FAQ, How to achieve low latency
Latency is another way of saying delay. So in sound and audio-cards, latency means a delay in your audio. When using software synths and VSTi the latency is the time the computer takes from when you press a key on your keyboard to when you hear the sound out of your speakers. Latency is needed because a computer is only capable of doing 1 thing at a time and audio is quite taxing on a computer let alone multiple tracks of audio. If your computer is busy and it doesn't process the audio in time then you will get glitches-pops-clicks in your recordings. To solve these drop out problems sound-cards use a buffer which as a side effect directly creates the delay. Latency comes from the computer reading ahead for example "1024 samples" and storing these in ram, as one sample is played another is added to the buffer. When the computer is busy theres hopefully enough samples stored in the buffer/ram to keep playing uninterrupted. This creates a waiting queue and is the delay you hear. As you increase the latency you give the computer more time to process the audio and it may result in lower cpu usage as well.
If u have a latency setting of more than 12ms and wish to use your computer to generate a reverb or other effect in realtime so u can hear that effect while u r recording then the sound u hear comming from the speakers will be 12ms behind what u sing into the microphone. Also some Plugins will add more latency to the audio since the computer has to process the sound to add the effect. With VST instruments becomming popular many keyboard players are finding latency annoying as when you press a key on the keyboard theres a delay that you can feel before the sound reaches their ears.
In short you cannot eliminate latency altogether, but you can reduce it so low as that you cannot feel or hear the effects or alternatively use hardware monitoring. As latency is only a problem when your using software monitoring or genearating sounds in realtime you can get zero latency from using a mixer to provide monitoring. Some audio-cards have a zero latency feature built in, refer to their manuals for more details on your card. The other solution if your having latency problems is to use an audiocard with low latency support and lower the cards buffer size. A buffer size of 512 samples or smaller will normally give acceptable results.
There are three ways.
If you have already lowered the buffer and the computer cant handle the buffer size then read the article here on tweaking windows 98 and XP also if your getting lots of clicks and pops then this article will assist you in eliminating them.
Facts about sound... Sound takes about 1 millsec to travel 1 foot. Agreed ? How far is your ears from your monitors in your studio ??? 3 foot with nearfield, WOW thats 3 whole ms of delay ! you had better start using headphones to mix now that I have told you that. LOL . OK you can hear/sense a latency of 12 ms and above. there are heaps of varibles so no one can define an acceptable latency. Remember if it sounds good then it is . I challenge people to hear less than 11 ms which is quite achievable these days (unless its a very fast transient) . Your ear cannot hear less than 10ms unless it hears the orignal sound at the same time as the dealyed sound as it then has a referrence with which to hear the delay. If it worries you so much just monitor the Input and buy an effects unit because no card on the market will give u ZERO latency! If a card states this then it is all marketing hype. Even rack mount digital effects processors will give u some latency. If you use software synths then you have to live with latency as you cant simply monitor the input. 2ms latency can be achieved on most good quality computers that are setup correctly. If your looking for a high end computer for low latency then read this article.