Digital Telephony Digital Telephony 1 Analog/digital systems Analog signal Digital signal Discrete signal Analog SP A/D converter QuantizSampler er FsFmax Error is Analog signal introduced -voltage -speech -pressure or -tape data from -simulations -digital devices DSP D/A Digital Signal Processor -digital computer -dedicated dig. hw -programmable hw Digital signal Digital Telephony 2 Issues Reconstruction accuracy Conditions for perfect reconstruction Digital signal is not just an approx. representation of an analog signal Could be generated digitally The processing being performed may not be realizable in analog The theory of discrete time signal processing is independent of continuous Digital Telephony 3 Digital vs. analog processing DSP implementations are flexible, programmable and modular More precise and repeatable Performance and cost effectiveness (riding the microelectronics wave) Direct mapping of mathematical expressions with less approximation possible (enables sophisticated algorithms) Digital Telephony 4 Digital vs. analog ... Digital hardware can be multiplexed better than analog. Allows integration of multiple operations and services on a h/w platform Digital storage is more reliable, cheaper and more compact Digital Telephony 5 On the other hand Analog SP still offers higher bandwidth Higher dynamic range Can be very low power Digital Telephony 6 Analog to Digital Conversion To convert “real-world” analog signals to digital signals for processing Sampling Quantizing and coding Xa(t) Analog signal X [n] Sampler Quantizer and Coder Discrete signal Digital Telephony Xq[n] Digital signal 7 Sampling Uniform One sample every T seconds (ideally) x[n] = xa(nT), -n Sampling period: T Sampling frequency: Fs=1/T Assume: xa(t) = Acos( 2pFt+q) = Acos(Wt+q) Then: x[n] = Acos[2pFnT+q] = Acos[WTn+q] = Acos[w n+q], where w=WT is called the normalized or discrete domain frequency Digital Telephony 8 f = F/Fs must be rational in order for x[n] to be periodic If f = k/N, then x[n] is periodic with period N Now, xa(nT) = Acos(WTn+q) = Acos((W+2pk/T)Tn+q) is periodic in Wwith period 2p/T Also, x[n] = Acos[w n+q] = Acos[(w+2pk) n +q] is periodic in wwith period 2p Digital Telephony 9 xs(t) = xa(nT) d(t- nT) convert to n=- discrete sequence xa(t) x[n] = xa(nT) S(t) = d(t- nT) n=- Digital Telephony 10 Let us look at the continuous time Fourier transform of xs(t) 1 Xs(jW) = 2p Xa(jW) * S(jW) S(jW) = 2p T Xs(jW) = 1 T d(W-kWs) k=- Xa(jW-kjWs) k=- Digital Telephony 11 Xa(jW) must be bandwidth limited If the max frequency in Xa(jW) is WN, then the sampling rate Ws2WN ensures no information is lost due to aliasing This sampling rate is known as Nyquist rate A lower sampling rate causes a distortion of the signal due to Aliasing If no Aliasing occurs, the signal can be perfectly reconstructed by passing through an ideal low pass filter with Thus, Digital Telephony 12 Reconstruction Hr(jW) Xs(jW) -Wc Wc Ws>2WN Digital Telephony 13 Xr(jW) = Hr(jW) Xs(jW) if WNWc(Ws-WN) then Xr(jW) = Xc(jW) Reconstruction Frequency response of ideal reconstruction filter Hr(jW) = { T, WcWWc 0, otherwise T -Wc Wc Impulse response of ideal reconstruction filter sin pt/T