Learning objectives

When you have completed this session, you should be able to

• grab audio CDs.

Digitizing Analog Sound Frequency Signals

Analog sound frequency signal can take any value of amplitude which may also vary constantly in time interval.

For digital sound storing in time units samples must be taken from the sound (time quantization). When amplitude samples, produced this way, are divided into determined number of units the size of the amplitude sample can be represented by a number (amplitude quantization). Thereafter the numbers received can be stored in digital format (coding).

The frequency of sample taking is determined by Shannon’s theorem. According to this the original analogue signal can only be restored from a sequence of impulse without distortion if the frequency of sampling is at least double of the highest frequency occurring in the original analogue signal.

Since the audible frequency range for man is between 20 Hz and 20 kHz according to Shannon’s theorem at least 40 kHz sample taking frequency is needed to produce perfect audio playback.

Higher than 20 kHz frequencies certainly must be filtered with a low pass filter.

The steps of digitizing are the following:

1. low pass filtering (LPF)

2. time quantization

3. amplitude quantization

4. coding

The equipment which executes these steps called analog-digital converter (ADC, A/D converter).

The equipment which converts digital signal to analog signal is called digital-analog converter (DAC, D/A converter).

Sound cards installed in computers usually contain both equipments.

For computer sound processing the parameters of Audio CD quantization is the standard which are the following:

• 44100 Hz sample taking frequency

• decomposition of 16 bit amplitude quantization ( the samples of amplitudes are decomposed to 65536 parts)

Certainly other quantization parameters can also be used. For studio technology 48 kHz sample taking and 20 or 24 bit decomposition is accepted. 8000, 11025, 22050 and 32000 Hz sample taking and 8 bit decomposition is used for lower requirements.

Sound is usually saved in Microsoft Wave (WAV) format with PCM coding. (PCM = Pulse Code Modulation) In case of 44,1 kHz, 16 bit quantization for saving one second stereo audio material 2 x 44100 x 16 bit = 176400 Byte is needed. This is more than 10 MByte per minutes.

For effective storing various compressing methods have been developed. The most important ones are described here.