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how to measure latency
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Caqueira
Started Topics :
2
Posts :
21
Posted : Sep 17, 2004 01:16
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hi guys, I have searched all over the forum for this, but without any success.
this could be a stupid question for a advanced producer, but since I am a newbie, I could use some help.
so, how do ya measure your latency? is it on the software, on the soundcard? is it any specific software to do that? is it any specific software o decrease latency? what should be an minimum accepted (or audible) latency?
please share your knowledge with this troubled mind.
thank you all.
boom!
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~Caqueira~
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Caqueira
Started Topics :
2
Posts :
21
Posted : Sep 17, 2004 01:17
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forgot to say:
I am running Logic on a Delta Audiophille 24/96 soundcard.
that´s it.
cheers! |
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dNETv2
Started Topics :
2
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168
Posted : Sep 17, 2004 01:36
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Look in your M-Audio control panel. DMA buffer size. |
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ZilDoggo
Started Topics :
4
Posts :
663
Posted : Sep 17, 2004 23:27
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you can measure the latency of the whole chain like this:
(the whole chain is: logic->card drivers->motherboard bus->card output->card input->motherboard bus->card drivers->logic)
make a track of audio with a sound that plays EXACTLY on the beat .,
make another track to use as an input.
set the output to off for the input channel so it will record but not make a sound while recording.
it's the little list box just above the fader for the channel you use as input.
this is important because otherwise you will get feedback.,
then connect the outputs of your card to the inputs.,
now, when you press play you should see a signal comming into the input track..,
then record .,
if everything goes well you will have a new piece of audio that has gone through the whole chain and back again.,
this should be off the beat a little.,
now you can see in the sample editor how many samples it is late.,(approximately)
if you know how many samples it is late and you know your sample rate you can calculate the latency.,
so, if your sample rate is 96khz and you see that the new audio is 3840 samples late then you divide 3840 by 96000 = 0.04 seconds = 40 miliseconds..
remember that this is two-way !
so when you only play back you will have about half of this latency (so 20ms).
also, when you record an external source it would also have a latency of about 20ms
hope this helps
greets.,
aka., |
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Caqueira
Started Topics :
2
Posts :
21
Posted : Sep 18, 2004 00:05
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Thanks a lot guys, this was very helpfull.
going to do what ZilDoggo suggested and get back to ya, sounds like an excelent ideia. My only fear is that I will get for sure a nasty latency.
best regards,
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~Caqueira~ |
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ZilDoggo
Started Topics :
4
Posts :
663
Posted : Sep 18, 2004 00:21
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this will propably be mostly influenced by the dma buffer setting, like dNETv2 said.,
if you want a lower latency you need to change that setting.,.
if you have a modern computer you can go as low as 10ms or lower., this is quite good for making music imo.
greets.,
aka |
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art
Started Topics :
2
Posts :
19
Posted : Sep 24, 2004 13:41
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oh i heard of a good site on another trance forum about this:
http://mark-knutson.com/t3/
he has made some free software that analyses your buffers and latency where your cpu is being used etc, pretty cool, he reckons video cards can be a major prob, hope it helps...
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High Pulse
Darkpsy
Started Topics :
57
Posts :
1187
Posted : Nov 10, 2004 05:45
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hey babba .. if u using logic u can chek the latençy in the transport bar .. u see there the number counting .. if u see them fast .. no latency .. if u see them get slower and stoped for some miliseconds ... there is latençy .. then .. use the less buffer from the delta .. use like maximum 128 samples .. 1 milesecond i think at 44khz
and go to audio . then system performance 
chek there too
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