Lucent Technologies Inc.

At&t Bell Laboratories

At+t Corp.

American Telephone & Telegraph Co., At&t Bell Laboratories

American Telephone and Telegraph, Co.

Willem Bastiaan Kleijn

Richard H Ketchum

Walter T Hartwell

Daniel J Krasinski

Edward C Bronson

Peter Kroon

Dimitrios P Prezas

Dror Nahumi

Jesper Haagen

Thomas E Jacobs

Hans Petter Knagenhjelm

A generalized analysis-by-synthesis method and apparatus are disclosed. A plurality of trial original signals are generated based on an original signal for coding. The trial original signals are constrained to be perceptually similar to the original signal. Trial original signals are coded to produce one or more parameters representative thereof. Estimates of the trial original signals are synthesized from these parameters. Errors between the trial original signals and the synthesized estimates are determined. A coded representation of the original signal is determined which comprises parameters of the trial original signal having an associated error which satisfies an error evaluation process. Trial original signals may be generated by application of time-warps or time-shifts to the original signal. Coding of a trial original signal may be performed with conventional analysis-by-synthesis coding such as code-excited linear prediction coding (CELP). A minimum square error process may serve as the error criterion.

Generalized analysis-by-synthesis speech coding method and apparatus

A speech coding system providing reconstructed voiced speech with a smoothly evolving pitch-cycle waveform. A speech signal is represented by isolating and coding prototype waveforms. Each prototype waveform is an exemplary pitch-cycle of voiced speech. A coded prototype waveform is transmitted at regular intervals to a receiver which synthesizes (or reconstructs) an estimate of the original speech segment based on the prototypes. The estimate of the original speech signal is provided by a prototype interpolation process which provides a smooth time-evolution of pitch-cycle waveforms in the reconstructed speech. Illustratively, a frame of original speech is coded by first filtering the frame with a linear predictive filter. Next a pitch-cycle of the filtered original is identified and extracted as a prototype waveform. The prototype waveform is then represented as a set of Fourier series (frequency domain) coefficients. The pitch-period and Fourier coefficients of the prototype, as well as the parameters of the linear predictive filter, are used to represent a frame of original speech. These parameters are coded by vector and scalar quantization and communicated over a channel to a receiver which uses information representing two consecutive frames to reconstruct the earlier of the two frames based on a continuous prototype waveform interpolation process. Waveform interpolation may be combined with conventional CELP techniques for coding unvoiced portions of the original speech signal.

Prototype waveform speech coding with interpolation of pitch,

A method and apparatus which allows the transmission of the perceptually important features of a speech-coding parameter at a low bit rate. The speech coding parameter may, for example, comprise the signal power of the speech. The parameter is processed on a block by block basis. The parameter value at the block boundaries is transmitted by conventional methods such as, for example, by means of differential quantization. The shape of the reconstructed parameter contour within block boundaries is based on a classification. The classification determines perceptually important features of the parameter contour within a block. The classification can be performed either at the transmitting end of the coder (using, for example, the original parameter contour with high time resolution and possibly other speech parameters as well) or at the receiving end of the coder (using, for example, the transmitted parameter values, and possibly other transmitted speech parameters as well). Based on the result of the classification as well as the parameter values at the block boundaries, a parameter contour (within the block) is selected from an inventory of possible parameter contours. The inventory may include a linear interpolation contour and a step function contour. The step function contour may be particularly useful when the features indicate the presence of a plosive. The inventory may adapt to the transmitted parameter values at the block boundaries.

Speech coding parameter sequence reconstruction by sequence

An improved long-term predictor (LTP) for use in analysis-by-synthesis coding systems, such as CELP is disclosed. The invention provides control of the periodicity of speech signals generated by the LTP. This control facilitates a reduction in perceptible noise/buzziness in reconstructed speech. An embodiment of the invention includes a conventional LTP in combination with a two-tap finite impulse response filter. The filter augments operation of the LTP by generating precursor signals of LTP output signals. These precursor signals are combined with the LTP output signals to form the output of the improved LTP.

Long term predictor

Synergy between operations performed by a speech-rate modification system and those operations performed in a speech coding system is exploited to provide a speech-rate modification system with reduced hardware requirements. The speech rate of an input signal is modified based on a signal representing a predetermined change in speech rate. The modified speech-rate signal is then filtered to generate a speech signal having increased short-term correlation. Modification of the input speech signal may be performed by inserting in the input speech signal a previous sequence of samples corresponding substantially to a pitch cycle. Alternatively, the input speech signal may be modified by removing from the input speech signal a sequence of samples corresponding substantially to a pitch cycle.