hr(t)= pt/T Digital Telephony 14 Reconstruction Xr(jW) = Hr(jW) Xs(jW) xr(t) = xs(t) * hr(t) = [k=-xa(kT) d(t-kT)]*hr(t) k=-xa(kT) hr(t-kT) sin p(t-nT)/T k=-xa(kT) p(t-nT)/T = = Digital Telephony 15 xa(t) xs(t) hr(t) xr(t) Digital Telephony 16 Sampling theorem If the highest frequency contained in a signal xa(t) is W0 and the signal is uniformly sampled at a rate WsW0, then xa(t) can be exactly recovered from its sample values using the interpolation function sin pt/T hr(t)= pt/T and then xa(t) = k=-xa(kT) hr(t-kT), where {xa(kT) } are the samples of xa(t), and T=2p/Ws Digital Telephony 17 Quantization and coding Quantization: Converting discrete time signal to digital xq(n) =Q [x(n)] D Quantization step Digital Telephony 18 Q(x) 3D D D -D/ -7D/ -5D/ -3D/ -D D/ 3D/ 5D/ 7D/ x -D -3D -4D Digital Telephony 19 Quantization Rounding: Assign x[n] to the closest quantization level Quantization error eq[n] = xq[n] - x[n] -D/eq[n] D/ Uniformly distributed mean =0 variance = D/1 Digital Telephony 20 Quantization of quantizer: xmax-xmin Quantization levels: m Assuming uniform quantization Range xmax-xmin D= m-1 = Xm/ (m-1) where Xm = (xmax-xmin)/2 is called the full-scale level of the A/D converter Digital Telephony 21 Coding Coding is the process of assigning a unique binary number to each quantization level Number of bits required log2m Alternatively, given b+1 bits D=(xmax-xmin)/2b+1 =Xm /2b For A/D devices, the higher Fs and m, the less the error (and the more the cost of the device) Digital Telephony 22 x(n) xq(n) Quantizer x(n) xq(n) + eq(n) Assuming dynamic range of A/D converter is larger than signal amplitude 2 2 2 SNR = 10 log10(sx/se) = 10 log10(sx/(D/1)) 2 2 2b = 10 log10(12.2 sx/(Xm)) =6.02b +10.8 + 20 log10(Xm/sx) Digital Telephony 23 Signal to Quantiiation Noise Ratio (dB) Uniformly Encoded PCM 13 12 11 10 9 8 80 60 Number of bits per sample 40 20 -40 -30 -20 -10 0 dB X/Xm Digital Telephony 24 Example What is the minimum bit rate that a uniform PCM encoder must provide to encode a high fidelity audio signal with a dynamic range of 40 dB? Assume the fidelity requirements dictate passage of a 20-kHz bandwidth with a minimum signal-to-noise ratio of 50 dB. For simplicity, assume sinusoidal input signals. Digital Telephony 25 Companding Companded PCM with analog compression and expansion Compressed Digital Codewords Input Signal A/D Compression D/A Linear PCM Encoder F ( x) = sgn( x) Linear PCM Decoder ln( 1 x ) ln( 1 ) 1 -1 F ( y ) = sgn( y) [(1 )| y|-1 ] Output Signal Digital Telephony Expansion -1 x 1 -1 y 1 26 Segment Approximation Uniform quantization 111 110 101 100 011 010 001 000 Input Sample Values Digital Telephony 27 T1 Channel Bank 1 2 A/D T1 transmission Line Analog Inputs D/A 24 •Eigth bits per PCM code word •companding functions with mu=255 Digital Telephony 28 Performance of a 55Encoder Signal-to-quantization noise ratio (dB) 40 30 20 1 7 noise power = pi qi2 12 i =0 Piecewise linear 8 bit 255 33 27 22 8 bit 255 7 bit 100 10 -70 -60 -50 -40 -30 -20 -10 Signal Power of sinewave (dBm0) Digital Telephony 0 3 dB 29 Total Noise Power 40 30 Pe = 10 -8 20 1 Pe =010 -6 -70 -60 -50 40 dB range of possible signals Signal-to-total noise