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EE 551/451, Fall, 2006 Communication Systems

EE 551/451, Fall, 2006 Communication Systems. Zhu Han Department of Electrical and Computer Engineering Class 13 Oct. 3 rd , 2006. Pulse Code Modulation (PCM). Pulse code modulation (PCM) is produced by analog-to-digital conversion process. Quantized PAM

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EE 551/451, Fall, 2006 Communication Systems

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  1. EE 551/451, Fall, 2006Communication Systems Zhu Han Department of Electrical and Computer Engineering Class 13 Oct. 3rd, 2006

  2. Pulse Code Modulation (PCM) • Pulse code modulation (PCM) is produced by analog-to-digital conversion process. Quantized PAM • As in the case of other pulse modulation techniques, the rate at which samples are taken and encoded must conform to the Nyquist sampling rate. • The sampling rate must be greater than, twice the highest frequency in the analog signal, fs > 2fA(max) • Telegraph time-division multiplex (TDM) was conveyed as early as 1853, by the American inventor M.B. Farmer. The electrical engineer W.M. Miner, in 1903. • PCM was invented by the British engineer Alec Reeves in 1937 in France. • It was not until about the middle of 1943 that the Bell Labs people became aware of the use of PCM binary coding as already proposed by Alec Reeves. EE 541/451 Fall 2006

  3. Pulse Code Modulation Figure The basic elements of a PCM system. EE 541/451 Fall 2006

  4. Encoding EE 541/451 Fall 2006

  5. Virtues, Limitations and Modifications of PCM Advantages of PCM 1. Robustness to noise and interference 2. Efficient regeneration 3. Efficient SNR and bandwidth trade-off 4. Uniform format 5. Ease add and drop 6. Secure DS0: a basic digitalsignaling rate of 64 kbit/s. To carry a typical phone call, the audio sound is digitized at an 8 kHz sample rate using 8-bit pulse-code modulation.4K baseband, 8*6+1.8 dB EE 541/451 Fall 2006

  6. Differential Encoding • Encode information in terms of signal transition; a transition is used to designate Symbol 0 Regeneration (reamplification, retiming, reshaping ) 3dB performance loss, easier decoder EE 541/451 Fall 2006

  7. Linear Prediction Coding (LPC) Consider a finite-duration impulse response (FIR) discrete-time filter which consists of three blocks : 1. Set of p ( p: prediction order) unit-delay elements (z-1) 2. Set of multipliers with coefficients w1,w2,…wp 3. Set of adders (  ) EE 541/451 Fall 2006

  8. Reduce the sampling rate Block diagram illustrating the linear adaptive prediction process. EE 541/451 Fall 2006

  9. Differential Pulse-Code Modulation (DPCM) Usually PCM has the sampling rate higher than the Nyquist rate. The encode signal contains redundant information. DPCM can efficiently remove this redundancy. 32 Kbps for PCM Quality EE 541/451 Fall 2006

  10. Processing Gain EE 541/451 Fall 2006

  11. Adaptive Differential Pulse-Code Modulation (ADPCM) Need for coding speech at low bit rates , we have two aims in mind: 1. Remove redundancies from the speech signal as far as possible. 2. Assign the available bits in a perceptually efficient manner. Adaptive quantization with backward estimation (AQB). EE 541/451 Fall 2006

  12. 8-16 kbps with the same quality of PCM ADPCM Adaptive prediction with backward estimation (APB). EE 541/451 Fall 2006

  13. Coded Excited Linear Prediction (CELP) • Currently the most widely used speech coding algorithm • Code books • Vector Quantization • <8kbps • Compared to CD 44.1 k sampling 16 bits quantization 705.6 kbps 100 times difference EE 541/451 Fall 2006

  14. Time-Division Multiplexing Figure Block diagram of TDM system. EE 541/451 Fall 2006

  15. DS1/T1/E1 • Digital signal 1 (DS1, also known as T1) is a T-carrier signaling scheme devised by Bell Labs. DS1 is a widely used standard in telecommunications in North America and Japan to transmit voice and data between devices. E1 is used in place of T1 outside of North America and Japan. Technically, DS1 is the transmission protocol used over a physical T1 line; however, the terms "DS1" and "T1" are often used interchangeably. • A DS1 circuit is made up of twenty-four DS0 • DS1: (8 bits/channel * 24 channels/frame + 1 framing bit) * 8000 frames/s = 1.544 Mbit/s • A E1 is made up of 32 DS0 • The line data rate is 2.048 Mbit/s which is split into 32 time slots, each being allocated 8 bits in turn. Thus each time slot sends and receives an 8-bit sample 8000 times per second (8 x 8000 x 32 = 2,048,000). 2.048Mbit/s • History page 274 EE 541/451 Fall 2006

  16. Synchronization • Super Frame EE 541/451 Fall 2006

  17. Synchronization • Extended Super Frame EE 541/451 Fall 2006

  18. T Carrier System Twisted Wire to Cable System EE 541/451 Fall 2006

  19. Fiber Communication EE 541/451 Fall 2006

  20. Delta Modulation (DM) EE 541/451 Fall 2006

  21. DM System: Transmitter and Receiver. EE 541/451 Fall 2006

  22. Slope overload distortion and granular noise The modulator consists of a comparator, a quantizer, and an accumulator. The output of the accumulator is EE 541/451 Fall 2006

  23. Slope Overload Distortion and Granular Noise ( differentiator ) EE 541/451 Fall 2006

  24. Delta-Sigma modulation (sigma-delta modulation) The modulation which has an integrator can relieve the draw back of delta modulation (differentiator) Beneficial effects of using integrator: 1. Pre-emphasize the low-frequency content 2. Increase correlation between adjacent samples (reduce the variance of the error signal at the quantizer input ) 3. Simplify receiver design Because the transmitter has an integrator , the receiver consists simply of a low-pass filter. (The differentiator in the conventional DM receiver is cancelled by the integrator ) EE 541/451 Fall 2006

  25. delta-sigma modulation system. EE 541/451 Fall 2006

  26. Adaptive Delta Modulation • Adaptive adjust the step size according to frequency, figure 6.21 • Out SNR • Page 286-287 • For single integration case, (BT/B)^3 • For double integration case, (BT/B)^5 • Comparison with PCM, figure 6.22 • Low quality has the advantages. • Used in walky-talky EE 541/451 Fall 2006

  27. Questions? EE 541/451 Fall 2006

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