When audio comes in through your audio interface, analog-to-digital (A/D) converters sample the audio stream a certain number of times per second. That number is the recorded digital audio file's sampling rate. CDAudio uses a sampling rate of 44.1 kHz, which means the audio stream has been sampled 44,100 times per second. This is a lot of little samples, and consequently CDAudio comes pretty close to representing the original audio waveform.
But the holy grail of digital audio is to copy an analog waveform perfectly. In pursuit of this perfect waveform, audio editors like Logic now enable us to record and edit audio at sampling rates of 48 kHz, 96 kHz, and even 192 kHz (audio hardware permitting). These sample rates provide significantly more samples per second than CDAudio, which means these higher sampling rates come closer to representing a true analog waveform. Consequently, they sound better!
Higher sampling rates mean more data to process every second, putting a strain on your computer. If your system's audio interface works with 96 or 192 kHz audio, you may want to edit at this sampling rate to preserve as much of the original analog signal as possible. But unless you're on a cuttingedge Mac, higher sampling rates won't let you use as many audio tracks and DSP effects as you can at lower sampling rates. The most commonly used sampling rate is 48 kHz because it provides good quality without straining your computer to the max.
As you saw at the beginning of the lesson, this song uses a sampling rate of 44.1 kHz. All of the song's audio files use this sampling rate too, except for the HH.aif file, which has been recorded at 48 kHz. You can still use the HH.aif file in the song, but it will not sound correct unless you convert its sampling rate to match the sampling rate of the song. The reason? Logic just plays samples per second. It's much the same as the frames per second (fps) setting in a video editor. If you play a 30 fps NTSC video at 25 fps (PAL), the video plays back much slower than it should because five fewer frames display every second. Similarly, playing a 48 kHz audio file in a 44.1 kHz song still works, but the file sounds like its playing slower than it should because fewer samples per second play than are supposed to.
The following steps demonstrate what happens if you add a 48 kHz file to a 44.1 kHz song.
Using the Digital Factory to Convert Sampling Rates
The Digital FactoryTM is a suite of digital signal processors designed to change audio files in several ways. For example, you can use the Factory to timecompress or timeexpand an Audio Region, change its pitch, add groove or swing to a machinelike audio loop, or, for the purpose of this exercise, alter its sampling rate.
Under most editing circumstances, Logic is a nondestructive audio editor that does not change your source audio files in the course of editing. But as with most rules, there is a notable exception.
Enter the Digital Factory. Almost all Digital Factory functions are destructive, which means they permanently alter your source audio files. This can cause problems if you ever need to get those source files back or if they are used in another song, so keep this in mind!