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Linear Predictive Coding

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Linear Predictive Coding
Linear Predictive Coding
Jeremy Bradbury December 5, 2000

0 Outline
I. II. Proposal Introduction A. Speech Coding B. Voice Coders C. LPC Overview III. Historical Perspective of Linear Predictive Coding A. B. C. IV. V. VI. History of Speech & Audio Compression History of Speech Synthesis Analysis/Synthesis Techniques

Human Speech Production LPC Model LPC Analysis/Encoding A. B. C. D. E. Input speech Voice/Unvoiced Determination Pitch Period Estimation Vocal Tract Filter Transmitting the Parameters

VII. VIII.

LPC Synthesis/Decoding LPC Applications A. B. C. D. Telephone Systems Text-to-Speech Synthesis Voice Mail Systems Multimedia

IX. X.

Conclusion References

1

1 Proposal
Linear predictive coding(LPC) is defined as a digital method for encoding an analog signal in which a particular value is predicted by a linear function of the past values of the signal. It was first proposed as a method for encoding human speech by the United States Department of Defence in federal standard 1015, published in 1984. Human speech is produced in the vocal tract which can be approximated as a variable diameter tube. The linear predictive coding (LPC) model is based on a mathematical approximation of the vocal tract represented by this tube of a varying diameter. At a particular time, t, the speech sample s(t) is represented as a linear sum of the p previous samples. The most important aspect of LPC is the linear predictive filter which allows the value of the next sample to be determined by a linear combination of previous samples. Under normal circumstances, speech is sampled at 8000 samples/second with 8 bits used to represent each sample. This provides a rate of 64000 bits/second. Linear predictive coding reduces this to 2400 bits/second. At this reduced rate the speech has a distinctive synthetic sound and there is a noticeable loss of quality. However, the speech is still audible and it can still be easily understood. Since there is information loss



References: [1] [2] V. Hardman and O. Hodson. Internet/Mbone Audio (2000) 5-7. Scott C. Douglas. Introduction to Adaptive Filters, Digital Signal Processing Handbook (1999) 7-12. Poor, H. V., Looney, C. G., Marks II, R. J., Verdú, S., Thomas, J. A., Cover, T. M. Information Theory. The Electrical Engineering Handbook (2000) 56-57. R. Sproat, and J. Olive. Text-to-Speech Synthesis, Digital Signal Processing Handbook (1999) 9-11 . Richard C. Dorf, et. al.. Broadcasting (2000) 44-47. Richard V. Cox. Speech Coding (1999) 5-8. Randy Goldberg and Lance Riek. A Practical Handbook of Speech Coders (1999) Chapter 2:1-28, Chapter 4: 1-14, Chapter 9: 1-9, Chapter 10:1-18. Mark Nelson and Jean-Loup Gailly. Speech Compression, The Data Compression Book (1995) 289-319. Khalid Sayood. Introduction to Data Compression (2000) 497-509. Richard Wolfson, Jay Pasachoff. Physics for Scientists and Engineers (1995) 376-377. [3] [4] [5] [6] [7] [8] [9] [10] 22

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