Speech-rate modification for linear-prediction based

An improved method of speech coding for use in conjunction with speech coding methods wherein speech is digitized into a plurality of temporally defined frames, each frame including a plurality of sub-frames, and the digitized speech is partitioned into periodic components and a residual signal. For each of a plurality of sub-frames of the residual signal, the improved method of speech coding selects and applies a time shift T to the sub-frame by applying a matching criterion to (a) the current sub-frame of the residual signal, and (b) a sample-to-sample (subframe-to-subframe) pitch delay determined by applying linear interpolation to known pitch delays occurring at or near frame-to-frame boundaries of previous frames. The matching criterion is applied by minimizing .epsilon., where: ##EQU1## (r(n-T)) is the residual signal of the current frame shifted by time T, r(n-D(n)) is the delayed residual signal from a previously-occurring frame, n is a positive integer, r is the instantaneous amplitude of the residual signal, and D(n) is the sample-to-sample pitch delay determined by applying linear interpolation to known pitch delay values occurring at or near frame-to-frame boundaries.

RCELP coder

A decoding method and apparatus for speech coding systems which takes into account the fact that the human auditory system is sensitive to changes in signal characteristics. For example, a sustained distortion of the spectral characteristic of reconstructed speech is usually less perceptible than an objectively smaller distortion which changes as a function of time. This property of the auditory system is advantageously exploited in the design of a speech coding system receiver in accordance with the present invention, by selecting the sequence of decoded speech parameter values on a perceptual basis. Illustratively, the sequence of decoded speech parameter values is selected so as to describe a smooth path through the sequence of Voronoi regions. The distance between successive parameter values is advantageously minimized, under the constraint that the resultant parameter values fall within, or nearly within, the appropriate Voronoi regions. In this manner, a smoother trajectory will result, thereby enabling the receiver to produce a perceptually superior reconstructed speech signal.

Speech coding parameter smoothing method

A method of coding a speech signal is described. In accordance with the method, a plurality of sets of indexed parameters are generated based on samples of the speech signal. Each set of indexed parameters corresponds to a waveform characterizing the speech signal at a discrete point in time. Parameters of the plurality of sets are grouped based on index value to form a first set of signals which represents the evolution of characterizing waveform shape; the signals of the first set are filtered to remove low frequency components and thereby produce a second set of signals which represents relatively high rates of evolution of characterizing waveform shape. The speech signal is then coded based on the second set of signals representing high rates of characterizing waveform shape evolution. Coding of the speech signal may further be based on a set of smoothed first signals.

Decomposition in noise and periodic signal waveforms in waveform

A speech coding method and apparatus which selectively applies speech coding techniques to time segments of speech information signals, such as, e.g., pitch cycle waveforms is disclosed. A speech information signal comprising N signal segments is coded with a first speech coder to provide a first coded representation for each of the N signal segments. A second speech information signal reflecting speech information not coded by the first coder is determined for each of one or more of the N signal segments. In addition to coding the N first speech information signal segments with the first speech coder, M of the second speech information signals are coded with a second speech coder, where 1.ltoreq.M.ltoreq.N-1. The selective coding of M of the second speech information signals is done responsive a coding criterion. By selective use of the second speech coder, the number of bits needed to represent speech information may be reduced, or alternatively, better performance may be obtained without an increase in bit rate. The first and second speech coders may be any of those known in the art.

Selective application of speech coding techniques to input signal

A method and apparatus for processing a reconstructed speech signal from an analysis-by-synthesis decoder are provided to improve the quality of reconstructed speech. By operation of the invention, one or more traces in a reconstructed speech signal are identified. Traces are sequences of like-features in the reconstructed speech signal. The like-features are identified by time-distance data received from the long term predictor of the decoder. The identified traces are smoothed by one of the known smoothing techniques. A smoothed version of the reconstructed speech signal is formed by combining one or more of the smoothed traces. The original reconstructed speech signal may be that provided by a long term predictor of the decoder. Values of the reconstructed speech signal and smoothed speech signal may be combined based on a measure of periodicity in speech.

Method and apparatus for smoothing pitch-cycle waveforms

A parameter communication arrangement where a parameter that is transmitted over a channel using m-bit codewords or labels is quantized before transmission as one of only p levels, where, significantly, p&lt;k=2.sup.m. Since only p labels are needed to transmit the p levels, the unused k-p labels are advantageously available to provide redundancy. The receiver decodes the redundant labels in accordance with an error routine. An encoding table mapping from the p levels to p labels and a decoding table inverse mapping from the p labels to p levels are obtained using an optimization procedure to minimize the effect of channels errors. The optimization is based on the probability distribution for the p levels such that a relatively high proportion of the error protection made available by having redundant labels inures to the benefit of parameter levels which are more likely to be transmitted. The optimization procedure is a well known technique referred to as simulated annealing which is for the first time applied to source dependent channel coding.

