CS 4594 Data Communications PCM

Pulse Code Modulation PCM = Pulse Code Modulation Converts analog signal to digital signal Analog signal is limited (filtered) to a range of frequencies Samples the analog signal at regular intervals Each sample is converted to a number that represents its amplitude May use non-linear scale or differential coding

PCM and the Telephone System Limits signal to less than 3400 Hz Samples at 8000 Hz (Nyquist says to sample at least twice the frequency) Uses 12 bits to digitize samples Compresses this to 7 or 8 bits per sample (depending on exact method, ulaw or Alaw) This is 56 to 64 kbits/sec called DS0

Steps of Telephone PCM voice resulting voice signal Linearly quantize to 12 bits band limit frequency to <= 3400 Hz Sample 8000 per sec Compress to 8 bits per sample voice Trans- mission line resulting voice signal Filter (fills in between pulses) Pulse generation (amplitude of pulses match digital values) Decompress (back to 12-bit)

-Law, A-law Companding Companding methods Companding = compressing/expanding Samples lower amplitudes more closely than higher amplitudes -law used in North America, etc. A-law used in Europe, etc.

Companding A-law -law Output signal Input signal

Variations on PCM Differential PCM - send difference in quantized values as 2 or 3 bit value Adaptive DPCM - varies quantization levels according to rate of change of input signal ITU-T has ADPCM standard for 4 kHZ voice at 40, 32, 24, and 16 kbits/sec. ADPCM standard for 7 kHZ voice at 64 kbits/sec