Abstract: An echo canceller provided with an expected echo generating unit which generates an expected echo of a transmission signal and a frequency offset correction unit which detects a phase error between an echo in a received signal and an expected echo from the expected echo generating unit and corrects the frequency offset for the expected echo by an offset frequency estimated based on the phase error, wherein use is made of the expected echo after offset correction by the frequency offset correction unit so as to suppress the echo in the received signal, the frequency offset correction unit using the phase error obtained after normalizing the phase error by the magnitude of the echo in the received signal. By this, it is possible to perform frequency offset correction adaptive to the far-end echo characteristics.

Abstract: A speech coding apparatus includes multipliers and prediction filters which successively process a plurality of signal vectors obtained from an index 2.sup.M and dimension N code book to obtain a reproduced speech signal. Error detectors are provided which find the error between the input speech signal and reproduced speech signal. Evaluators are also provided which calculate the optimum signal vectors giving the smallest errors. The multipliers are connected to a reduced code book, which is constituted of n number of code book blocks of index 2.sup.M/n and dimension N/n (where n is an integer of two or more). There are n number of multipliers, n number of prediction filters, n number of error detectors, and n number of evaluators corresponding to the code book blocks.

Abstract: A gain-shape vector quantization apparatus is provided for encoding and decoding, to transmit and receive compressed speech signals. A selected plurality of vectors are read from a code book based upon an index signal. The vectors are added in an adder and synthesis filtered by a synthesis filter, in either order, to produce an output. This output is subtracted from an input speech signal to produce an error signal. An evaluation unit produces an index to select the plurality of vectors read from the code book memory based on the error signal in order to minimize this error signal. The evaluation unit produces gain adjusting signals which can be used to adjust gains of the vectors read from the code book. In an encoder, signals indicative of the gain adjusting signal and the index signal are transmitted by a transmitter of the encoder to send a quantized speech signal to a receiver of a decoder.

Abstract: A code excited linear prediction (CELP) type speech signal coding system is provided, a code vector obtained by applying linear prediction to a vector of a residual speech signal of white noise is stored in a code book. A pitch prediction vector obtained by applying linear prediction to a residual signal of a preceding frame is given a delay corresponding to a pitch frequency and added to the code vector. Use is made of an impulse vector obtained by applying linear prediction to a residual signal vector of impulses having a predetermined relationship with the vectors of the white noise code book. Variable gains are given to at least the above code vector and impulse vector, a reproduced signal is produced, and this reproduced signal is used for identification of the input speech signal. Thus, a pulse series corresponding to the sound source of voiced speech sounds is created.

Abstract: Pitch periods for a long term predictor included in a speech codec are searched in two searching stages. In the first searching stage, probable pitch periods are searched skipping a constant number of pitch periods, and in the second searching stage, pitch periods including the pitch period determined in the first searching stage and pitch periods neighboring the pitch period on both sides are searched.

Abstract: A speech coding apparatus coupled to a transmission channel includes m (m is an integer greater than 1) coders, m decoders and m or (m-1) error correcting coders. The apparatus also includes an evaluation unit which evaluates a quality of each of reproduced speech signals from the input speech signal and the reproduced speech signals and which outputs an evaluated quality of each of the reproduced speech signals. The quality of each of the reproduced speech signals is evaluated in a state having no transmission error. A decision unit identifies one of the m coders which provides the reproduced speech signal having a smallest distortion on the basis of the evaluated quality of each of the reproduced speech signals, a current error rate of the transmission channel and error correcting abilities of the error correcting coders, and generates a coder identification number representative of a selected one of the m coders.

Abstract: A variable rate coder for transforming an input signal into a low bit rate digital signal, comprising: a plurality of coding units each comprising an output quantizer having a different number of output bits, and each independently transforming the input signal into a respective compressed digital signal. A plurality of bit rate determined circuits are provided to determine an output of one of the plurality of coding units as a candidate output of the coder by a respective individual determining method, in which method, the quality of the signal which is regenerated from the output of each coding unit is compared with an individual predetermined quality evaluating standard, and then the candidate output of the coding unit having the smallest number of output bits, among the candidate outputs of coding units satisfying each standard, is determined as the actual output of the coder.

Abstract: A speech coding apparatus which selects an optimum code from a code book, the optimum code giving the minimum magnitude of error signal between the input signal and the reproduced signal obtained by a filter calculation using a linear prediction parameter from a linear predictive analysis unit with respect to the codes of the code book, wherein the code book is formed by thinning to 1/M (M being an integer of two or more) the plurality of sampling values constituting the codes. To compensate for the deterioration of the quality of the reproduced signal caused by thinning the sampling values in this way, an additional linear predictive analysis unit is further introduced and use made of an amended linear prediction parameter instead of the linear prediction parameter from the originally provided linear predictive analysis unit.

