Digital Communications

Digital Advantages
Better control of noise and distortion
Regenerative repeaters avoid accumulation of errors

Easier switching and multiplexing Easier signal processing

Channels and Information Capacity

All communication systems follow Hartleys Law:

I ! ktB

Capacity of a Noiseless Channel

Limited by frequency response of channel and the complexity of the code used Only the fundamental of the signal used needs to pass through the channel Given by Shannon-Hartley Theorem:

C ! 2 B log 2 M

Data Rate in Noisy Channel

Limits bit rate regardless of modulation scheme or coding Given by Shannon limit:

B log 2 (1  S / N )

Finding Logs to Base 2

log10 N log 2 N ! log10 2

Pulse Modulation
Sending an analog signal digitally requires sampling Sampling rate must be at least twice the highest frequency component in the signal If sampling rate is too low, aliasing results
Very undesirable

Aliasing
Produces distortion products whose frequency is difference between sampling rate and signal frequency f a = fs - f m

Pulse-Amplitude Modulation (PAM)

Consists of samples only Used mainly as first stage in digital system PAM by itself is not digital

Pulse-Duration Modulation (PDM)

Also called Pulse-Width Modulation (PWM) Used in large audio amplifiers Also used in control systems Formerly used in communications but not much any more

Pulse-Position Modulation (PPM)

Rarely used

Pulse-Code Modulation (PCM)

PCM is the commonest digital scheme The amplitude of each sample is expressed as a binary number The more bits are used for the number the lower the distortion Process is called quantizing and the resulting distortion is quantizing noise or quantizing error.

PCM Dynamic Range

Depends on number of bits used to encode each sample DR = 1.76 + 6.02m dB

Linear PCM
Steps between voltage levels are all the same size Gives very good results provided enough bits are used Problems arise when few bits used Many signals are towards the low end of range and are represented by only a few bits These low-level signals are greatly distorted

Companding
Low-level signals are boosted at transmitter so they are represented by more bits This reduces distortion for low-level sigs Distortion for high-level sigs is increased but is still less than for low-level signals At the receiver, gain must be reduced for low level signals to compensate

Companding
At transmitter (or recorder): compression At receiver (or playback): expansion Dolby and dBx are well-known forms of companding used in consumer audio systems Telephone system has its own version called mu-law companding

Mu-Law Compression
Vi ln(1  Qvi / Vi ) Vo ! ln(1  Q ) Q ! 255

PCM Coding
Signal must be low-pass filtered to avoid aliasing Next it is sampled Samples go to A/D converter to be converted to a binary number Parallel to serial conversion required before transmission If done, compression can be applied to the analog signal before sampling or to the digital number

Decoding
Serial-to-parallel conversion Signal goes to D/A converter If compression done at transmitter, expansion must be done at receiver Expansion can be done by doing arithmetic on the binary numbers, or on the analog signal Analog signal goes through a low-pass filter as at the transmitter

Codec
Stands for coder/decoder One chip can do both Sometimes called a combo chip

Differential PCM
Transmits only the differences between one sample level and the next Can be done with fewer bits per sample, based on the assumption that consecutive samples usually have similar amplitudes The most extreme form of differential PCM is delta modulation

Delta Modulation
An alternative to PCM Only one bit per sample Bit determines which direction voltage has moved since last sample
1 = up 0 = down

Delta Modulation
Sampling rate must be much higher than for PCM Total bit rate can still be lower

Slope Overload
Occurs in delta modulation system when sampling rate is too low] System cant keep up with rapid changes in voltage

Adaptive Delta Modulation

Reduces effect of slope overload After several bits of the same type, the step size increases to allow the system to catch up to rapid changes

Line Codes
A line code describes how logical ones and zeros are translated to physical values like voltage and current Many different line codes are possible. Here are a few examples

Unipolar NRZ Code

Unipolar means only one polarity of voltage NRZ means non-return-to-zero
There is no requirement for the voltage to return to zero at the end of each bit period

A positive (or negative) voltage can represent a logic one Zero volts can represent a logic zero The code for zero and one can be reversed

