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One of the biggest problems when recording in the digital world is latency. Latency is the measure of time it takes in milliseconds for your audio to pass through your computer during the recording process. This delay is caused by the fact that your computer has to receive, understand, and process the signal, then send the signal back to the outputs for you to hear. In this movie I'm going to show you how to lower latency so it doesn't bother you when you are recording. High Latency means that you hear a note from your DAW a way after you play or sing it. I really want to avoid this especially if you are doing overdubs.
High Latency means it's taking too long for the audio input to get to the audio output, which means there is lag time between the time you play a note, and when you hear it. A very small lag time of 3 to 6 milliseconds is tolerable, but anything beyond that creates everything from a phasing sound to a full echo. This makes it anywhere from distracting, to impossible to sing or play with. The lower your latency, the more your recording will stay in sync with the music that you're playing back up to a point. If you try to set the latency parameter too low the audio stream can break up into random static since the computer doesn't have the time to process it.
Here is an example of some really long latency. (music playing) The key is to adjust your latency as low as it can go without causing the computer to stutter. You do this either through your sound card or interface settings or through third party audio drivers.
How low your latency can be set is dependent upon such factors as computer speed, system bus speed, sound card performance, and system memory. Most computers purchased today are powerful enough that you can get latency pretty low, but you still have to experiment to find the settings that provide the best performance. The parameter that most computer audio interfaces use to set the latency is called the Input Buffer. The smaller the buffer, the lower the latency, but the harder the CPU has to work. If you lower the buffer size too much, the setting can produce crackling noises.
Although this is a function of the horsepower of the computer. These noises crop up when the CPU literally has to drop audio bites because it can't keep up with the audio stream. Today's fast computers can get the I/O buffer size down to 32 samples, which results in latency of 0.65 milliseconds at a 48 K sampling rate. The more tracks and processing you add, especially when running at sampling rates higher than 48K, the harder the computer's CPU will have to work, which means that you need to increase the buffer size to prevent dropouts.
It should be noted that it's best to not use any software plug-in processing like compressors or EQ when recording. That's because each plug-in adds anywhere from a little to a lot of latency just by the fact that it's inserted in the signal path. Keep that path as efficient as possible with as few things inserted as you can and your signal will not only sound better but will stay in sync as well. Many audio interfaces are equipped with zero latency monitoring which is an analog bust that loops directly from the interface's input to its output without passing through the computer.
Once you've set up this routing in your interface's control panel applet, the player or singer will be able to monitor the backing tracks and get his or her performance in sync without any time delay whatsoever. To sum it up latency is the measure of time it takes for your audio signal to pass through your computer during the recording process. Lowering your DAW latency will make all your recording in overdubs go smoother. You do this either by using a zero latency DAW interface or lowering your I/O buffer size. Finally, it's best not to use any plug-ins while recording as each plug-in adds to your latency.
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