Instruction Manual

529
Digital Audio Fundamentals
Editing Audio
See:
Digital Audio Fundamentals
Recording a Sound
To record digital audio, your computer monitors the electrical signal
generated by a microphone (or some other electroacoustical device).
Because the signal is caused by a sound, the signal strength varies in direct
proportion to the sound’s waveform. The computer measures and saves the
strength of the electrical signal from the microphone, thus recording the
waveform.
There are two important aspects of this measuring process. First is the
sampling rate, the rate at which the computer saves measurements of the
signal strength. It is a known fact of physics that you must measure, or
sample, the signal at a rate at least twice that of the highest frequency you
wish to capture. For example, suppose you want to record a moderately
high note on a violin—say the A whose fundamental frequency is 440 Hz
and all overtones up to five times the fundamental. The highest frequency
you want to capture is 2,200 Hz, so you need to measure the electrical
signal from the microphone at least 4,400 times per second.
Since humans can hear frequencies well above 10 kHz, most sound cards
and digital recording systems are capable of sampling at much higher rates
than that. Typical sampling rates used by modern musicians and audio
engineers are 22 kHz, 44.1 kHz, and 48 kHz. The 44.1 kHz rate is called
CD-quality, since it is the rate used by audio compact discs.
The other important aspect of the measuring process is the sampling
resolution. The sampling resolution determines how accurately the
amplitude of each sample is measured. At present, the music industry has
settled on a system that provides 65,536 different values to assign to the
amplitude of a waveform at any given instant. Thus, each sample saved by
your computer requires 2 bytes (16 bits) to store, since it takes 2 bytes to
store a number from –32,768 to 32,767. The scaling of the electrical input
signal level to amplitude value is determined by your audio hardware and by
the position of your input level control.
What if the amplitude of the sampled signal gets too high, such that a 16-bit
number is not large enough to represent it? What typically happens is that
the signal is clipped, cut off at the maximum value.
Here is what a clipped waveform might look like: