctly. It seems to work for some though from what I read.
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Old 03-23-2007, 11:06 AM   #8
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Originally Posted by jimst57 View Post
It's under preferences,audio,recording. There is a check box for "Auto-Adjust" or you can uncheck it and enter a value manually.

I have have to do the manual method because with my interface, a Digidesign MBox2, the auto-adjust does work correctly. It seems to work for some though from what I read.
so how did you figure what to enter? Just trial and error, or something more mathematical?
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Old 03-23-2007, 11:11 AM   #9
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Mr. Moon,

I guess that to use your RME you went ahead and purchased a 1394b PCI card? I noticed that your board comes with a 1394a jack.

I have a:

Asus P5W DH Deluxe Wireless Edition with Remote Core 2 Intel 975X Chipset 1066MHz FSB Dual Channel DDR2 800 LGA775 Socket 775/T ATX Motherboard w/ ATi Crossfire 2x PCI Express x16, Audio, Dual GB LAN, SATA 3G/II, eSATA, RAID, USB 2.0, Firewire 1394

so I'm on the same boat.

Unless , you just went straight thru but in that case you would still be running 400 speeds.

Pls, let me know. thanks
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Old 03-23-2007, 11:15 AM   #10
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Originally Posted by charles.monteiro View Post
fantastic and I got to learn about "auto-adjust" to boot,
also it seems that our boxes are very similar in setup i.e. cpu , even the Corsair, are those the Dominators, right? the only diff is that I have an Edirol FA-101 which is still standard Firewire i.e. 1394 and you have the new Firewire 800 which is apparently twice as fast. I am also on Win XP Pro 32 bit, I'm a bit scared of going into 64, don't think Audition works there but Reaper certainly does

thanks again, didn't think about testing for latency in the way you described.

I'm just using a standard 1394a card, as RME states that you really don't see a throughput difference using 1394b (800) unless your daisy chaining FF800's or FW drives together.

An interesting aside: ...Reaper's auto-adjust works for me without having to add any sample amounts manually, whereas I have to add 96 samples to the auto-adjusted amount in SONAR to get them aligned correctly when running the same tests.

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Old 03-23-2007, 11:21 AM   #11
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Quote:
Originally Posted by charles.monteiro View Post
so how did you figure what to enter? Just trial and error, or something more mathematical?

Loopback test as someone explained above and then trial and error until they line up. It can take some time.
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Old 03-23-2007, 08:24 PM   #12
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Quote:
Originally Posted by mr. moon View Post
I'm just using a standard 1394a card, as RME states that you really don't see a throughput difference using 1394b (800) unless your daisy chaining FF800's or FW drives together.

An interesting aside: ...Reaper's auto-adjust works for me without having to add any sample amounts manually, whereas I have to add 96 samples to the auto-adjusted amount in SONAR to get them aligned correctly when running the same tests.

-mr moon
really ? sorry, am I being dense but then am I correct in assuming that my Edirol Firewire interface which is 1394a based and which connects to my ASUS motherboard's 1394a jack has the same throughput?

What I'm wondering is whether I should go for a Firewire 800 setup which I figured would require not only getting a FW 800 DAC which btw, the RME seems to be the only one , but also getting a 1394b i.e. FW 800 PCI card.

thanks
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Old 03-23-2007, 08:38 PM   #13
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Quote:
Originally Posted by mr. moon View Post
I'm just using a standard 1394a card, as RME states that you really don't see a throughput difference using 1394b (800) unless your daisy chaining FF800's or FW drives together.

An interesting aside: ...Reaper's auto-adjust works for me without having to add any sample amounts manually, whereas I have to add 96 samples to the auto-adjusted amount in SONAR to get them aligned correctly when running the same tests.

-mr moon
The only reference to "auto adjust" that I seemed to find is here:

http:///www.monteirosfusion.com/it/r...ustLatency.JPG

Is that what we are talking about? So, therefore auto adjust means just don't specify a thing.
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Old 03-23-2007, 09:12 PM   #14
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Its here, down the bottom...

