original analog signal (voice) to digital data

Original Analog to Digital data:
Sometime we have analog signal such as one created by a microphone or camera.

There are many techniques to convert analog to digital data. We will learn main two techniques.

* Pulse Modulation (PM)

* Delta Modulation (DM)

Pulse Modulation:

There are four predominant methods of pulse modulation are

1. Pulse Width Modulation (PWM)

2. Pulse Position modulation (PPM)

3. Pulse Amplitude Modulation (PAM)

4. Pulse Code Modulation (PCM)

Pulse Code Modulation (PCM): 
In 1937 Alex H. Reeves invented PCM. PCM is the most common techniques to change analog signal to digital data (digitization).
A band pass filter limits the frequency of the analog input signal to the standard voice-band frequency range of 300 Hz to 3400 Hz. 

A PCM encoder has three processes.

a) Sampling

b) Quantized

c) Encoded.

a) Sampling:

            The analog signal is sampled every T_s second. Where T_s is the sample interval or period.

There are basic two sampling method used to perform the sampling function

i) Ideal
ii) Natural

iii) Flat-Top

ii) Natural Sampling: A high-speed switch is turned on for only the small period of time when the sampling occurs. The result is a sequence of samples that retains the shape of the analog signal. This is difficult for an ADC to convert the sample to a PCM codes.

iii) Flat-Top: The most common sampling method which accomplished in a sample and hold circuits. The circuits periodically sample the continuous changing analog input voltage and convert those samples to a series of constant-amplitude PAM voltage levels. With flat-top sampling, the input voltage is sampled with a narrow pulse and then held relatively constant until the next sample is taken.

Aperture Error it creates when the amplitude of the sampled signal changes during the sample pulse time. However, flat-top introduces less aperture distortion than natural sampling and can operate with a slower analog-to-digital converter. The sampling process creates Pulse Amplitude Modulation (PAM).

Sampling Rate: There is a restriction on sampling rate (T_s).

According to Nyquist theorem, to reproduce the original analog signal, each cycle must be sampled at least twice. Again, the minimum sampling rate (f_s) is equal to twice the highest audio input frequency (f_a). If not so an impairment called alias or foldover distortion occurs.

f_s > = 2f_a

b) Quantization:

            With PCM, the analog input signal is sampled then converted to a serial binary code. The binary codes used for PCM are n-bit codes, where n may be any positive integer greater than 1. A PCM code could have highest only eight bits, which can make 2⁸ or 256 combinations. The codes currently used for PCM are sign-magnitude codes, where most significant bit (MSB) is the sign bit and the remaining bits are used for magnitude. For three-bit sign-magnitude code can make eight possible combination.

Sign bit	Magnitude
1	11
1	10
1	01
1	00
0	00
0	01
0	10
0	11

Sign bits 1 are all positive and 0 are all negatives. This type of code is called a Folded Binary Code because the codes on the bottom half of the table are a mirror of the codes on the top half.

The result of sampling is a series of pulses with amplitude values between the maximum and minimum amplitudes of the signal. The set of amplitudes can be infinite with non-integral values between two limits. These values cannot be used in the encoding process.

We have to follow the following steps –

* Divide the sample analog signal by some levels. The number of levels (L) is set by the user and each level identified by a quantization code. But number of level we will choose depends on the range of the amplitudes of the analog signal and how accurately we need to recover the signal. If the amplitudes of the signal fluctuate between two values only, we need only two levels. If the signal is voice, has many amplitude values, need only more levels. In audio digitizing, L is normally chosen to be 256. In video it is normally thousands.

First quantized value is 1. Cause the maximum amplitude is between 0 and 1 quantization code. Second value is 3. 3^rd one is 3. In this way 4: 3, 5: 3, 6: 2, 7: -1, 8: -1, 9: -3, 10: -3, 11: -3, 12: -3, 13: -1

Quantization Error: All these quantization value of amplitude is not perfect value. Cause for an example for 1^st amplitude is between level 0 and 1, suppose 0.9. But its quantized value is 1. So the 0.1 is extra to make it an integer value. So for this amplitude 0.1 is its quantization error.

C) Encoding:

            Last step is PCM is encoding. After each sample is quantized and the number of bits per sample is decided, each sample can be changed to a binary code word we see before.

           

PWM, PPM, PAM: 
PWM is sometimes called Pulse Duration Modulation (PDM) or Pulse Length Modulation (PLM). The width of a constant amplitude pulse is varied proportional to the amplitude of the analog signal at the time the signal is sampled. The maximum analog signal amplitude produces the widest pulse, and the minimum analog signal amplitude produces the narrowest pulse. But all pulses have the same amplitude. PWM used in special purpose communications systems mainly for the military and sometimes used in commercial digital transmission systems.

 

PPM, the position of a constant-width pulse within a prescribed time slot is varied according to the amplitude of the sample of the analog signal. In figure the higher the amplitude of the sample, the farther to the right the pulse is positioned within the prescribed time slot. The highest amplitude sample produces a pulse to the far right, and the lowest amplitude sample produces a pulse to the far left. PPM also used in special purpose communications systems mainly for the military and sometimes used in commercial digital transmission systems.

PAM, the amplitude of the constant width, constant-position pulse is varied according to the amplitude of the sample of the analog signal. In figure it can be seen that the amplitude of a pulse coincides with the amplitude of the analog signal. PAM waveforms resemble the original analog signal more than the waveforms for PWM or PPM. PAM is used as an intermediate form of modulation with PSK, QAM.

PCM is very complex technique. To reduce the complexity other modulation has been developed. This is Delta Modulation.

Delta Modulation (DM):

With conventional PCM, each code is a binary representation of both the sign and the magnitude of a particular sample. Multiple-bit codes are required to represent the many values that the sample can be. With delta modulation, rather than transmit a coded representation of the sample, only a single bit is transmitted, which simply indicates whether that sample is larger or smaller than the previous sample.

The Modulator is used at the sender site to create a stream of bits from an analog signal. The process records the small positive or negative changes, called delta (d). If the d is positive, the process records a 1, if it negative, the process records a 0. The process needs a base against which the analog signal is compared. The modulator builds a second signal that resembles a staircase. Finding the change is then reduced to comparing the input signal with the gradually make staircase signal.

The Modulator, at each sampling interval, compares the value of the analog signal with the last value of the staircase signal. If the amplitude of the analog signal is larger, the next bit in the digital data is 1, otherwise 0.

Here we found some problem in DM

1. Slope overload: some times we can see for this staircase, it make a big distance from original analog signal. This is Slope overload.

2. Granualar Noise: Sometimes the analog signal can like flat. Than the reconstructed signal can be changed for DM.

This problem is solved by Adaptive DM

Adaptive DM:

If get three 1 then it increase its up stair level, and continues this level until get any 0. If get three 0 then it increases its down stair levels until get any 1.

If found three 10 or 01 that’s mean 101010 or 010101, then it decrease it’s up and down stair level to avoid granular noise.

After PCM or DM we got digital data (combination of 0 and 1). But media passes signal. So we need to make this digital data into digitally converted analog signal. But directly cannot make it. So first we have to make digital data into digital pulse.

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