Abstract: The present invention relates to the decoding/playback part of received sound data packets in systems for transmission of sound over packet switched networks. According to the invention, the lengths of received signal frames are manipulated by performing time expansion or time compression of one or more signal frames at time varying intervals and with time varying lengths of the expansion or the compression, said intervals and said lengths being determined so as to maintain a continuous flow of signal samples to be played back.

Abstract: The present invention relates to the decoding-/playback part of received sound data packets in systems for transmission of sound over packet switched networks. According to the invention, the lengths of received signal frames are manipulated by performing time expansion or time compression of one or more signal frames at time varying intervals and with time varying lengths of the expansion or the compression, said intervals and said lengths being determined so as to maintain a continuous flow of signal samples to be played back.

Abstract: The present invention relates to the decoding-/playback part of received sound data packets in systems for transmission of sound over packet switched networks. According to the invention, the lengths of received signal frames are manipulated by performing time expansion or time compression of one or more signal frames at time varying intervals and with time varying lengths of the expansion or the compression, said intervals and said lengths being determined so as to maintain a continuous flow of signal samples to be played back.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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<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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.