The elements of the block diagram are :

- A The microphone : It converts the audio input into an electrical signal v
_{i}(t); - B Low Pass Filter : It passes low frequencies, in telephony the maximum sampled audio frequency is about f
_{i,max}≅ 3.4 kHz. The purpose of the low pass filter is to prevent aliasing when we attempt to convert back to analog the audio signal. We sometimes refer to the filter as anti aliasing filter; - C Sampler : This is an . It produces the samples at every T
__electronic switch__A transistor or an IC._{s}seconds, where the sampling frequency in Hz is equal to f_{s}= 1/T_{s}. The value of f_{s}is dictated by the Nyquist theorem. The voltage values of the produced samples are called V^{*}and the wave is called a PAM signal or Pulse Amplitude Modulated; - D Uniform quantizer : It converts the random values of V
^{*}to a predetermined ones called V_{q}. The operation of the uniform quantizer is as follows :

V^{*}= V_{q}if V_{q}≤ V^{*}< V_{(q+1)}. The level value depends on the number of bits per sample called n. A quantizer has a number of levels equal to q = 2^{n}where n is the number of bits per sample; - The coder converts V
_{q}values into a binary sequence made of n bits.

The GUI above is a script i have written to illustrate the relation between the audio frequency f

The objectives of this exercise is to verify the relation between the number of uniform levels q and the number of bits n per sample produced by the coder.

In the figure above, the number of levels produced by the uniform quantizer D is equal to 4. The number of levels can be calculated by using the following : q = 2