Speech encoder adaptively applying pitch preprocessing with...

Details Speech encoder adaptively applying pitch preprocessing with... Speech encoder adaptively applying pitch preprocessing with...

1998-09-18

2001-12-11

Tsang, Fan (Department: 2645)

Data processing: speech signal processing, linguistics, language

Speech signal processing

For storage or transmission

C704S219000, C704S229000, C704S230000

Reexamination Certificate

active

06330533

ABSTRACT:

MICROFICHE APPENDICES B AND C
A microfiche appendix containing Appendix B (pages 88-89) and Appendix C (pages 90-109) of the originally submitted U.S. Patent Application, prepared in accordance with the standards set forth in 37 C.F.R. § 1.96(c)(2) per the Examiner's request, consisting of one (1) slide and 24 frames, is hereby incorporated herein by reference in its entirety and made part of the present U.S. Patent Application for all purposes.
MICROFICHE APPENDIX
A microfiche appendix is included in the application.
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 Serial No. 60/097,569 (Attorney Docket No. 98RSS325), entitled “Adaptive Rate Speech Codec,” filed Aug. 24, 1998;
2) U.S. patent application Ser. No. 09/154,675 (Attorney Docket No. 97RSS383), entitled “Speech Encoder Using Continuous Warping In Long Term Preprocessing,” filed Sep. 18, 1998;
3) U.S. patent application Ser. No. 09/156,814 (Attorney Docket No. 98RSS365), entitled “Completed Fixed Codebook For Speech Encoder,” filed Sep. 18, 1998;
4) U.S. patent application Ser. No. 09/156,649 (Attorney Docket No. 95E020), entitled “Comb Codebook Structure,” filed Sep. 18, 1998;
5) U.S. patent application Ser. No. 09/156,648 (Attorney Docket No. 98RSS228), entitled “Low Complexity Random Codebook Structure,” filed Sep. 18, 1998;
6) U.S. patent application Ser. No. 09/156,650 (Attorney Docket No. 98RSS343), entitled “Speech Encoder Using Gain Normalization That Combines Open And Closed Loop Gains,” filed Sep. 18, 1998;
7) U.S. patent application Ser. No. 09/156,832 (Attorney Docket No. 97RSS039), entitled “Speech Encoder Using Voice Activity Detection In Coding Noise,” filed Sep. 18, 1998;
8) U.S. patent application Ser. No. 09/154,654 (Attorney Docket No. 98RSS344), entitled “Pitch Determination Using Speech Classification And Prior Pitch Estimation,” filed Sep. 18, 1998;
9) U.S. patent application Ser. No. 09/154,657 (Attorney Docket No. 98RSS328), entitled “Speech Encoder Using A Classifier For Smoothing Noise Coding,” filed Sep. 18, 1998;
10) U.S. patent application Ser. No. 09/156,826 (Attorney Docket No. 98RSS382), entitled “Adaptive Tilt Compensation For Synthesized Speech Residual,” filed Sep. 18, 1998;
11) U.S. patent application Ser. No. 09/154,662 (Attorney Docket No. 98RSS383), entitled “Speech Classification And Parameter Weighting Used In Codebook Search,” filed Sep. 18, 1998;
12) U.S. patent application Ser. No. 09/154,653 (Attorney Docket No. 98RSS406), entitled “Synchronized Encoder-Decoder Frame Concealment Using Speech Coding Parameters,” filed Sep. 18, 1998;
13) U.S. patent application Ser. No. 09/154,663 (Attorney Docket No. 98RSS345), entitled “Adaptive Gain Reduction To Produce Fixed Codebook Target Signal,” 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.
Speech encoding becomes increasingly more difficult as data transmission bit rates decrease. In the absence of embedded intelligence to select an optimal encoding mode or scheme, many speech encoders do not maximize their inherent computational capacity in response to varying operating conditions. Particularly within data transmission systems that operate at varying bit rates, the inability to adapt to a particular encoding scheme based upon the available transmission bit rate at a given time results in an inefficient use of the encoder's resources.
Additionally, the inability to determine the optimal encoding mode for a given speech signal at a given bit rate also contributes to inefficient resource allocation. For a given speech signal and available bit rate, the ability to adaptively select an optimal coding scheme at a given bit rate would provide more efficient use of an encoder processing circuit. Moreover, the inability to select the optimal encoding mode for a given signal after identifying the computational resources required by the various available encoding modes often results in over-dedicating computational resources of a speech encoding system.
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. An intelligent encoding process adaptively selects from among various available encoding schemes. The speech encoding system may then apply the selected encoding scheme to provide optimal computational resource allocation within an encoder processing circuit. The encoding schemes may include code excited linear prediction and pitch preprocessing coding. One of the encoding schemes may include pitch preprocessing that includes continuous warping of the speech signal itself or of various coding parameters of the speech signal.
In certain embodiments of the invention, the encoder processing circuit may perform code excited linear prediction coding if the available transmission bit rate is above a predetermined upper threshold. Conversely, if the available bit rate is below a predetermined lower threshold, pitch preprocessing coding may be performed. If the available bit rate lies between the predetermined upper and lower thresholds, an operational selection process may adaptively select the optimal encoding scheme from various coding schemes for efficient use of the encoder processing circuit's computational resources.
In other embodiments, the encoder processing circuit may perform long term prediction if the speech signal is substantially non-stationary speech. Conversely, if the speech signal is substantially stationary speech, pitch preprocessing coding may be performed.
For example, in this interim bit rate range, pitch preprocessing coding could be used if it were to require less of the encoder processing circuit's computational resources for a given speech signal. However, code excited linear prediction coding could be employed if it were to place less of a burden on the encoder processing circuit to process the given speech signal.
The present invention, by employing adaptive selection among various encoding schemes, can provid

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