Abstract: Several encoders perform a local decoding of a speech signal and extract excitation information and vocal tract information from a speech signal for an encoding operation. The transmission rate ratio between the excitation information and the vocal tract information are different for each encoder. An evaluation/selection unit evaluates the quality of decoded signals subjected to a local decoding in each of the encoders, determines the most suitable encoders from among the several encoders based on the result of the evaluation, and selects the most suitable encoder, thereby outputting the selection result as selection information. The decoder decodes a speech signal based on selection information, vocal tract information and excitation information. The evaluation/selection unit selects the output from the encoder in which the quality of a locally decoded signal is the most preferable.

Abstract: Speech presence versus silence is decided by a discriminator which can use a certain combination of parameter values: signal power, prediction error power, prediction error power deviation, and zero crossings.

Abstract: Disclosed is a voice coding/decoding system having a transmitting part for transmitting a coded signal of an input voice signal at a bit rate lower than a predetermined transmission bit rate and a receiving part for receiving and decoding the coded signal transmitted from the transmission part. To enable the coding and transmitting of an input voice signal in an optimum state without passing through a buffer memory and without having a negative influence on the coder, the transmitting part provides coders for coding the input voice signal and groups of entropy coders. The inputs of the entropy coders in each group are connected to the output of one of the plurality of coders. The transmitting part further provides an evaluation part for evaluating the characteristics of the outputs of the coders and the entropy coders.

Abstract: A gain-shape vector quantization apparatus for compressing the data of voice signal. A code book portion is constituted by a plurality of shape vectors which produce a plurality of selected shape vectors. A plurality of variable gain circuits impart gains to each shape vector produced from the code book portion. A plurality of synthesis filters regenerate signals from the outputs of the variable gain circuits. An adder adds the signals regenerated by the synthesis filters. An evaluation unit produces an index to select a plurality of shape vectors in the code book portion in order to minimize an error between the output of the adder and an input speech signal and further produces gain adjusting signal for the variable gain circuits.

Abstract: An echo canceller in a system, having a long impulse response such as in an acoustic system, which employs a fast Fourier transform to transform input data represented in the time domain into signals represented in the frequency domain to reduce calculations. To solve the problem of a long delay, the impulse response length is divided into a plurality of blocks. Each block then has a decreased number of samples within each block. Thus, a fast Fourier transform and finite impulse response type digital filtering are effected, so that the processing delay is decreased while the amount of calculations is kept small.

Abstract: Speech is data-compressed and coded for transmission using TDHC (Time Domain Harmonic Compression) for the voiced signal, and decimated sampling for the unvoiced signal. Features include voiced/unvoiced detection, pitch period detection, border detection, coding by ADPCM, and optimum quantization coding. Reception involves decoding and data-reconstruction.

Abstract: Coding transmission equipment wherein an plurality of adaptive coding units having different processing characteristics are provided, and the adaptive coding unit having the optimum processing characteristic for a current input signal is selected from among the plurality of adaptive coding units, and an output signal from the optimum adaptive coding unit and the unit number thereof are transmitted to the receiver side. At the receiver side, the processing characteristic of a decoder is changed to become the optimum processing characteristic indicated by the received unit number, whereby a high quality recovered signal is realized.

Abstract: An automatic equalization device used in a data communication system includes a unit for extracting a single pulse from a training signal sent from a transmitter and initializing a tap coefficient using the extracted single pulse. The automatic equalization device also includes a first equalization circuit and a second equalization circuit. In the first equalization circuit, an auto-correlation series of the signal corresponding to the single pulse is calculated to provide a symmetric single pulse. In the second equalization circuit, an inverse matrix is calculated from the symmetric single pulse using the auto-correlational series of single pulses. By these calculations, the speed of the initial setting of the tap coefficient is increased.

Abstract: A digital automatic gain control circuit provided with a first AGC loop and a second AGC loop; activated alternately. The first AGC loop is activated when an input signal is initially supplied. The loop produces a certain digital value calculated as an inverse number of the digital level of the input through predetermined digital arithmetic operations utilizing an approximation polynominal. The calculated digital value is then preset in certain portions of the second AGC loop. The second AGC loop starts operating by using said present value, then uses the digital level of the input itself, instead of the preset value, and produces an automatic gain controlled digital output. Utilizing said preset value as an initial input to the second AGC loop allows for rapid stabilization of the automatic gain controlled digital output.

Abstract: The present invention relates to a digital phase locked loop circuit, particularly to a circuit which realizes accurately digital phase locked loop pull-in operation at a high speed with a simplified circuit structure.In the present invention, in order to obtain a phase difference between a single frequency signal and the digital phase locked loop clock signal which is obtained by dividing a specified frequency signal with a dividing counter, the phase difference is obtained in accordance with the signs, absolute values and amplitude ratio of two adjacent sample values. The sample values of said single frequency signal are taken at two points based on said digital phase locked loop clock signal corresponding to a phase difference of .pi./2 radians of said single frequency signal. A fast pull-in of the digital phase locked loop is realized by setting a value corresponding to the obtained phase difference into a dividing counter.