PROCESSING OF THE AUDIO SIGNAL
For converting the analog signal from the microphone into a digital signal, pulse-code modulation (PCM) is used. In this system the signal is periodically sampled and each sample is translated into a binary number. From Nyquist's sampling theorem the frequency of the sampling should be at least twice as high as the highest frequency to be accounted for in the analog signal. The number of bits per sample determines the signal-to-noise ratio in the subsequent reproduction.
In the compact disc system the analog system is sampled at a rate of 44.1 kHz, which is sufficient for the reproduction of the maximum frequency of 20 kHz. The signal is quantized by the method of uniform or linear quantization, the sampled amplitude ...
Become an O’Reilly member and get unlimited access to this title plus top books and audiobooks from O’Reilly and nearly 200 top publishers, thousands of courses curated by job role, 150+ live events each month,
and much more.
Read now
Unlock full access