Faculty of EngineeringElectrical Engineering Department

Communication II Lab (EELE 4170)

BASEBAND & BANDPASS MODULATION In BASEBAND MODULATION

waveform usually take the form of shaped pulses.

In BANDPASS MODULATION

the shaped pulses modulate a sinusoid called a carrier wave

PCM (Pulse code modulation) ( Baseband modulation )

PCM modulation is a kind of source coding.

The meaning of source coding is the conversion from analog signal to digital signal.

PCM modulation is commonly used in audio and telephone transmission.

PCM in Wired Telephony

Voice circuit bandwidth is 3400 Hz.

Sampling rate is 8 KHz (Ts=125 µs).

Each sample is quantized to one of 256 levels.

Each quantized sample is coded into a 8-bit word.

The 8-bit words are transmitted serially (one bit at a time) over a digital transmission channel.

The bit rate is: 8(bits/sample)*(8K sample/sec)=64000 bit/sec

Block diagram of PCM modulation

LPF (low pass filter )

which is used to remove the noise in the audio signal

Sampler

audio signal will be sampled to obtain a series of sampling values

Quantizer

the signal will pass through it to quantize the sampling values

Encoder

Then the signal will pass through it to encode the quantization values and then convert to digital signal.

Encoder produce information as parallel form , it must convert to serial form (Why?)

The implementation of PCM modulator

FS0 and FS1 are the data format selection of PCM encoder as shown in table

Block diagram of PCM demodulation

Comparator During transmission, the PCM signal is hardto avoid the noise interference. Therefore, before the PCMsignal sends into the PCM demodulator, we utilize acomparator to recover the signal to the original level

Serial to parallel converter

Decoder

wants digital bits parallel form to produce analoginformation.

S/H sample and hold (convert analog value to analogsignal) at the same clock of comparator.

LPF (low pass filter ) to remove the unwanted signal at thefinal part

The implementation of PCM demodulator

Uniform

Quantization

Non-Uniform

Uniform Quantization When the quantization levels are uniformly distribute over

the full range (step size is equal for all levels) then the quantizer is called a uniform or linear quantizer.

A system (with uniform quantization) would be use wasteful for speech signals.

With uniform quantization the (SNR) is worse for law signals than for high level signals.(Why ?)

Because many of the quantized steps would rarely be used.

Uniform Quantization

Non-Uniform To provide fine quantization of weak signals .

(SQNR)= signal to quantization noise ratio .

To improve SQNR by reducing the noise of quantizer .

Can be used to make the SNR a constant for all signals within the input signal .

The standard telephone technique of handling the large range of possible input signal levels is. a logarithmic- compressed quantizer

Non-Uniform

Non-uniform quantization is achieved by first distorting the original signal with a logarithmic compression characteristic. And then using uniform quantizer .

After compression the distorted signal is used as the input to a uniform (linear quantizer)

At receiver , an inverse compression characteristic , called expansion is applied so that overall transmission is not distorted .

The processing pair (compression and expansion) is usually referred to as companding .

compressionlinear

quantizerexpansion

In North America , a µ-law compression characteristic

Another compression characteristic, used mainly in Europe , is the A-law compression characteristic.

For linear quantizer µ = 0 , A = 1 .

In North America the stander value for µ = 255 . For several values a stander value of A = 87.6

HW#1 Your report .

Write Matlab code for Nonuniform Quantization?

( use quantization function code and compression µ-law)

( µ = 255 , levels = 4 , x(t)= sin(2πt) )

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