This course emphasizes processing of the human speech waveform, primarily using digital techniques. Theory of speech production and speech perception as related to signals in time and frequency-domains is covered, as well as the measurement of model parameters, short-time Fourier spectrum, and linear predictor coefficients. Speech coding, recognition, speech synthesis, and speaker identification are discussed. Application areas include telecommunications telephony, Internet VOIP, and man-machine interfaces. Considerations for embedded realization of the speech processing system will be covered as time permits. Several application-oriented software projects will be required.
There are multiple aspects to speech processing: speech analysis, speech compression for storage and transmission, noise cancellation and speech enhancement, and the various types of recognition applications.
A review of basic DSP is useful; we shall examine time-based and frequency-based analyses, as well as time-frequency based analyses. We shall also cover the basics of stochastic signal processing; this will necessitate a review of random processes.
We shall also cover the practical aspects of speech processing. We shall perform some hands-on work.
The class will select an overall, collaborative project-paper. The problem addressed shall be selected after a class discussion.
We shall be leveraging MATLAB for much of our experimental work. It is suggested that download the freeware tool Audacity also.
We shall be performing some in-class labs.
The following topics are planned to be covered:
1. Introduction to the Speech Processing Problem Set
2. Signals & Transforms in General
3. Signal Processing Fundaments & its Friends
4. Introduction to Sound and Acoustics
5. Human Speech Production & Perception Chain
6. Speech analysis: spectral analysis
7. Speech Coding: quantization and source coding [waveform and parametric]
8. Recognition Techniques (including Robust Recognition)
9. Hardware considerations
10. Impact
The course has three distinct goals:
1. Review and enhancement of digital signal processing techniques; while focused on speech signals, the broader application of these techniques will be presented and measured.
2. Provide the student with the collaborative skills to address and solve a TBD problem.
3. Improve the confidence of the students in communicating technical problems, approaches, solutions, assumptions and risks.
There are no required textbooks. However, I do have two recommended texts: Discrete-Time Speech Signal Processing by T. Quatieri for the speech analysisa nd parametric speech coding [ISBN 0-13-242942-x] and Deep Learning for NLP and Speech Recognition by Kamath, et al. [ISBN 978-3-030-14595-9
I will try to supply the relevant IEEE and other refereed papers for reference to obviate the need for the texts; however, they are quite good.
MATLAB (or equivalent) will be used.
1. Three short technical summaries of three technical papers provided to the student [20%]
2. Three online quizzes [10%]
3. One mini-project to investigate some aspect of the speech signal [collaborative] [30%]
4. Class selected research project [40%]
Score Range | Letter Grade |
---|---|
100-97 | = A+ |
96-93 | = A |
92-90 | = A− |
89-87 | = B+ |
86-83 | = B |
82-80 | = B− |
79-77 | = C+ |
76-73 | = C |
72-70 | = C− |
69-67 | = D+ |
66-63 | = D |
<63 | = F |
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