WEEK 4

 

AUDIO SIGNAL 

An audio signal is a representation of sound, typically as an electrical voltage. Audio signals have frequencies in the audio frequency range of roughly 20 to 20,000 Hz (the limits of human hearing). Audio signals may be synthesized directly, or may originate at a transducer such as a microphone, musical instrument pickup, phonograph cartridge, or tape head. Loudspeakers or headphones convert an electrical audio signal into sound. Digital representations of audio signals exist in a variety of formats.

Likewise, digital audio is composed of discrete points representing the amplitude of the waveform rather than a continous sinusoidal wave. At these points are the sequence of seperate samples taken from an analog audio waveform. Hence, the more the sample in use, the higher the representation will be.

Meanwhile the analog audio is an audio wave recorded or used in its original form with the signals stored as a continuous signal (sinusoidal waves). 

HOW ANALOG TO DIGITAL CONVERT?

Microcontroller can't read the values unless it is digital data. This is because the microcontroller can only see the 'levels' of the voltage, which is depends on the resolution of the analog-digital converter (ADC) and the system voltage. So, the ADC follow a sequence when converting analog signals to digital. There are two important aspects of the ADC which is sampling rate and resolution.

When converting an analog audio source into digital audio, the digital encoding shall require: 

• sampling frequency
• the number of bits used in taking samples    

Two main factor affecting the digital audio quality :

• bit depth or sample format 
• sample rate or number of bits 

    1. Sample rate 

Sample rate is the value representation of the waveform in terms of the number of samples captured per second. Higher sample rates can support the representation of higher audio frequencies. The sample rate needs to be more than double the highest quality audio currently present. Digital samples cannot correctly represent frequencies that are more than half the sample rate. The half of the sample rate is representation of upper limit called Nyquist frequency. 

When correctly representing the analog waveform digitally, it should be below this limit. Representing analog frequencies at this limit or above will be incorrect and would lead to kind of distortion called aliasing.  

2. Bit depth or sample formats 

Computer stores information in 1 and 0s. Those binary values are called bits. The higher the number of bits indicates, more space for information storage. When a signal is sampled, it needs to store the sampled audio information in bits. This where the bit depth comes into place. The bit depth determines how much information can be stored. A sampling with 24-bit depth can store more nuances and hence, more precise than a sampling with 16-bit depth.