Audio not in sync

[ski]

How much of a level difference was there?

(BTW, if you want to adjust level in really fine increments -- like .1 dB at a time -- use the gain plug's gain slider, holding down shift before you click on itl; that lets you get .1dB adjustments. You can't get this kind of fine resolution from the channel strip's fader).

[ElMariachi78]

:shock:

How much of a level difference was there?

(BTW, if you want to adjust level in really fine increments -- like .1 dB at a time -- use the gain plug's gain slider, holding down shift before you click on itl; that lets you get .1dB adjustments. You can't get this kind of fine resolution from the channel strip's fader).

I am not sure, i just saw a difference in the waveforms , must have been between 3 - 6 db , enough that the phase cancelation didnt work properly, sound was getting thinner but no cancelation.

[darkecho]

I am trying to do this, first of all I am using the duet, so I have a variety of in/out options (from output to -10 or instrument amp) then I have inputs of (mic xlr/+4xlr/-10xlr/instrument)

I thought i'd better not use the xlr ones as they go through the preamp and I want to avoid coloration for this test right? so I chose instrument input, then I used some TR or TRS cables to patch the output to the instrument input and in the apogee software I set the output to -10.

I recorded and got a replication but it was quieter, so I had to mess with gain (added 4 db of gain on both instrument input channels to get something SIMILAR to the original in amplitude) I cant get them to cancel completely but get close, and its only one sample space to the right. is this normal? why cant I get a perfect cancellation?

[ski]

I am not sure, i just saw a difference in the waveforms , must have been between 3 - 6 db , enough that the phase cancelation didnt work properly, sound was getting thinner but no cancelation.

It's hard to tell without being there to know exactly why you had such a difference in level and why you were only getting partial cancellation. BUT... if the sound was getting thinner, that's a good sign. Chances are that you're better off having set your recording delay based on however many samples of offset it took to achieve the thinning of the sound than what you had before.

[ski]

I cant get them to cancel completely but get close, and its only one sample space to the right. is this normal? why cant I get a perfect cancellation?

Can you explain in a little more detail what you meant by "it's only one sample space to the right"?

[SWAN]

Hi There-interesting topic.

Would there be increased delay on audio analogue OUTs vs Digital eg SPDIF?...or can I do this test with either?

[ski]

Hi Swan,

Would there be increased delay on audio analogue OUTs vs Digital eg SPDIF?...or can I do this test with either?

Maybe it would be good to clarify what the recording delay parameter does before I take a stab...

Let's look at analog recording for a second... if you were to play a cowbell with a stick (sharp transient) into a mic and record that on tape, here is the chain of events:

• sound wave travels through the air hits microphone, causing its diaphragm to vibrate
• microphone signal is amplified by preamp (analog electronics)
• signal travels to tape recorder's analog electronics
• tape heads get energized
• magnetic pattern gets imparted to moving tape

From the time the sound wave hits the mic, there is virtually no delay between the time the mic's diaprhagm starts moving and the magnetic particles start moving on tape. This is because electrons (analog electronics) travel at near the speed of light. For all intents and purposes, there is zero delay introduced by the electronics.

The only place any kind of delay would be introduced has to do with your distance from the mic. If you are 2" away, it will take the sound of the cowbell less time to hit the mic than if you are 22" away. If the cowbell part sounded better the further away you were from the mic, your human musicianship would cause you to instinctively play head of the beat. Human Delay Compensation!

Digital recording is just soooo different. Once the signal hits the audio interface, it has to be sampled (time sliced) at the sample rate of your session. It will always take -- at minimum -- one sample's worth of time for the computer within the audio interface to calculate the proper value for each time slice it samples. But it will usually take many more samples' worth of time to do this calculation.

Once the calculation is done for each sample, a digital value for that sample passes to your computer (either via FW or PCI card). Both of these portals are governed by software which may introduce its own delay of at least one sample. But more realistically, this delay can often be more significant, on the order of tens or even hundreds of samples.

