Completed fixed codebook for speech encoder

Data processing: speech signal processing – linguistics – language – Speech signal processing – For storage or transmission

Reexamination Certificate

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C704S201000, C704S200000

Reexamination Certificate

active

06173257

ABSTRACT:

INCORPORATION BY REFERENCE
The following applications are hereby incorporated herein by reference in their entirety and made part of the present application:
1) U.S. Provisional Application Ser. No. 60/097,569, entitled “Adaptive Rate Speech Codec,” filed Aug. 24, 1998;
2) U.S. patent application Ser. No. 09/154,675, entitled “Speech Encoder Using Continuous Warping In Long Term Preprocessing,” filed Sep. 18, 1998;
3) U.S. patent application Ser. No. 09/156,649, entitled “Comb Codebook Structure,” filed Sep. 18, 1998;
4) U.S. patent application Ser. No. 09/156,648, entitled “Low Complexity Random Codebook Structure,” filed Sep. 18, 1998;
5) U.S. patent application Ser. No. 09/156,650, entitled “Speech Encoder Using Gain Normalization That Combines Open And Closed Loop Gains,” filed Sep. 18, 1998;
6) U.S. patent application Ser. No. 09/156,832, entitled “Speech Encoder Using Voice Activity Detection In Coding Noise,” filed Sep. 18, 1998;
7) U.S. patent application Ser. No. 09/154,654, entitled “Pitch Determination Using Speech Classification And Prior Pitch Estimation,” filed Sep. 18, 1998;
8) U.S. patent application Ser. No. 09/154,657, entitled “Speech Encoder Using A Classifier For Smoothing Noise Coding,” filed Sep. 18, 1998;
9) U.S. patent application Ser. No. 09/156,826, entitled “Adaptive Tilt Compensation For Synthesized Speech Residual,” filed Sep. 18, 1998;
10) U.S. patent application Ser. No. 09/154,662, entitled “Speech Classification And Parameter Weighting Used In Codebook Search,” filed Sep. 18, 1998;
11) U.S. patent application Ser. No. 09/154,653, entitled “Synchronized Encoder-Decoder Frame Concealment Using Speech Coding Parameters,” filed Sep. 18, 1998;
12) U.S. patent application Ser. No. 09/154,663, entitled “Adaptive Gain Reduction To Produce Fixed Codebook Target Signal,” filed Sep. 18, 1998;
13) U.S. patent application Ser. No. 09/154,660, entitled “Speech Encoder Adaptively Applying Pitch Long-Term Prediction and Pitch Preprocessing With Continuous Warping,” filed Sep. 18, 1998.
BACKGROUND
1. Technical Field
The present invention relates generally to speech encoding and decoding in mobile cellular communication networks; and, more particularly, it relates to various techniques of using sub-codebooks for pulse-like excitation in speech reproduction through a limited bit rate communication channel.
2. Related Art
Signal modeling and parameter estimation play significant roles in communicating voice information with limited bandwidth constraints. To model basic speech sounds, speech signals are sampled as a discrete waveform to be digitally processed. In one type of signal coding technique, called linear predictive coding (LPC), the signal value at any particular time index is modeled as a linear-function of previous values. A subsequent signal is thus linearly predictable according to an earlier value. As a result, efficient signal representations can be determined by estimating and applying certain prediction parameters to represent the signal.
In speech encoding and decoding, it is well-known that pulse-like excitation provides better quality than noise-like excitation for voiced speech. Previously, exclusively pulse-like excitation was used with ACELP (Adaptive Code Excited Linear Predictive) systems in which codebooks with fixed numbers of pulses, fixed pulse position resolution and fixed pulse magnitude was utilized. Nevertheless, ACELP systems did not work well for certain types of speech signals.
The present invention addresses these problems by recognizing that, depending on the circumstances, either the number of pulses or the pulse position resolution may be more important. Accordingly, sub-codebooks are designed in such a way that either frequency of pulses or pulse resolution can be emphasized.
Further limitations and disadvantages of conventional systems will become apparent to one of skill in the art after reviewing the remainder of the present application with reference to the drawings.
SUMMARY OF THE INVENTION
Various aspects of the present invention can be found in a speech encoding system using an analysis by synthesis coding approach on a speech signal. The speech encoder comprises a plurality of codebooks comprising a plurality of codevectors, with each of the codevectors comprising at least a first and second pulse index. The speech encoder also comprises an encoder processing circuit, coupled to the first codebook, that identifies one codevector from the plurality of codevectors by considering the first pulse index from each of the plurality of codevectors before considering the second pulse index from any of the plurality of codevectors.
In other embodiments of the invention, the encoder processing circuit considers at least a portion of the pulses of the second pulse index after considering each pulse of the first pulse index or reconsiders a portion of the pulses of the first pulse index after considering at least a portion of the pulses of the first pulse index.
The speech encoder may also select one of the codebooks for further consideration after considering at least two of the plurality of codebooks. A weighting factor may also be applied in selecting one of the codebooks.
Further aspects of the present invention can be found in a method for searching a fixed codebook having a codevector that defines a plurality of pulses. The method comprises locating then fixing a pulse position for at least one of a plurality of pulses and then locating a pulse position for at least one other of the plurality of pulses.
Other aspects, advantages and novel features of the present invention will become apparent from the following detailed description of the invention when considered in conjunction with the accompanying drawings.


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Sridha Sridhan & John Leis, “Two Novel Lossless Algorithms to Exploit Index Redundancy in VQ Speech Compression,” Proceedings of the 1998 IEEE International Conference on Acoustics, Speech and Signal Processing, May 1998.
Sung Joo Kim, Seung Jong Park, and Yung Hwan Oh, “Complexity Reduction Methods for Vector Sum Excited Linear Prediction Coding,”XP002126377 Abstract, 1994.
W. Bastiaan Kleijn and Peter Kroon, “The RCELP Speech-Coding Algorithm,” vol. 5, No. 5, Sep.-Oct. 1994, pp. 39/573 -47/581.
C. Laflamme, J-P. Adoul, H.Y. Su, and S. Morissette, “On Reducing Computational Complexity of Codebook Search in CELP Coder Through the Use of Algebraic Codes,” 1990, pp. 177-180.
Chih-Chung Kuo, Fu-Rong Jean, and Hsiao-Chuan Wang, “Speech Classification Embedded in Adaptive Codebook Search for Low Bit-Rate CELP Coding,” IEEE Transactions on Speech and Audio Processing, vol. 3, No. 1, Jan. 1995, pp. 1-5.
Erdal Paksoy, Alan McCree, and Vish Viswanathan, “A Variable-Rate Multimodal Speech Coder with Gain-Matched Analysis-By-Synthesis, ” 1997, pp. 751-754.
Gerhard Schroeder, “International Telecommunication Union Telecommunications Standardization Sector,” Jun. 1995, pp. i-iv, 1-42.
“Digital Cellular Telecommunications System; Comfort Noise Aspects for Enhanced Full Rate (EFR) Speech Traffic Channels (GSM 06.62),” May 1996, pp. 1-16.
W. B. Kleijn and K.K. Paliwal (Editors), Speech Coding and Synthesis, Elsevier Science B.V.; Kroon and W.B. Kleijn (Authors), Chapter 3: “Linear-Prediction Based on Analysis-by-Synthesis Coding”, 1995, pp. 81-113.
W. B. Kleijn and K.K. Paliwal (Editors), Speech Coding

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