Fujitsu Corporation

Fumio Amano

Shigeyuki Unagami

Tomohiko Taniguchi

Yoshinori Tanaka

Yasuji Ohta

Kohei Iseda

Koji Okazaki

Akira Sasama

Yasuji Ota

Yoshihiro Sakai

Ryota Akiyama

Naoya Torii

Yoshihiro Tomita

Hideaki Kurihara

Mitsuhiro Azuma

Kazuo Murano

Shoji Tominaga

Shigeyuki Unnagami

Sigeyuki Unagami

Hector M Meana

Kazumi Satoh

Yoshikazu Itoh

Tatsuki Hayashi

Mohammad R Asharif

Mohammad Reza Asharif

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.

Speech coding apparatus for separately processing divided signal vectors

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. In the decoder, after the signals indicative of the gain adjusting signal and the index are received by the receiver of the decoder, an index and gain adjusting signal is derived for use to control reading of vectors from a code book and gains thereon to reproduce the speech signal.

Gain-shape vector quantization method and apparatus

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.

Code excited linear prediction speech coding system

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.

Pitch period searching method and circuit for speech codec

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. An output part outputs a multiplexed transmission signal including the coded speech signal generated by the one of the m coders identified by the decision unit and the error correcting code generated by a corresponding one of the m error correcting coders.

Speech coding apparatus using multimode coding

An apparatus for compressing and decompressing speech signals. The encoder or decoder comprises a plurality of codebooks each controlling a plurality of indexed code vectors for a different frequency band. Each of the codebooks is provided with a synthesis filter for reproducing a signal wave shape based on a code vector provided by the corresponding codebook. The encoder or decoder further comprises an adder for computing a sum of signal wave shapes reproduced by the synthesis filters. This arrangement can reduce the memory size of each codebook used for an encoding or a decoding process and an amount of operations of the encoding or decoding process.

Vector quantization encoder and vector quantization decoder

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.

Speech coding apparatus

A sub-band acoustic echo canceller includes a first division and decimation process part for dividing a reception signal from a line into first band signals of N channels and for decimating each of the first band signals to output decimated first band signals in a form of complex signals, where N is an integer greater than or equal to two, a second division and decimation process part for dividing a transmission signal into second band signals of N channels and for decimating each of the second band signals to output decimated second band signals in a form of complex signals, an echo canceller group made up of a group of echo cancellers for generating a pseudo echo in each band based on a corresponding one of the decimated first band signals received from the first division and decimation process part by referring to a corresponding one of the decimated second band signals received from the second division and decimation process part and for outputting a residual signal in each band by subtracting the pseudo echo of one band from the decimated second band signal of the same band, and an interpolation and synthesis process part for subjecting the residual signals received from the echo canceller group to interpolation and synthesis to output a synthesized residual signal which is transmitted to a line as the transmission signal, where the echo canceller group carries out an echo cancelling operation for each band in a complex signal region.

Sub-band acoustic echo canceller

Speech encoding/decoding apparatus having selected encoders

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. The evaluation part extracts those entropy coders having output bit rates lower than the transmission bit rate and extracts, from the coders connected to the extracted entropy coders, a coder having the best output characteristic. Then, the evaluation part outputs a selecting signal indicating the combination of the selected coder and an entropy coder from the extracted entropy coders. The transmitting part further provides a selecting part for selecting, in response to the selecting signal, the codeword passed through the combination of the coder and the entropy coder to be transmitted.

Voice coding/decoding system having selected coders and entropy coders

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.

Gain-shape vector quantization apparatus

A voice coding apparatus includes a pitch detecting circuit which detects a pitch period of a voice signal; a pitch waveform generating circuit which samples the voice signal for a plurality of pitches based on the pitch period detected by the pitch detecting circuit and which generates a waveform of one pitch from the waveform of the plurality of pitches; a band restriction circuit which restricts the frequency band of the one pitch waveform generated in the pitch waveform generating circuit; and a coding circuit for coding the voice waveform which is band restricted in the band restriction circuit. The sampling number of the waveform for a plurality of pitches and the restricted bandwidth can be changed in accordance with the amount of the pitch period extracted in the pitch detecting circuit. Further, the pitch detecting circuit is able to correctly detect the pitch period even when the pitch period is not a multiple of the sampling period.

Voice coding apparatus

Disclosed is a secret speech equipment for ensuring the secrecy of an analog voice signal, and used in voice communication systems such as an analog telephone and mobile radio. The equipment comprises a sub-band signal generating unit for treating input digital samples as frequency-multiplexed signals of voice spectra, to split the frequency-multiplexed signals into a plurality of frequency bands, and thus obtain sub-band signals of the frequency bands, a sub-band signal permutating unit for permuting the sequence of the sub-band signals, and a sub-band signal multiplexing unit for multiplexing the permutated sub-band signals.

Secret speech equipment

Echo canceller with short processing delay and decreased multiplication

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.

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Tokyo, Japan

Fumio Amano