To summarize, the analog-to-digital conversion introduces a delay and the driver software introduces a delay. Let's say the total accumulated delay is 100 samples. When you record audio, you do it in real time. By the time it gets actually recorded in your DAW, it's 100 samples late! The recording delay lets you compensate for this.

Now, to your question!

If you record digitial--->digital via your audio interface, no time-consuming A/D conversion has to take place. But you'll likely still have a delay of some amount, because all digital audio recording systems use a clock (word clock or similar) to keep the audio streams flowing synchronously. So there may still be a delay, it just won't be as great as when doing analog recording.

[darkecho]

what if I do this and I dont have any delay?

[ski]

what if I do this and I dont have any delay?

Not sure what you mean... do what?

Aside from that, in any digital recording system there will be some delay. It's inevitable and unavoidable, even if it's only a few samples-worth of delay (though it's usually much more than just a few...)

[m-m-m]

hey ski ...

Could I do this in mono?? My audio interface ha 2 ins, but 1 is a Hi-Z 1/4" jack while the other is female XLR Low-Z. Anyways, I'd be worried that things wouldn't match up to well if I tried to loopback like this. I read quickly through your instructions and didn't see any reason why it needed to be stereo .... or did I miss something??

I was thinking that I could just record a mono midi track, and then just boune mono and import that to the arrange window, or would I be better off using stereo tracks as you listed but only using one side .... help me out here, can't you see my head is spinning!!!!


[deep breaths} ..... {/deep breaths] ..... I'm OK now!

[shivermetimbers]

Arrgghhhh,

I did this to one of those kung fu movies (where the voice and mouth are out of sync) and after the adjustments the voices were in sync with the mouth movements and it kinda ruined the whole movie...

[ski]

Deep breath... hold it... hold it... hold it... let go...

Aaaaaahhhhhhh......

hey ski ...

Could I do this in mono??

Yes, but...

I was thinking that I could just record a mono midi track, and then just boune mono and import that to the arrange window [snip]

Bouncing audio and then bringing it back into the arrange window doesn't result in a recording that includes the amount of delay introduced by your interface/blah blah blah. The only way you can get your recording delay setting set properly is to do a loopback test.

Hope that makes sense. If not, post back.

@Shivermetimbers, LOL!!

[darkecho]

I mean that I did this process and the resulting audio is aligned at the most equal-phase position possible when I move the anchor around. I flip the phase and shift the anchor and it only gets louder either direction I go, as in, I have no delay in the system!

I don't know what I am doing wrong. I can never quite get the sound to cancel completely no matter what I do.

[Ploki]

hey.. just one question.
i did a loopback on S/PDIF, and what i recorded was EARLIER in time than the original!
i had to move it BAKCWARDS to get complete cancelation, i did the MAX zoom in the arrange, and it was about 8 samples IN FRONT. i turned off PDC for "all", turned of software monitoring
buffer size 128
:S i dont get it

should i try the analog ports? i mean, it makes no sense that without recording delay compensation even enabled i get audio recorder EARLIER than it even is played

[m-m-m]

I was thinking that I could just record a mono midi track, and then just boune mono and import that to the arrange window [snip]

Bouncing audio and then bringing it back into the arrange window doesn't result in a recording that includes the amount of delay introduced by your interface/blah blah blah. The only way you can get your recording delay setting set properly is to do a loopback test.

I'm not 100% sure, but I think that you misunderstood the part that you quoted above.
What I meant was to bounce a midi region to create an audio region for use in the loopback (as the original source that gets fed into the input of my interface and then loopedback). I was thinking Klopfgeist (sp?) ... nice clean transients!


Loopback, loopback, loopback, loopback .... if you say it enough the word looses all meaning.

[Ploki]

pardon me, i was doing the first check 6AM in the morning...
i must have screwed something up or mixed the tracks..

8 msec @ 88200 Hz via S/PDIF :S
thats a bit much isnt it.
and then i have to change the settings everytime i work at different sample rate?
anyone listed automatic delay compensation under new features suggestions?

anyway, i dont have two balanced patch cables on me right now, but, is the delay compensation the same with the Analong i/o? spdif doesnt make any conversion and analog io does...

