Assessment - Source Coding Implementation Using MATLAB/Simulink

Unit Learning Outcomes covered in this assessment b. Apply and evaluate the principles used in the generation, transmission and reception of digitally modulated signals;
c. Report on the characteristics of sampling and analogue to digital conversion and source coding;
d. Distinguish the digital modulation and demodulation techniques, channel coding and decoding and their performance;
f. Apply the techniques of, and report on, digital communication applications using
matlab and hardware devices.

Assignment Description:

The purpose of this assignment is to motivate students to seek application of waveform coding techniques for digital audio transmission and digital audio recording. The assignment has two parts: Part A and Part B.

PART A: Implementation of µ-law and A-law Compander

Most modern communications systems operate in the digital domain for various reasons. However, most practical signals handled by these systems are naturally analogue in nature. Thus, an interface is needed between the signals and the systems handling them. This interface is called analogue-to-digital converter (ADC) and it's reciprocal, digital-to analogue converter (DAC). The three main functions of the ADC are sampling, quantization and encoding.

Requirements

1) Implement nonlinear quantizers µ-law and A-law compander in MATLAB/Simulink and compare the results. Matlab/Simulink code is required to be included in the report.

2) Prepare a report incorporating theories behind non-uniform quantization. Report should include comparison result and its interpretation from the simulation.

PART B: Source Coding in Digital Audio Systems

Digital audio has become an integral part of our lives. Nearly all of the transmission of speech signals over telephone channels is currently digital. Moreover, digital audio recording in the form of compact disks (CDs) has become very popular. This assignment exposes the student to real world applications of digital communication in digital audio transmission and recoding. Therefore, the student shall acquire skills to understand the real world requirements, critically review the existing techniques and write a technical report based on his or her investigation.

Requirements

The student needs to address the following three areas:

1) a) Requirement analysis: student is required to analyze the following for digital audio in telephone transmission systems:
- Bandwidth of speech signals and the required sampling rates to avoid aliasing
- Required number of bits per sample to achieve voice quality in telephony

b) In addition, student is also required to analyze the following for digital audio recording:
- Bandwidth of high quality audio and the required sampling rates
- Required number of bits per sample to achieve a high signal to quantization noise level for high quality audio.

2) Investigation of current methods: student is required to review and analyze waveform-coding techniques used in digital audio transmission and recording at present. At least PCM and DPCM methods need to be covered.

Attachment:- Assignment.rar