Synchronized encoder-decoder frame concealment using speech...

Details Synchronized encoder-decoder frame concealment using speech... Synchronized encoder-decoder frame concealment using speech...

1998-09-18

2001-02-13

Zele, Krista (Department: 2748)

Data processing: speech signal processing, linguistics, language

Speech signal processing

For storage or transmission

Reexamination Certificate

active

06188980

ABSTRACT:

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, 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. 09/156,650, 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, entitled “Speech Encoder Using Voice Activity Detection In Coding Noise,” filed Sep. 18, 1998;
8) U.S. patent application Ser. No. 09/154,654, entitled “Pitch Determination Using Speech Classification And Prior Pitch Estimation,” filed Sep. 18, 1998;
9) U.S. patent application Ser. No. 09/154,657, entitled “Speech Encoder Using A Classifier For Smoothing Noise Coding,” filed Sep. 18, 1998;
10) U.S. patent application Ser. No. 09/156,826, entitled “Adaptive Tilt Compensation For Synthesized Speech Residual,” filed Sep. 18, 1998;
11) U.S. patent application Ser. No. 09/154,662, entitled “Speech Classification And Parameter Weighting Used In Codebook Search,” 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.
In conventional speech codecs, linear predictive coding (LPC) produces LSF (Line Spectral Frequencies) vectors that occasionally have LSF entries that are out of ascending order. LSF vectors may also become out of ascending order when errors are introduced, for example, through a communication channel. Flipped LSFs at a decoder produce unwanted clicking or popping sounds due to filter instability.
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 approach on a speech signal. The speech encoding system comprises an encoder that generates a series of line spectral frequencies that are occasionally produced out of order. If so, the encoder selects from a plurality of correction techniques to accommodate the out of order series. One of the correction techniques involves reordering the series of line spectral frequencies that are produced out of order.
Depending on the embodiment, there may be several types of other correction techniques employed. For example, line spectral frequency concealment, application of at least a portion of one previous series of line spectral frequencies, or frame erasure might be applied. Similarly, a further technique might involve disregarding at least in part the series of line spectral frequencies that are produced out of order.
The encoder may select one of the plurality of correction techniques by examining the series of line spectral frequencies that are produced out of order, for example, by considering a number of line spectral frequencies out of order.
Other aspects of the present invention may be found in a method used by a speech encoder that operates on a speech signal. The method involves producing from the speech signal a series of line spectral frequencies that are occasionally produced out of order. When out of order, the method also involves deciding whether to reorder the series of line spectral frequencies that are produced out of order, or to replace at least a portion of the series of line spectral frequencies that are produced out of order using at least a portion of one previous series of line spectral frequencies.
The method may also involve determining a number of line spectral frequencies out of order in the series of line spectral frequencies that are produced out of order. Therein, the number of line spectral frequencies out of order can be used in the decision to either reorder or replace.
A speech decoder may also be included which reorders a received series of line spectral frequencies that are out of order. The speech decoder may also or alternatively replace at least a portion of a received series of line spectral frequencies that are out of order using at least a portion of one previously received series of filter coefficients.
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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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-Ma

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