[leahbasskitten]

i'm actually kinda shocked i understood most of this, but i do have an interesting angle that would help me if i understood...

so, instead of a mic, i'm direct in with a... lets say a bass. There is no room for delay like a cowbell being across the room. now there is latency caused more likely by the audio device, maybe some plugins. i know that i use ski's patent pending Human Delay Compensation technology.

crap, i think i answered my own question, but i've gone this far.... i was going to say, then am i recording the latency i hear, or what it actually what i was playing. BUT, the latency would be caused by the device, what i heard is what is being laid down.

well, let me ask another question (i'm sure the answer is the obvious). lets say its software causing this delay such as a plugin, or maybe logic itself. now what i hear from monitoring logic even if its just some amp sim, what i'm hearing is different then what i'm playing, and what i'm hearing is also different from logic is scribbling down.

is there away around this? i mean, i can't play a naked guitar sound in the middle of a rock song, it just takes away from the groove. do people have their low latency guitar patches, and then spruce it up later?

[ski]

I mean that I did this process and the resulting audio is aligned at the most equal-phase position possible when I move the anchor around.

Are you basing this on the visual display you're seeing in the arrange window, zoomed in all the way so that you can compare the waveform display of the two different tracks (original and looped-back)? If not, how are you determining this? (See #2 below).

I flip the phase and shift the anchor and it only gets louder either direction I go, as in, I have no delay in the system!

I don't know what I am doing wrong. I can never quite get the sound to cancel completely no matter what I do.

Maybe... just maybe... based on what you're saying, it's possible that your recording delay setting is already at an ideal setting. But then again, maybe not... Not being able to get it cancel completely means one or two things:

1) there's a level (or tonal) discrepancy between original and looped-back recording

2) you've got your waveforms aligned to a point that isn't at the right place and the nature of the signal is such that you're able to get partial cancellation anyway.

Regarding #1, one way to determine this is to normalize both the original and looped-back recordings. See if you get cancellation afterwards...

Regarding #2, post a screenshot of your arrange page showing the first cycle of the waveform on both your original and looped-back tracks.

[ski]

hey.. just one question.
i did a loopback on S/PDIF, and what i recorded was EARLIER in time than the original!
i had to move it BAKCWARDS to get complete cancelation, i did the MAX zoom in the arrange, and it was about 8 samples IN FRONT. i turned off PDC for "all", turned of software monitoring
buffer size 128
:S i dont get it

should i try the analog ports? i mean, it makes no sense that without recording delay compensation even enabled i get audio recorder EARLIER than it even is played

On some systems the looped-back signal will indeed be earlier. I recall this being the case with certain RME audio systems, though I don't recall which ones. So it's not unheard of that your recording delay will have to be a positive number rather than a negative number.

I also wouldn't be surprised if the SPDIF loopback exhibits a different loopback latency than an analog loopback. That's because a SPDIF loopback doesn't require time-consuming A/D and D/A conversion.

Keeping in mind the ol' mantra of "never say never"... I don't think there'd be too much reason to loopback digital signals (like SPDIF out--->SPDIF in), though it's worth investigating how much recording delay there is in the event you did have to do lots of that kind of thing. But anyway, probably the more common reason for having a properly calibrated recording delay setting is to compensate for the latency introduced into the positioning of analog signals once they're converted to digital, i.e., anything analog that's coming into your audio interface and getting converted to digital audio.

[Ploki]

thanks for the reply
just figured out i only need one cable anyway, since i have symmetrical 8 in/outs, so there is really no reason to record stereo, otherwise everthing would go to hell for every added input, and if THAT happens, well, as far as im concerned, i can throw this away!.

i did it on S/PDIF to get sample accurate loopback (i actually cancelled out everything completely, i guess that works A OK.

yeah sorry, i actually read your whole post but had too little time on my hands to reply

will calibrate analong asap.. if only AudioFW would crash every 35minutes...