noise ratio Pe = 0 30 dB required for good communication Pe = 10 -5 Pe = 10 -3 15 dB at which persons fin communication difficult Pe = 10 -4 -40 -30 -20 -10 0 3 Signal Power relative to full-load signal (dBm0) Digital Telephony dB 30 Error Performance Fewer than 10% of 1 min intervals to have BER worse than 10E-6 Fewer than 0.2% of 1 sec intervals to have BER worse than 10E-3 92% error free sec Digital Telephony 31 DS1 Signal Format bits in 125 s 1 = 125 10 -6 8000 193 x 8000 = 1.544 Mbs Bit “robbing” technique used on each sixth frame to provide signaling information (8x24)+1=193 Digital Telephony 32 Plesiochronous Transmission Rates Japanese Standard North America Standard 97728 kbits/s European Standard 564992 kbits/s x4 x4 97728 kbits/s 274176 kbits/s x3 x6 32064 kbits/s 139264 kbits/s x4 x3 44736 kbits/s x5 x4 x7 6312 kbits/s x4 1544 kbits/s x24 34368 kbits/s 8448 kbits/s x3 2048 kbits/s x4 x30 64 kbits/s Digital Telephony 33 Plesiochronous Digital Hierarchy MULTIP LEXING LEVELS (DS) # OF VOICE NORTH EUROPE JAPAN CHANNELS AMERICA 0 1 0.064 1 24 1.544 30 2 (4xDS1) 0.064 0.064 1.544 2.048 48 3.152 3.152 96 6.312 6.312 120 8.448 Digital Telephony 34 Multiplexing # OF VOICE Levels CHANNELS 480 3 (7xDS2) NORTH AMERICA 672 44.376 1344 91.053 EUROPE JAPAN 34.368 32.064 1440 4 (6xDS3) 97.728 1920 4032 139.264 274.176 5760 397.200 7680 565.148 Digital Telephony 35 Plesiochronous Digital Hierarchy The output of the M12 multiplexer is operating 136 kbs faster than the agragate rate of four DS1 6.312 vs 4x1.544=6.176 M12 frame has 1176 bits, i.e. 294-bit subframes ; each subframe is made of up of 49-bits blocks; each block starts with a control bit followed by a 4x12 info bits from four DS1 channels Digital Telephony 36 Makeup of a DS2 Frame M1 01 02 03 04 C1 01 02 03 04 F0 01 02 03 04 C2 01 02 03 04 C3 01 02 03 04 F1 01 02 03 04 Bit stuffing M1 01 02 03 04 C1 01 02 03 04 F0 01 02 03 04 C2 01 02 03 04 C3 01 02 03 04 F1 01 02 03 04 4 M bits (O11X X=0 alarm) C=000,111 bit stuffing present/absent nominal stuffing rate 1796 bps, max 5367 Digital Telephony 37 Regenerative Repeaters Amplifier Equalizer Input Spacing Pair diameter (mm) Regenerator Output Timing recovery between adjacent repeaters 0.9 Loop attenuation at 1 MGHz (dB/km) 12 Loop Maximum Total Max resistance distance Repeaters distance (km) system (W/km) (km) 60 3 18 54 0.8 16 100 2.25 Digital Telephony 16 36 38 Digital Transmission Systems Designati Administra Bit on tion Rate Line Code Media T1 AT&T 1.544 AMI/B8ZS Twisted pair 6000 ft CEPT1 CCITT 2.048 B4ZS Twisted pair 2000 m T1C AT&T 3.152 Bipolar Twisted pair 6000 ft T148 ITT 4B3T Twisted pair 6000 ft 9148A GTE 2.37 ternary 3.152 Twisted pair 6000 ft T1D AT&T 3.152 T1G AT&T 6.443 1-DD duobinary 1+D duobinary 4-level LD-4 Canada 274.176 B3ZS Coax 1900 m T4M AT&T 274.176 Polar Coax 5700 ft Digital Telephony Repeater Spacing Twisted pair 6000 ft Twisted pair 6000 ft 39 PCM System Enhancements North America Superframe of 12 DS0’s has a sync sequence 101010 for odd (001110 for even frames) Extended superframe 24 frames - (4 S bits for frame allignment signal); 6 S bits for CRC-6 check; the rest 12 constitute 4 kbs data link Digital Telephony 40