[IMG]http://img261.**************/img261/8437/adjustrecoa3.png[/IMG]

Ticking the box enables it, the value sets how much. I posted a little guide here.
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Old 03-23-2007, 09:22 PM   #15
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Originally Posted by Billoon View Post
Its here, down the bottom...

[IMG]http://img261.**************/img261/8437/adjustrecoa3.png[/IMG]

Ticking the box enables it, the value sets how much. I posted a little guide here.
thank you very much, quite helpful. BTW, you all may find this interesting:

http://www.centrance.com/products/ltu/

Its a utility to measure latency which basically does a loop back test. I'm actually in the middle of playing around with it. My first test resulted in 36ms latency if I read that correctly. That does not seem very good . I'm now going to play changing my Asio buffer sizes to see if I can get it down. I'll report back for those that may find this interesting.
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Old 03-23-2007, 10:08 PM   #16
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So here are the results of my test using the Centrance Asio Latency test utility. As instructed by Centrance I connected one of the physical inputs of my audio interface to one of its physical outputs. I used standard 1/4 plug cables , not balanced. Using the min buffer settings for the Edirol ASIO driver I got the following:



which looks better but now I need to test whether I get the pops and clicks. I do have a monster box so I'm hoping no. Now, 15 ms - 36 ms significant enough to affect the player's performance timing? Not only do I need to have what I play recorded when I play it but I need to be able to hear what I'm playing and whatever other track I'm monitoring while recording. Calibrating latency deals with "all" of the above, right?
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Old 03-23-2007, 10:15 PM   #17
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Billoon,

What about using Reainsert to do the loopback test for you?
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Old 03-23-2007, 10:42 PM   #18
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Originally Posted by charles.monteiro View Post
Calibrating latency deals with "all" of the above, right?
Should,...as long as you monitor using hardware. If you software monitor, there will be some latency that you may unconsciously compensate for in your playing...this would make the recorded file out of place.

BTW...direct monitoring will help with that, its being worked out ATM, check the FR section if youre interestred in having a say.

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What about using Reainsert to do the loopback test for you?
Not sure that would work, id just use the loopback test.
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Old 03-23-2007, 11:01 PM   #19
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Billoon, I'm confused about...
Quote:
Ticking the box enables it, the value sets how much. I posted a little guide here.
I would have expected that ticking the box would turn on auto and thus the manual figure entered in the box would therefore be ignored. You're saying that it's the other way round, it seems. I guess one could test by putting a silly figure (10000 samples?) in the box, leaving the auto box ticked, and seeing if things get out of sync.

Last edited by Art Evans; 03-23-2007 at 11:17 PM.
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Old 03-23-2007, 11:17 PM   #20
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Hmmm - doing some playalong tests here, putting a figure of 10000 in the box adjusts latency on replay regardless of whether the auto box is ticked.

I got the best correlation between what I recorded on track two while listening to the playback of track one when I had a figure of zero in the box and auto unticked.

Now I'm confused.

My soundcard is set to a buffer size of 512 samples and I had no issue with playing along with myself accurately (or as accurately as I'll ever get it).
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Old 03-23-2007, 11:18 PM   #21
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Quote:
Originally Posted by Billoon View Post
Should,...as long as you monitor using hardware. If you software monitor, there will be some latency that you may unconsciously compensate for in your playing...this would make the recorded file out of place.

BTW...direct monitoring will help with that, its being worked out ATM, check the FR section if youre interestred in having a say.



Not sure that would work, id just use the loopback test.
I guess that may have to deal with this blurb from the Edirol manual:


Direct monitor volume (direct monitor section)
This adjusts the monitor volume.
Turning this knob toward the left (OUT 1/2) will decrease the level of the sound being input through the input jacks. Turning the knob toward the right (MONITOR) will decrease the level of the sound being output from your computer. At the center position, the sound being output from your computer and the
sound being input via the input jacks will both be at 100% of their level.


Ok, I admit to not understanding the diff between hardware monitoring and software monitoring, forgive my newbie-ness. I monitor via headsets that are directly connected to my Edirol interface.