Bipolar NRZ
Similar to Unipolar NRZ except equal positive and negative voltages are used for logic one and zero If the number of ones and zeros is equal in the message there will be no dc component to the signal This is important if the line has transformers or amplifiers

RZ Codes
Voltage is required to return to zero once every bit period Better when clock information has to be carried with the data Bipolar RZ codes can be designed to have no dc component regardless of the data

AMI Code
Used for telephone signals Logic zero = zero volts Logic one = voltage pulse
Polarity of pulse switches from plus to minus with each pulse

Never any dc Long string of zeros will cause timing info to be lost

Manchester Code
Voltage transition signifies the value of a bit Transition from negative to positive = 1 Transition from positive to negative = 0 Never any dc component Timing info is always present Used in Ethernet LANs Requires more bandwidth than other codes

Time-Division Multiplexing
Recall from Hartleys Law that time and bandwidth are equivalent In FDM (frequency-division multiplexing) each signal has part of the bandwidth on a full-time basis In TDM (time-division multiplexing) each signal has the full bandwidth for part of the time

TDM in Telephony
Lowest level is DS-1 signal 1 sample from each of 24 signals is transmitted in sequence 1 framing bit also needed for synchronization

DS-1 Bit rate

8 bits per sample for each signal 8000 samples per second for each signal 24 signals 8 v 8000v 24 ! 1.536 Mb/s Add1 framing bit per frame ! 8000 framing bits per second f b ! 1.5356v106 8 v103 ! 1.544v106 b/s

DS-1 Bit Rate

Another way to calculate it Each frame has 1 sample (8 bits) from each of 24 signals plus 1 framing bit Each sample has
8 v 24 1 ! 193 bits We need 8000 frames per second Total bit rate : f b ! 193 v 8000 ! 1.544 Mb/s

Digital Hierarchy
DS-1 signal, T-1 line are lowest level Other, higher bit rates can be created by multiplexing DS-1 signals using TDM

Data Compression
Data compression reduces number of bits that must be transmitted. Two main types of data compression: lossless and lossy. Lossless compression takes advantage of redundant data or codes it more efficiently. Lossy compression removes data while having as little effect as possible on received signal.

Run-length Encoding
Use a short string of bits to represent a longer string, for instance say 20 zeros instead of 00000000000000000000. Use shorter strings to encode more common symbols, for instance, the letter e could use fewer bits than the letter q.

More Lossless Compression

Voice transmission systems can stop transmitting (or transmit at a lower rate) during pauses in speech. Video systems can transmit a frame once, and thereafter transmit only changes
Changes due to camera panning or zooming can be transmitted as vectors.

Lossy Compression Examples

Loud sounds mask quieter sounds so the quieter sounds can be ignored during presence of louder ones. People expect objects moving rapidly to be blurred --- these objects can use fewer bits in digital video transmission.

Vocoders
A vocoder (voice coder) is an example of lossy compression applied to human speech A typical vocoder reduces the amount of data that needs to be transmitted by constructing a model of the human vocal system

Model of Vocal Tract

Sound source
Voiced sounds: pulse generator with variable amplitude and frequency. Unvoiced sounds: noise generator with variable amplitude.

Filters
3 to 6 bandpass filters to represent the resonances in the vocal tract.

Vocoder Design
Start with excitation function. Follow up with parameters for a multi-pole filter. Transmit the parameters for excitation and filter at about 20 ms intervals. Vocoders like this are called linear predicted coders. Vocoders operate on a conventional PCM signal.

Vocoder Types
There are two main ways of generating the excitation signal in a linear predictive vocoder:
Pulse Excited Linear Predictive (PELP) Residual Excited Linear Predictive (RELP)

PELP Vocoder
Receiver has pulse generator and noise generator and receives parameters for them from transmitter.

PELP Vocoder

RELP Vocoder
Receiver has fixed filter Transmitter applies inverse of this filter to the voice signal to generate a residual signal. Transmitter then finds the closest signal to the residual in a codebook and transmits the code for that signal to the receiver. Receiver generates residual from code and applies it to filter.

Generation of Excitation Signal Using Codebook