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-- Fixing my Latency Settings in Logic

Fixing my Latency Settings in Logic

I went to the audio tab to set my latency settings, but it won't let me choose anything lower than 64 milliseconds..

Im using Logic 5.5 for Pc and my soundcard is a delta 44


When i first began i used to use an Audigy 2 soundcard.. .and when i would go into the audio tab it would let me choose 0 if i wanted to.

Now that I'm on a much better soundcard it wont let me go lower than 64 for some reason.

Anyone know whats up?

Do you notice any problems with latency?

Have a look in the sound card device window. You may have to alter it externally from Logic

it takes me to my sound card properties externally from logic... but the lowest setting is 64..

i cant figure out how to get it less than that.


does anyone know how?

Are you sure the 64 doesn't refer to SAMPLES and NOT ms? 64 samples is something like 2 ms.

es

you are right vert... it does say samples.

what does that mean anyways 64 samples? how come it's not displayed in miliseconds? and how do i get it down to zero miliseconds?

There is no such thing as a zero sample latency. Even hardware has at least one sample as it takes time to process and as long as you have to process input you'll have to wait one sample to have it AD convertetd.
Latency in samples is more accurate because 2ms at 44,1kHz relate to 88.2 samples. Buffers always will be a few samples long. You can fix that with predelays in samples, not in ms.

Try to live with some latency. The smaller the buffers, the more time you need for bufferhandling the less time for audioprocessing. If you don't need to play live, better use longer latencies and compensate for it. Logic hast that option, so use it

why should i compensate for it? what are the benefits? how do i compensate for it? why is production so confusing? :/

Latency is important if you're trying to sync hardware to software. Latency is the time between any hardware event and the response of your software.

That means if you trigger a midi note from your external controller it takes at least one buffersize for the triggeres softsynth to react. Same applies for external synths triggered from your sequencer. It takes at least one inputbuffer till the external sound can be put out..

Now basically every sequencer has the option to predelay some tracks just by sending the midi notes a bit before the acual event should occur. By that external audio will arrive just when needed. Logic for example can automatically compensate for every insert you use.

This only works if you trigger hardware from your sequencer, as it knows all notes before they are about to be played. For the other way you'll have to deal with latency issues.

My system actually has a latency of 150ms. I'd be unable to play any softsynth live. But as I'm only seqeuncing and only use softsynths... no problem at all.

latency doesnt affect softsynths? and why would you want a higher latency setting?... i dont understand that part.

Latency does affect softsynths... but all in the same way. No one can tell the difference of 140 or 14 ms when clicking the play button. Everything just starts a tad later. I'm talking about set arrangements. No live inputs. Every bit of data is known ahead and can be processed at the time it's needed.

As I said before small buffers need alot of handling, as everytime a buffer is filled it needs to be send to output, flushed, reinitialized and refilled. In a strict hardware environment the buffer goes the shortest way down to output. But most of us are running windoze. Just one priority event an a short buffer will be underrun. Just try to defrag jour drive with small buffers and enjoy the crackling and looping (Well you don't do that normally but think of loading samples or memory swapping).

Larger buffers need less handling leaving more processortime for synthesizing. As Logic opens all available channels at all time this might be essential.

i just want my stuff to be sequenced properly... the timing most (if not all of the time) seems to be off.

sounds like the bass and synths are playing at different speeds or something... but i dont know how to correct them.


if i apply my instruments to a midi file that is supposed to play them properly... it's most likely that when i put my instruments to them, the sequencing is totally messed up for the most part.

do you know how to sequence your stuff properly? can you lend me a hand along these lines? cuz i dont know whats going on... why its happening.. whta is causing it.. how to fix it.

thanks

Have you tried higher latencies?
I had the that problem with the audigy2. Everything started off correct but went totally wrong after a few seconds. Just like Bass and Bassdrum were running at different speeds. Only got the Windows Media Drivers to work. Switched back to SB live because of that. I suppose it's some ASIO driver issue, as Audigy2 ASIO drivers are known to be crap.

see... thats what i cant understand here... isnt my goal to have the smallest amount of latency possible?

Well maybe it's the opposite. If a buffer can't be processed in time it will get lost, causing one instument to be one buffer ahead of the others.

Imagine an array of water tanks. What you do is run along them and fill a bucket. Each round you'll have to empy it at an exact time. Ensuring a constant output flow. With a small bucket you'll have to run long distances for small amounts of water. If one of the tanks does not fill the bucket fast enough you wont be able to make it in time do flush the water where it belongs. You'll have to run back to the start and empty the bucket on the way. But now the last tanks weren't properly processed and emptying them will take more time... the water coming from them is basically delayed... now if you take a larger bucket, you won't be in such a hurry to make it in time from flushing back to the start, as the water of the last bucket will take a litte time to flush.

As I said, with small buffers it's the buffer handling, that becomes critical and time consuming.

Hope that will make it clearer.

yes good analogy... so how would i set up my buffers so that i can produce music that is timed perfectly?

Is your system running in sync using only Logic instument? If not, try larger buffers or different drivers.
If you got that, simply predelay the MIDI Tracks for your hardware corresponding to to the buffersize.

(trackdelay/ms) = - (buffersize/samples) * (samplerate/kHz)

Timing doesn't need to be that exact. MIDI is a serial protocol anyway so timing can never be exactly on spot. The human ear percieves impulses up to 10ms apart as only one impulse. MIDI itself transmits at 8kHz meaning two sent MIDI notes are in any case at least 5 samples apart.

right now my set up is just logic audio platinum and installed vst's.... z3ta, albino.... that stuff.

So my guess would be a driver problem. Can you switch to a different driver system like PC AV? Does it work there?

im not able to use my production computer and be here at the same time..... but i know there are so many different options in my audio driver tab 1 & 2... i wouldn't know where to start.. and im pretty sure i've messed them up beyond belief as well lol.

i dont understand what options are for what and which ones i need to set to whatever so that they do so and so.

i have no understanding of my audio driver tabs at all.

i want it set up so that my timing is flawless :/

i believe my driver system has always been pc av... which means? and what is the alternative and what does that mean?

Your Delta 44 supports ASIO. This is the way to go.
PC AV uses Windows MME and DirectSound drivers. ASIO is a low latency soundsystem designed especially for multichannel streaming audio.

Just uncheck PC AV and check ASIO.

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