The direct monitor is also controlled by the following:

10. Direct monitor soft control switch (direct monitor mixer section)
Turn this on ( pressed inward) if you want to control the direct monitor mixer (see block diagram
➔frontcover) from your ASIO 2.0 compatible software. If this is on, your software will be able to control the monitor volume, pan, and on/off status of each input jack.
If this is off, all settings of the direct monitor mixer will be ignored (= bypassed), allowing you to monitor the input signals from all input jacks. This is convenient if you want to check the connections while temporarily ignoring the software settings.
* This function is available only on Windows.


Is the "software controlled" above what you mean by software monitoring?

and thank you for your patience.
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Old 03-23-2007, 11:26 PM   #22
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Billoon, I'm confused about...


I would have expected that ticking the box would turn on auto...
EDIT: Yes turning on "Auto adjust" does automatically adjust using the reported latency...but if the reported latency isnt correct you can adjust it by using the manual offset.

...or you can leave Auto Adjust off and just correct the total recording latency manually.

See the Loopback project below in Post 27.

Last edited by Billoon; 03-24-2007 at 08:13 PM. Reason: I stuffed up.
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Old 03-23-2007, 11:32 PM   #23
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Originally Posted by charles.monteiro View Post
Is the "software controlled" above what you mean by software monitoring?
Thers 2 types...the one we have currently, the audio input goes through REAPER and any plugins on the track...being delayed by any latency introduced by the FX.

and Direct Monitoring, which doesnt go through the tracks/FX...it basically patches the in to the out of your soundcard, bypassing the software(REAPER), so is quicker.

CockOS are looking at adding the option of allowing REAPER to "control" the Direct Monitoring of your soundcard from within REAPER.
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Old 03-24-2007, 06:26 PM   #24
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Hi all,

First of all thanks for all the help.

I performed the loopback test as directed, here's a pic:



To review , I did the following:

#Loaded track 1 with a drum loop
#Directed Track 1 to output to my DAC's output #3
#Setup Track 2 to record and setup its input to my DAC's input #3
#Physically connected my DAC's output 3 jack to the DAC's input 3
#Went ahead and recorded

Also I had turned off "auto adjust"

Conclusion:

Configuring my DAC to use the smallest buffer possible the system reports a latency of 3ms or approx 160 sample offset. Not sure if I will be able to get away with using the smallest buffer setting but again I have a pretty beefy machine... confused as to whether it matters that much in this case.

I then went ahead and measured what "auto adjust" did and I still got an offset which I measured by using markers and subtracting the reported positions. This resulted in an offset of 224 samples.

So I'm going to repeat the experiment with again and measure my offset with auto adjust off and see if actually specifying the offset and NOT using auto adjust will give me results closer to dead on.

I have zipped up the project in question including the base wav file used in case somebody wants to check it out. Its in .rar format which I hope is not too inconvenient.

Here it is:

http://www.monteirosfusion.com/it/re...alibration.rar

Last edited by charles.monteiro; 03-24-2007 at 06:50 PM. Reason: forgot to mention offset value
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Old 03-24-2007, 07:26 PM   #25
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update,

with "auto adjust" off and setting a manual offset of 615 samples, I got the two wav files to match up exactly.

I think I have the hang of this now
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Old 03-24-2007, 07:54 PM   #26
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update,

with "auto adjust" off and setting a manual offset of 615 samples, I got the two wav files to match up exactly.

I think I have the hang of this now
From what I understand it's up to the driver to report the correct latency information to the host. However, there must also be some internal processing that each host application must do on it's own, as Reaper records perfectly for me with auto-adjust enabled, whereas Sonar remains 96 samples off with Sonar's auto-adjust enabled (same system, FF800, and everything).

Anywho, I'm glad to hear you got it figured out!

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Old 03-24-2007, 08:03 PM   #27
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OK, i got it wrong before..."Auto adjust" does in fact automatically adjust...but the manual offset is for correcting if its not 100% accurate.

EG. With "auto adjust" on...its still out of sync by 23 samples, so put 23 in the box and it syncs properly.

...or leave "Auto Adjust" off and just use the manual compensation for the full amount.

Sorry for the confusion. Anyway, try this project...

Loopback test

1)Setup the i/o on the Master so it goes out 3/4(for your setup).

2)Setup track 2 so it inputs from 3.

