Speech encoder using gain normalization that combines open...

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

Reexamination Certificate

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C704S229000

Reexamination Certificate

active

06260010

ABSTRACT:

MICROFICHE APPENDIX
A microfiche appendix is included in the application of 1 slide and 24 frames.
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. 09/156,814, entitled “Completed Fixed Codebook For Speech Encoder,” filed Sep. 18, 1998;
4) U.S. patent application Ser. No. 09/156,649, entitled “Comb Codebook Structure,” filed Sep. 18, 1998;
5) U.S. patent application Ser. No. 09/156,648, entitled “Low Complexity Random Codebook Structure,” 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 voice communication systems; and, more particularly, it relates to various techniques used with code-excited linear prediction coding to obtain high quality 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 LPC (linear predictive coding), 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.
Applying LPC techniques, a conventional source encoder operates on speech signals to extract modeling and parameter information for communication to a conventional source decoder via a communication channel. Once received, the decoder attempts to reconstruct a counterpart signal for playback that sounds to a human ear like the original speech.
A certain amount of communication channel bandwidth is required to communicate the modeling and parameter information to the decoder. In embodiments, for example where the channel bandwidth is shared and real-time reconstruction is necessary, a reduction in the required bandwidth proves beneficial. However, using conventional modeling techniques, the quality requirements in the reproduced speech limit the reduction of such bandwidth below certain levels.
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. Therein, the speech encoding system has an encoder processing circuit and a plurality of codebooks that generate excitation vectors. The encoder processing circuit calculates open loop gain and closed loop gain. The encoder processing circuit selectively applies the open and closed loop gains in gain normalization processing.
The selective application of the open loop gain and the closed loop gain by the encoder processing circuit may further involve the use of a weighting factor of linear predictive coding gain. The encoder processing circuit may use weighting factor to linearly combine the open and closed loop gains.
In certain embodiments, the selective application of the open and closed loop gains by the encoder processing circuit comprises applying the lesser of the open loop gain and the closed loop gain to the background noise. When using the linear predictive coding gain as a weighting factor, the encoder processing circuit may exclude such background noise from the application of the weighting factor.
Additionally, the encoder processing circuit may apply a maximum limit, a minimum limit, or both in gain normalization processing.
Further aspects of the present invention can also be found in a method used by a speech encoding system that applies an analysis by synthesis coding approach to a speech signal. The method involves the identification of open and closed loop gains for combination of contributions therefrom to generate a gain normalization factor.
In some embodiments, such method may further involve the use of linear predictive coding gain to identify appropriate contributions of the open and closed loop gains. One specific way to accomplish this involves the use of the linear predictive coding gain as a weighting factor.
When the speech signal comprises background noise, the encoder processing system may also select without combination either the open loop gain or the closed loop gain in generating the gain normalization factor. In some cases, this involves selection of the lesser of the open loop gain and the closed loop gain.
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.


REFERENCES:
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patent: 5751903 (1998-05-01), Swaminathan et al.
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patent: 5956683 (1999-09-01), Jacobs et al.
patent: 0501420A2 (1992-02-01), None
Jae H. Chung and Ronald W. Schafer, “Gain Normalization in a 4200 BPS Homomorphic Vocoder”,IEEE International Conference on Communications ICC '90 Including Supercomm Technical Sessions. Supercom ICC '90 Conference Record, vol. 3, 16-19, Apr. 1990, pp. 942-946.
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 Speechand 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 Synt

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