Ableton Forum

just installed and running the live intro demo on windows 7 with a line6 toneport gx usb audio device.

Just ran throught the audio setup tutorial(its all pretty clear). turned off the warpsetting bit and used the sample start window (3rd box)to correct the delay. the figure it gives me is 12. ive entered 12 into the driver error compensation box and rebooted Live. i do the test again and i get the same result each time. the same delay on the recorded audio. the error compensation appears to do nothing.

ive used the program before but ive just started using it to record guitars. if i can get this recording delay thing sorted out im going to buy the program.

can anyone help me suss out whats going wrong here? im not going to part with my cash if it means im going to have to manually adjust every clip i record. thats just plain annoying. is this a bug or am i missing something here?

anyone else had this problem? or has anyone else used it successfully?

anyone?

without diving into it myself at the moment,

i can say: I have tried setting it without following _to the letter_ the procedure in the lesson. and got no result.

however, if I was super anal about following the directions, it did align the waveforms.

in my case, it ends up at 0.88ms so hardly a deal-breaker (?)

well i need to correct by 45milliseconds which is hellish for a recording of guitars. (its only 12 milliseconds on my laptop, but it still makes recordings sound awful)

as far as im concerned this feature simply does not do what the manual and lesson says it should.

i can get the waveforms to align. the figure im given is 45 so im assuming i put 45 in the driver error correction and it should correct my recording from then on.

i just find it hard to believe that the value is 45 ms. i think you may be confused as to what this function is for. if the preference pane tells you that your total latency is 45ms, that DOES NOT mean that you enter that in the compensation box!

this preference setting is for correcting a mis-reporting of soundcard latency to the host (live). it is calculated by physically routing the soundcard output to it's input. (with a cable). best to follow the lesson on this topic, to the letter.

it is not for compensating monitoring latency.

^^ no offense intended of course.

i'm just trying to figure the ENORMOUS difference between me (0.88ms with a discontinued, 6-7 year old audio interface) and you (45ms)

??

45 ms just sounds like a monitoring latency, with a buffer of somewhere in the 512 range