3)Make sure Auto Adjust is selected and press record.

4)Draw a loop from the start of the sample on tk1 to the start of the recorded sample on tk2....with the timeline set to samples.

5)Put the loop length value in the manual offset box.

...and record again...should sync properly now.

For this to work...the click has to go out through the Master track and come back in a soundcard input...not using any internal routing.

Last edited by Billoon; 03-24-2007 at 09:11 PM. Reason: Fixed link.
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Old 03-24-2007, 08:50 PM   #28
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billoon,

when I try to click on your link for the "Loopback" project , I get the following:

Quote:
Invalid Attachment specified. If you followed a valid link, please notify the administrator
I also can't just do a "save as" on the link.

The way I figure it, if "auto adjust" is going to be off i.e. at least in those cases when it is I might as well use the manual offset

I'll try your project out when you have a new link.

thanks again,
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Old 03-24-2007, 09:10 PM   #29
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OK, try again.

Loopback test
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Old 03-25-2007, 03:02 AM   #30
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Default Loopback test recording a miked guitar amp

Hey all,
After seeing this thread I had to try the test. My last session with R I was using midi drums and bass. Then recording a miked guitar amp. Now I thought I was just having a bad day. Because I couldn't seem to get my timing right. My notes where pretty hit and miss. I pride myself on knowing how to "sit" on a beat, but I was all over the place. So I put down my guitar, ashamed of my horrible playing.
Now after reading this thread I thought maybe there was some latency in the signal path that was throwing me out. So I took a line out of my headphone amp (as a reference point of where I was hearing the beat) and fed that into my guitar amp (valve), SM75, to the mixer, into the 2496 sound card and reaper. This seemed to be the best way of approximating the latency in my hearing the midi beats from reaper and then having my ears/brain/fingers respond back through my guitar/mic/amp/mixer/soundcard.
Well after an hour of farting around I've found that I have a latency of around 4000 frames. Setting this manually without the check box checked. What really threw me out was the Clav (doink sound) which has a very short lead in, was not getting picked up by reaper when the latency was adjusted correctly!!!
This only became evident when I suspected something was going on and looped the Clav x 4. You can see from the attachment that there is little or no latency in the two items but the first beat is missing? Why is that I wounder. Anyway I would be very interested to hear if anyone else has had problems mixing midi with audio and had latency issues. Does 4000 frames seem like alot? Many thanks.
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File Type: jpg loopbacktest.jpg (73.0 KB, 236 views)
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Old 03-25-2007, 06:57 AM   #31
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just because things seem to jive visually at one zoom level does not mean that there's no latency, so I trust that you zoomed in enough to know.

Secondly, could not help but notice that the loopback track was in mp3 format. Not sure if it makes a difference but to me it does not make for a "controlled" experiment.

I'm using BFD for drums i.e. midi. In that case a midi track is triggering the midi instrument, that should be immediate , not much data flying thru, then there's the output of the instrument back into the mix, so that would seem to be an issue with monitoring. It would be interesting to understand if there was any latency between the moment the trigger happens and the midi instrument's output hits your ears. Another important aspect of making sure things are "jiving" sonically. Don't believe that the current experiments we have been doing would be appropriate for this. Or would they? Instead of a recorded tick, maybe a midi file with a tick event and have the midi instrument's output routed to record?
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Old 03-25-2007, 07:35 AM   #32
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billoon,

update:

used your project file which you very kindly setup and with "auto adjust" on I still had to put in a manual offset of 218 to make things jive perfectly.

Also with no "auto adjust" a manual offset of 598 does the trick.

I used a third party wav editor (Audition) to actually check out the wav files since its easier to zoom to the very individual sample level.

thanks again
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Old 03-25-2007, 10:53 PM   #33
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Cool, im glad we finally got it worked out.
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Old 04-09-2007, 01:26 AM   #34
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Found this all very interesting, so i just had to check it out on my new system. The soundcard was reading 1ms, but it was a little higher, as the image shows...

[img]http://img366.**************/img366/2568/testresultssk0.png[/img]

Thanks for all the above info guys. Love this place, and REAPER is the dogs bits
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