Pulse Code Modulation
Pulse Code Modulation
PCM
PCM is a digital scheme for transmitting analog data. The signals in PCM are binary; that is, there are only two possible states, represented by logic 1 (high) and logic 0 (low)
PCM
n M=2
M number of Quantization levels n number of bits per sample
PCM
SAMPLING
generation of flat top PAM signal
PCM
QUANTIZING
converting each sample value to a finite number of levels so that digital words with amplitude closest to the actual sampled value is used When sampling at NYQUIST RATE(2B) or faster and there is negligible noise, there will still be QUANTIZING NOISE Will have an output called QUANTIZED PAM signal
PCM
ENCODING PCM is obtained from quantized PAM signal by converting each sample value into digital word Gray code is used as binary code word
PCM
PCM
PCM
PCM
a. Low-pass Filter limits analog input signal to standard frequency range b. Instantaneous sampler and hold periodically samples signals and converts them to a multilevel PAM signal c. Converts PAM to parallel PCM codes
PCM
Parallel PCM codes are converted to serial binary data and then outputted to the transmission line as digital pulses Trans line repeaters are placed at a prescribed distances to regenerate digital pulses
PCM
- Converts serial pulses from transmission line to parallel PCM codes - Converts parallel PCM codes to multilevel PAM signals - Hold circuit is basically Low-Pass filter that converts PAM back to its original form
PCM
Three popular techniques are used to implement the analog-todigital converter (ADC) encoding operation: The counting or ramp, ( Maxim ICL7126 ADC) Serial or successive approximation, (AD 570) Parallel or flash encoders. ( CA3318) The objective of these circuits is to generate the PCM word. Parallel digital output obtained (from one of the above techniques) needs to be serialized before sending over a 2-wire channel This is accomplished by parallel-to-serial converters [Serial Input-Output (SIO) chip] UART, USRT and USART are examples for SIOs
PCM
Where
R=nfs
n is the number of bits in the PCM word (M=2n) fs is the sampling rate
PCM
PCM
Bpcm = R = nfs
PCM
QUANTIZATION NOISE
PCM
PCM
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
M is the number of quantized levels used in the PCM system. Pe is the probability of bit error in the recovered binary PCM signal at the receiver DAC before it is converted back into an analog signal.
PCM
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
= 4.77 for peak SNR = 0 for average SNR
PCM
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
BPCM
R nf s 2 2
PCM
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. 2. Coding with parity bits does NOT affect the quantizing noise, 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
PCM
2 4 6
Input sample X
COMPANDING
An alternative is to first pass the speech signal through a nonlinearity before quantizing with a uniform quantizer. The nonlinearity causes the signal amplitude to be Compressed.
The input to the quantizer will have a more uniform distribution.
PCM
At the receiver, the signal is Expanded by an inverse to the nonlinearity. The process of compressing and expanding is called Companding.
PCM
-LAW COMPANDING
1
0
Input |x(t)|
PCM
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.
Effect of non linear quantizing can be can be obtained by first passing the analog signal through a compressor and then through a uniform quantizer.
x
C(.) Compressor
Q(.)
Uniform Quantizer
PCM
10000
1 0.5 0 -0.5 -1 0 1000 2000 3000 4000 5000 6000 7000 8000 9000 10000
Segment of x[n]
Segment of y[n] Companded Signal
1 0.5 0 -0.5
2300
2400
2500
2600
2700
2800
2900
3000
PCM
The output SNR is a function of input signal level for uniform quantizing.
But it is relatively insensitive for input level for a compander
SNR of Compander
PCM
PCM
The output SNR is a function of input signal level for uniform quantizing.
But it is relatively insensitive for input level for a compander.
for Uniform Quantizer V is the peak signal level and xrms is the rms value
for A-law companding
= 4.77 - 20 log[1 + Ln A]
PCM
The V.90 PC Modem transmits data at 56kb/s from a PC via an analog signal on a dial-up telephone line. A law compander is used in quantization with a value for of 255. The modem clock is synchronized to the 8-ksample/ sec clock of the telephone company. 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). SNR of the line should be at least 52dB to operate on 56kbps. If SNR is below 52dB the modem will fallback to lower speeds ( 33.3 kbps, 28.8kbps or 24kbps).
PCM
GROUP MEMBERS:
KEVIN CEA ALJOHN VALDEZ JAY ANGELO BALANA RUSSEL FERNANDO JR.