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 wrote: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 wrote: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?
Would there be increased delay on audio analogue OUTs vs Digital eg SPDIF?...or can I do this test with either?
darkecho wrote:what if I do this and I dont have any delay?
m-m-m wrote:hey ski ...
Could I do this in mono??
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]
ski wrote: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.
darkecho wrote: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 wrote: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
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