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Digital Signal Processing with DSP Hardware

Lecturer

Dr. Maciej Gratkowski

Schedule

Lecture Tu. 08:15 - 09:45 Z 613

Lab Session

Wed.

10:00 - 13:15

Z 613

In case of overlap with other courses of participants, we will try to move the time slots to ones that suit everyone.

LSF-Link, Course SVN-Repository

Learning Content

Code Composer Studio

  • Introduction to DSP Development Systems and ANSI C
  • Analog Input/Output with DSP, Sampling of Signals
  • Periodic Signal Generation
  • Sample-Based and Frame-Based DSP
  • FIR/IIR Filters
  • Spectral Analysis and Windowing
  • Adaptive Filters
  • Code Optimization
  • DSP BIOS and Real-Time Data Exchange

Prerequisites

Ideally the student have already taken classical Digital Signal Processing Course (INF-10150-20132) but it is not a prerequisite. This course can be taken independently or in case of special interests in DSP both courses could be also taken parallel. Basic knowledge of C programming language is an advantage.

Learning Objectives

The student after the course:DSP Lab Equipment

  • can design a digital signal processing algorithm and run it in real-time on a DSP hardware
  • is be able to design and implement filters that for example reduce noise or realize various audio effects
  • is able to use an oscilloscope to investigate time-varying signals
  • is able to converse with DSP experts and to understand some of the jargon, enabling to learn more about the subject over his career
  • understands the tradeoffs in the practical design of digital filters including how data types affect memory usage, execution speed, and roundoff error and the pro's and con's of two common types of digital filters, namely FIR and IIR filters

Credit Requirements and Work

The final grade will be based on the laboratory problems and projects. The course will be flexibly split into lecture and lab session parts. Most of the student work will be done during the lab sessions using the C6713 DSK. There will be no homework assignments.

Example laboratory problems:

  • Implementation of various audio effects (echo, reverb, stereo enhancement etc.)
  • Generation and detection of DTMF tones (Goertzel algorithm)
  • Implementation of real-time FIR and IIR filters
  • Implementation of Fast Fourier Transform (FFT)
  • Implementation of tripple-buffering scheme for signal processing
  • Time-domain and frequency-domain convolution
  • Implementation of LMS algorithm for adaptive filtering

Example student projects:

  • Voice detection and reverse playback
  • 3D sound using head related transfer functions
  • Audio transmission using amplitude modulation and demodulation
  • Acoustic direction tracking

Literature

  • Rulph Chassaing, Donald Reay, Digital Signal Processing and Applications with the TMS320C6713 and TMS320C6416 DSK, 2nd Edition, May 2008, Wiley-IEEE Press - Link
  • Thad B. Welch, Cameron H.G. Wright, Michael G. Morrow, Real-Time Digital Signal Processing from MATLAB to C with the TMS320C6x DSPs, Second Edition, CRC Press, Inc., 2012 - Link
  • Rulph Chassaing, Digital Signal Processing: Laboratory Experiments Using C and the TMS320C31 DSK, Wiley-Interscience, 1998 - Link
  • Joshua D. Reiss, Andrew McPherson, Audio Effects, CRC Press, Inc., 2015 - Link