Lucent Technologies Inc.

Alcatel Lucent

Craig R Watkins

Juin-Hwey Chen

Wei Wang

David Anthony Campana

Robert J McAulay

Joseph Gerard Aguilar

Robert W Zopf

Xiaoquin Sun

Robert B Dunn

Raymond Chen

A system and method for processing of audio and speech signals is disclosed, which provide compatibility over a range of communication devices operating at different sampling frequencies and/or bit rates. The analyzer of the system divides the input signal in different portions, at least one of which carries information sufficient to provide intelligible reconstruction of the input signal. The analyzer also encodes separate information about other portions of the signal in an embedded manner, so that a smooth transition can be achieved from low bit-rate to high bit-rate applications. Accordingly, communication devices operating at different sampling rates and/or bit-rates can extract corresponding information from the output bit stream of the analyzer. In the present invention embedded information generally relates to separate parameters of the input signal, or to additional resolution in the transmission of original signal parameters. Non-linear techniques for enhancing the overall performance of the system are also disclosed. Also disclosed is a novel method of improving the quantization of signal parameters. In a specific embodiment the input signal is processed in two or more modes dependent on the state of the signal in a frame. When the signal is determined to be in a transition state, the encoder provides phase information about N sinusoids, which the decoder end uses to improve the quality of the output signal at low bit rates.

Scalable and embedded codec for speech and audio signals

A speech coding system robust to frame erasure (or packet loss) is described. Illustrative embodiments are directed to a modified version of CCITT standard G.728. In the event of frame erasure, vectors of an excitation signal are synthesized based on previously stored excitation signal vectors generated during non-erased frames. Specifically, the decoder generates and stores samples of a first excitation signal in a memory, and then, in response to a signal indicating a frame erasure, the decoder synthesizes a second excitation signal based on the previously stored samples. In particular, the second excitation is synthesized by correlating a first subset of the stored samples with a second subset thereof, identifying a set of stored excitation signal samples based on the correlation, and synthesizing the second excitation signal based on the identified samples. Finally, the decoder then filters the second excitation signal to synthesize a signal reflecting human speech.

Linear prediction coefficient generation during frame erasure or packet

A speech coding system robust to frame erasure (or packet loss) is described. Illustrative embodiments are directed to a modified version of CCITT standard G.728. In the event of frame erasure, vectors of an excitation signal are synthesized based on previously stored excitation signal vectors generated during non-erased frames. This synthesis differs for voiced and non-voiced speech. During erased frames, linear prediction filter coefficients are synthesized as a weighted extrapolation of a set of linear prediction filter coefficients determined during non-erased frames. The weighting factor is a number less than 1. This weighting accomplishes a bandwidth-expansion of peaks in the frequency response of a linear predictive filter. Computational complexity during erased frames is reduced through the elimination of certain computations needed during non-erased frames only. This reduction in computational complexity offsets additional computation required for excitation signal synthesis and linear prediction filter coefficient generation during erased frames.

A method and apparatus for improving the performance of coding systems in the presence of frame erasures or lost packets. The encoded signal is modified after transmission but prior to decoding by a decoder preprocessor. The preprocessor recognizes that a given frame has been corrupted and modifies the encoded signal so that the decoding thereof will result in improved coding system performance. Specifically, based on the decoding process and on a predetermined target signal, the encoded signal is modified so that the decoding thereof will generate an approximation to the target signal. In a first illustrative embodiment, a CELP speech coder is used and the target signal is an excitation signal comprised of all-zero excitation vectors. In this case, the portion of the corrupted excitation signal indices which identify the corresponding gain factors are set to values which represent a low gain factor. In a second illustrative embodiment, a CELP speech coder is used and the target signal comprises an extrapolation of the excitation signal represented by the encoded signal for one or more previous frames. In this case, the preprocessor encodes the extrapolated excitation signal using the best codebook matches available. In either case, the effect of corrupted frames in the reconstructed speech signal is minimized.

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Hamilton, Australia

Craig R Watkins