Abstract: A method and apparatus for performing entropy coding and decoding of a sequence of coded symbols representative of, for example, a speech, audio or video signal, in which variable-size vectors are coded and decoded based on radix arithmetic. The encoding technique uses a first radix and the numerical values of individual symbols to be coded, in order to determine the length of a first subsequence of symbols, which is then coded with use of a single (a first) combined symbol, and uses a second radix and the numerical values of other individual symbols to be coded, in order to determine the length of a second subsequence of symbols, which is then also coded with a single (a second) combined symbol, wherein the length of the first and second subsequences of symbols are also based on the size of the set from which the combined symbols are selected. The number of symbols in the first subsequence and the second subsequence are unequal—that is, the vectors (i.e.

Abstract: A low-complexity method and apparatus for performing signal decomposition in a low bit-rate WI speech encoder. A time-ordered sequence of sets of time-domain parameters is generated based on samples of a speech signal to be coded, each set of time-domain parameters corresponding to a waveform characterizing the speech signal. A cross correlation is then performed between two or more of said sets of time-domain parameters to produce a set of signals which represents relatively high rates of evolution of characterizing waveform shape across the time-ordered sequence of sets. Finally, the speech signal is coded based on the produced set of signals. A set of signals which represents relatively low rates of evolution of characterizing waveform shape across the time-ordered sequence of sets may also be produced.

Abstract: A low-complexity method and apparatus for performing waveform interpolation in a low bit-rate WI speech decoder, wherein interpolation between received waveforms is performed with use of spline coefficients generated based thereupon. Specifically, two signals are received from a WI encoder, each comprising a set of frequency domain parameters representing a speech signal segment of a corresponding pitch period. Then, spline coefficients are generated from each of the received signals, wherein each set of spline coefficients comprises a spline representation of a time domain transformation of the corresponding set of frequency domain parameters. Finally, the decoder interpolates between the spline representations to generate interpolated time domain data which is used to synthesize a reconstructed speech signal. In certain embodiments of the present invention, the time scale of at least one of the spline representations is modified to enable the interpolation therebetween.

Abstract: A CELP speech decoder includes a first portion comprising an adaptive codebook and a second portion comprising a fixed codebook. The CS-ACELP decoder generates a speech excitation signal selectively based on output signals from said first and second portions when said decoder fails to receive reliably at least a portion of a current frame of compressed speech information. The decoder does this by classifying the speech signal to be generated as periodic (voiced) or non-periodic (unvoiced) and then generating an excitation signal based on this classification. If the speech signal is classified as periodic, the excitation signal is generated based on the output signal from the first portion and not on the output signal from the second portion. If the speech signal is classified as non-periodic, the excitation signal is generated based on the output signal from said second portion and not on the output signal from said first portion.

Abstract: In CELP coding, stochastic (noise-like) excitation is used in exciting a cascade of long-term and short-term all-pole linear synthesis filters. This approach is based on the observation that the ideal excitation, obtained by inverse-filtering the speech signal, can be modeled for simplicity as Gaussian white noise. Although such stochastic excitation resembles the ideal excitation in its global statistical properties, it contains a noisy component that is irrelevant to the synthesis process. This component introduces some roughness and noisiness in the synthesized speech. The present invention reduces this effect by adaptively controlling the level of the stochastic excitation. The proposed control mechanism links the stochastic excitation to the long-term predictor in such a way that the excitation level is inversely related to the efficiency of the predictor.

Abstract: In a speech decoder which experiences frame erasure, the pitch delay associated with the first of consecutive erased frames is incremented. The incremented value is used as the pitch delay for the second of consecutive erased frames. Pitch delay associated with the first of consecutive erased frames may correspond to the last correctly received pitch delay information from a speech encoder (associated with a non-erased frame), or it may itself be the result of an increment added to a still previous value of pitch delay (associated with a still previous erased frame).

Abstract: The invention is used to shape noise in time domain and frequency domain coding schemes. The method advantageously uses a noise weighting filter based on a filterbank with variable gains. A method is presented for decoding an encoded signal based on received side information and on a masking matrix derived from the masking properties of speech. In particular, the encoded signal is separated into subband signal components, each of which is multiplied by a corresponding gain value based on the masking matrix. These multiplied subband signal components are then combined to produce a decoded signal.

Abstract: The invention is used to shape noise in time domain and frequency domain coding schemes. The method advantageously uses a noise weighting filter based on a filterbank with variable gains. A method is presented for computing the gains in the noise weighting filterbank with filter parameters derived from the masking properties of speech. Illustrative embodiments of the method in various coding schemes are illustrated.

Abstract: A new method for high quality speech coding, Timing-Frequency Interpolation (TFI) which offers advantages over conventional CELP (code-excited linear predictive) algorithms for low rate coding. The method, provides a perceptually advantageous framework for voiced speech processing. The general formulation of the TFI technique is described.

Abstract: In Code Excited Linear Predictive (CELP) coding, stochastic (noise-like) excitation is used in exciting a cascade of long-term and short-term all-pole linear synthesis filters. This approach is based on the observation that the ideal excitation, obtained by inverse-filtering the speech signal, can be modeled for simplicity as Gaussian white noise. Although such stochastic excitation resembles the ideal excitation in its global statistical properties, it contains a noisy component that is irrelevant to the synthesis process. This component introduces some roughness and noisiness in the synthesized speech. The present invention reduces this effect by adaptively controlling the level of the stochastic excitation. The proposed control mechanism links the stochastic excitation to the long-term predictor in such a way that the excitation level is inversely related to the efficiency of the predictor.

Abstract: An improved digital communication system, e.g., a CELP code/decoder based system, is improved for use with a wide-band signal such as a high-quality speech signal by modifying the noise weighting filter used in such systems to include a filter section which affects primarily the spectral tilt of the weighting filter in addition to a filter component reflecting formant frequency information in the input signal. Alternatively, the weighting is modified to reflect perceptual transform techniques.