ECE 426-Digital Communications | Department of Electrical & Communication Engineering
PCM is a digital representation of an analog signal where the magnitude of the signal is sampled regularly at uniform intervals, then quantized to a series of symbols in a digital (usually binary) code. It is the standard form of digital audio, digital telephony, and other digital applications.
A continuous-time signal can be completely represented in its samples and recovered back if the sampling frequency satisfies:
Where fs is the sampling frequency and fm is the maximum frequency component in the analog signal.
The minimum sampling rate required to reconstruct the original signal without aliasing.
Occurs when fs < 2fm, causing spectral overlap and distortion. Anti-aliasing filters are used to prevent this.
Quantization is the process of mapping continuous input values to a finite set of discrete output values.
For a sinusoidal input with uniform quantization:
Each additional bit improves the SQNR by approximately 6 dB.
To improve the SQNR for small amplitude signals, companding is used:
Companding effectively makes the quantization steps smaller for small amplitudes and larger for large amplitudes.
Objective: Verify the Nyquist-Shannon sampling theorem and observe aliasing.
Objective: Analyze quantization error and verify the 6n dB rule.
Objective: Compare uniform vs. companded quantization.
Objective: Analyze the trade-off between quality and bandwidth.
Shows linear relationship between bits per sample and SQNR (6 dB per bit rule).
| Bits (n) | Levels (L) | Theoretical SQNR | Resolution (12V range) |
|---|---|---|---|
| 2 | 4 | 13.8 dB | 3.00 V |
| 4 | 16 | 25.8 dB | 0.80 V |
| 8 | 256 | 49.9 dB | 46.9 mV |
| 12 | 4096 | 74.0 dB | 2.93 mV |
| 16 | 65536 | 98.1 dB | 0.18 mV |
fs = 8 kHz, n = 8 bits
fs = 44.1 kHz, n = 16 bits, Stereo
Brief overview (150-200 words) covering objectives, methodology, key results (SQNR values, sampling rate verification), and conclusions.
Required Tables and Plots:
Summarize key findings, discuss sources of error (numerical precision in simulation, finite sampling), and state practical implications for digital audio/telephony system design.
List textbooks (e.g., Proakis, Sklar), IEEE papers on PCM, and ITU-T standards (G.711 for μ-law/A-law).