Source dependent channel coding with error protection

Apparatus for encoding speech using a code excited linear predictive (CELP) encoder using a virtual searching technique during speech transitions such as from unvoiced to voiced regions of speech. The encoder compares candidate excitation vectors stored in a codebook with a target excitation vector representing a frame of speech to determine the candidate vector that best matches the target vector by repeating a first portion of each candidate vector into a second portion of each candidate vector. For increased performance, a stochastically excited linear predictive (SELP) encoder is used in series with the adaptive CELP encoder. The SELP encoder is responsive to the difference between the target vector and the best matched candidate vector to search its own overlapping codebook in a recursive manner to determine a candidate vector that provides the best match. Both of the best matched candidate vectors are used in speech synthesis.

Code excited linear predictive vocoder using virtual searching

Apparatus for encoding speech using a code excited linear predictive (CELP) encoder using a recursive computational unit. In response to a target excitation vector that models a present frame of speech, the computational unit utilizes a finite impulse response linear predictive coding (LPC) filter and an overlapping codebook to determine a candidate excitation vector from the codebook that matches the target excitation vector after searching the entire codebook for the best match. For each candidate excitation vector accessed from the overlapping codebook, only one sample of the accessed vector and one sample of the previously accessed vector must have arithmetic operations performed on them to evaluate the new vector rather than all of the samples as is normal for CELP methods. For increased performance, a stochastically excited linear predictive (SELP) encoder is used in series with the adaptive CELP encoder. The SELP encoder is responsive to the difference between the target excitation vector and the best matched candidate excitation vector to search its own overlapping codebook in a recursive manner to determine a candidate excitation vector that provides the best match. Both of the best matched candidate vectors are used in speech synthesis.

Code excited linear predictive vocoder

A speech analyzer and synthesizer system using sinusoidal encoding and decoding techniques for voiced frames and noise excitation or multiple pulse excitation for unvoiced frames. For voiced frames, the analyzer transmits the pitch, values for each harmonic frequency by defining the offset from integer multiples of the fundamental frequency, total frame energy, and linear predictive coding, LPC, coefficients. The synthesizer is responsive to that information to determine the phase of the fundamental frequency and each harmonic based on the transmitted pitch and harmonic offset information and to determine the amplitudes of the harmonics utilizing the total frame energy and LPC coefficients. Once the phase and amplitudes have been determined for the fundamental and harmonic frequencies, the sinusoidal analysis is performed for voiced frames. For each frame, the determined frequencies and amplitudes are defined at the center of the frame, and a linear interpolation is used both to determine continuous frequency and amplitude signals of the fundamental and the harmonics throughout the entire frame by the synthesizer. In addition, the analyzer initially adjusts the pitch so that the harmonics are evenly distributed around integer multiples of this pitch.

Digital speech vocoder

A speech analyzer and synthesizer system using a sinusoidal encoding and decoding technique for voiced frames and noise excitation or multipulse excitation for unvoiced frames. For voiced frames, the analyzer transmits the pitch, values for a subset of offsets defining differences between harmonic frequencies and a fundamental frequency, total frame energy, and linear predictive coding, LPC, coefficients. The synthesizer is responsive to that information to determine the harmonic frequencies from the offset information for a subset of the harmonics and to determine the remaining harmonics from the fundamental frequency. The synthesizer then determines the phase for the fundamental frequency and harmonic frequencies and determines the amplitudes of the fundamental and harmonics using the total frame energy and the LPC coefficients. Once the phase and amplitudes have been determined for the fundamental and harmonic frequencies, the synthesizer performs a sinusoidal analysis. In another embodiment, the remaining harmonic frequencies are determined by calculating the theoretical harmonic frequencies for the remaining harmonic frequencies and grouping these theoretical frequencies into groups having the same number as the number of offsets transmitted. The offsets are then added to the corresponding theoretical harmonics of each of the groups of the remaining harmonic frequencies to generate the remaining harmonic frequencies. In a third embodiment, the offset signals are randomly permuted before being added to the groups of theoretical frequencies to generate the remaining harmonic frequencies.

Growing community of inventors

Basking Ridge, NJ, United States of America

Willem Bastiaan Kleijn