Communication PCM
Communication PCM
Sampling
Quantizing
Encoding
a) Uniform Quantizer
b) Quantization Error
c) Quantized PAM signal output
3. PCM signal is obtained from the quantized PAM signal by encoding each
quantized sample value into a digital word.
Analog to Digital Conversion
The Analog-to-digital Converter (ADC)
performs three functions:
Sampling process
Analog Makes the signal discrete in time.
Input If the analog input has a bandwidth of
Signal W Hz, then the minimum sample
frequency such that the signal can be
Sample reconstructed without distortion.
Quantization process
ADC Makes the signal discrete in amplitude.
Quantize Round off to one of q discrete levels.
111
110
101
100
Encoding process
011
010 Maps the quantized values to digital
Encode
001
000
words that are n bits long.
1
optimal when the input
-8 -6 -4 -2 -1 2 4 6 8 distribution is uniform. When
-3
Input sample X
all values within the Dynamic
Range of the quantizer are
equally likely.
-5
-7
Quantization Characteristic
Analog signal
Sampling
Quantization levels.
Quantized to 5-levels
Quantization levels
Quantized 10-levels
PCM encoding example
M=8
M 2n , n log 2 ( M )
Encoding is the process of representing each quantized
sample by an n bit code word.
The mapping is one-to-one so there is no distortion introduced by
encoding.
Some mappings are better than others.
A Gray code gives the best end-to-end performance.
M 2n n log 2 ( M )
M is the number of Quantization levels
(c) Error Signal n is the number of bits per sample
PCM
Practical PCM Circuits
The spectrum of the PCM signal is not directly related to the spectrum of the
input signal.
The bandwidth of (serial) binary PCM waveforms depends on the bit rate R
and the waveform pulse shape used to represent the data.
The Bit Rate R is
R=nfs
Where n is the number of bits in the PCM word (M=2n) and fs is the sampling
rate.
For no aliasing case (fs≥ 2B), the MINIMUM Bandwidth of PCM Bpcm(Min) is:
The Minimum Bandwidth of nfs//2 is obtained only when sin(x)/x pulse is used
to generate the PCM waveform.
Bpcm = R = nfs
PCM Noise and Companding
Quantization Noise
Signal to Noise Ratio
PCM Telephone System
Nonuniform Quantization
Companding
Quantization Noise
Average Power{ X }
( SNR)Q
Average Power{nQ }
Effect of Noise in PCM
Two main effects produce the noise or distortion in the PCM output:
Quantizing noise that is caused by the M-step quantizer at the PCM transmitter.
Bit errors in the recovered PCM signal, caused by channel noise and improper filtering.
If the input analog signal is band limited and sampled fast enough so that the
aliasing noise on the recovered signal is negligible, the ratio of the recovered
analog peak signal power to the total average noise power is:
The ratio of the average signal power to the average noise power is
If Pe is negligible, there are no bit errors resulting from channel noise and no ISI,
the Peak SNR resulting from only quantizing error is:
Where, M = 2n
α = 4.77 for peak SNR
α = 0 for average SNR
DESIGN OF A PCM SIGNAL FOR TELEPHONE SYSTEMS
Assume that an analog audio voice-frequency(VF) telephone signal occupies a band from 300
to 3,400Hz. The signal is to be converted to a PCM signal for transmission over a digital
telephone system. The minimum sampling frequency is 2x3.4 = 6.8 ksample/sec.
To be able to use of a low-cost low-pass antialiasing filter, the VF signal is oversampled with
a sampling frequency of 8ksamples/sec.
This is the standard adopted by the Unites States telephone industry.
Assume that each sample values is represented by 8 bits; then the bit rate of the binary
PCM signal is
This 64-kbit/s signal is called a DS-0 signal (digital signal, type zero).
The minimum absolute bandwidth of the binary PCM signal is
R nf s
BPCM
2 2
This B is for a sinx/x type pulse sampling
DESIGN OF A PCM SIGNAL FOR TELEPHONE SYSTEMS
If we use a rectangular pulse for sampling the first null bandwidth is given by
We require a bandwidth of 64kHz to transmit this digital voice PCM signal, whereas the
bandwidth of the original analog voice signal was, at most, 4kHz.
Note:
1. Coding with parity bits does NOT affect the quantizing noise,
2. However coding with parity bits will improve errors caused by channel or ISI,
which will be included in Pe ( assumed to be 0).
Nonuniform Quantization
Many signals such as speech have a nonuniform distribution.
The amplitude is more likely to be close to zero than to be at higher levels.
Nonuniform quantizers have unequally spaced levels
The spacing can be chosen to optimize the SNR for a particular type of signal.
Output sample
XQ 6
-8 -6 -4 -2 2 4 6 8
-2
Input sample
X
-4
-6
Companding
Nonuniform quantizers are difficult to make and expensive.
An alternative is to first pass the speech signal through a
nonlinearity before quantizing with a uniform quantizer.
l n (1 | x ( t )|)
| y ( t ) |
ln (1 )
0 1
Input |x(t)|
Non Uniform Quantizing
Voice signals are more likely to have amplitudes near zero than at extreme
peaks.
For such signals with non-uniform amplitude distribution quantizing noise
will be higher for amplitude values near zero.
A technique to increase amplitudes near zero is called Companding.
x x’ x’ y
Q(.)
C(.)
Compressor Uniform Quantizer
Example: µ-law Companding
x[n]=speech /song/
0 .5
-0 .5
-1
0 1 0 0 0 2 0 0 0 3 0 0 0 4 0 0 0 5 0 0 0 6 0 0 0 7 0 0 0 8 0 0 0 9 0 0 0 1 0 0 0 0
0 .5
y[n]=C(x[n]) 0
Companded Signal -0 .5
-1
0 1 0 0 0 2 0 0 0 3 0 0 0 4 0 0 0 5 0 0 0 6 0 0 0 7 0 0 0 8 0 0 0 9 0 0 0 1 0 0 0 0
0 .5
Close View of the Signal
Segment of x[n] 0
-0 .5
-1
2 2 0 0 2 3 0 0 2 4 0 0 2 5 0 0 2 6 0 0 2 7 0 0 2 8 0 0 2 9 0 0 3 0 0 0
Segment of y[n]
0 .5
Companded Signal -0 .5
-1
2 2 0 0 2 3 0 0 2 4 0 0 2 5 0 0 2 6 0 0 2 7 0 0 2 8 0 0 2 9 0 0 3 0 0 0
A-law and µ-law Companding
These two are standard companding methods.
u-Law is used in North America and Japan
A-Law is used elsewhere to compress digital telephone signals
SNR of Compander
The output SNR is a function of input signal level for uniform quantizing.
The output SNR is a function of input signal level for uniform quantizing.
But it is relatively insensitive for input level for a compander.
where, V is the peak signal level and xrms is the rms value
7 bits of the 8 bit PCM are used to get a data rate of 56kb/s (
Frequencies below 300Hz are omitted to get rid of the power line noise
in harmonics of 60Hz).
If SNR is below 52dB the modem will fallback to lower speeds ( 33.3
kbps, 28.8kbps or 24kbps).
Differential Pulse Code Modulation (DPCM)
Figure 1: A differential PCM (DPCM) encoder. Figure 2: A differential PCM (DPCM) decoder.
The prediction is based upon the quantized The prediction (based on past quantized
prediction error eˆ(n) together with past predictions outputs) is added to the received error signal
s˜(n). eˆ(n) to generate each output sˆ(n).
References