Fujitsu Corporation

Asahi Glass Company, Limited

Nakashima Manufacturing Co., Ltd.

Hideaki Kurihara

Yoshinori Tanaka

Tomohiko Taniguchi

Yasuji Ohta

Yoshihiro Sakai

Teruyuki Sato

Mikio Kozuka

Mark A Johnson

Ryuichiro Takeda

Yasuko Shirai

Toshihiro Ootani

Mitsuru Takahashi

Yoshinori Soejima

Hiroshi Katayama

Kohei Iseda

Shigeyuki Unagami

Fumio Amano

Noboru Kobayashi

Naoji Fujino

Takehiro Nakai

Koji Okazaki

Toshiaki Nobumoto

Chiharu Kawai

Fumiaki Nishida

Kazuhiro Nomoto

Masato Ito

Kiyotaka Mishima

Hiroshi Ichimura

Tadashi Fukuwatari

Kazumi Satoh

Mark Johnson

The present invention provides an acoustic echo canceler for use in a telephone set having a handsfree conversation function. This acoustic echo canceler simultaneously achieves both the stabilization and speedup of an estimation operation and also attains the more improvement of speech quality. Moreover, the present invention provides a handsfree telephone set employing this acoustic echo canceler. Furthermore, this acoustic echo canceler is operative to cancel an echo generated by inputting an output of an acoustic device to itself substantially simultaneously with the outputting the output thereof. Further, this acoustic echo canceler comprises an adaptive filter for performing an operation of estimating the frequency characteristic and the phase characteristic of a part, whose frequency characteristic and phase characteristic vary, on an acoustic path and a fixed coefficient filter for simulating a part, whose frequency characteristic and phase characteristic are fixed, on the acoustic path.

Acoustic echo canceler and handsfree telephone set

A multi-channel signal processing apparatus comprises a memory unit, a signal processing unit for single-channel implementing predetermined signal processing using a memory region of the memory unit according to a signal processing program for single-channel, an address generation unit generating a first memory address for single-channel processing, a pointer adding unit adding a pointer value indicating the head of an unused memory region of the memory unit to the first memory address generated by the address generation unit each time the signal processing in the signal processing unit is completed to generate a second memory address, and an address selecting unit selecting either the first memory address or the second memory address as a real address for the memory unit, thereby enabling multi-channel signal processing having a high processing efficiency without adding a large modification to the signal processing program for single-channel.

Memory management apparatus in a multi-channel signal processor

A CODEC has a DSP which can consecutively execute a plurality of algorithms without restriction of a memory capacity. The DSP performs an encoding/decoding operation on a digital signal. A program memory stores a program divided into a plurality of block programs, the program being stored on an individual block program basis. A data memory stores a set of data used for executing each block program stored in the program memory, the set of data being divided into a plurality of data blocks and stored on an individual data block basis. A program executing unit executes each block program stored in the program memory by using a corresponding data block stored in the data memory. A program changing unit obtains a new block program from an external device each time execution of one of the block programs by the program executing unit is completed so as to store the obtained new block program in the program memory.

CODEC for consecutively performing a plurality of algorithms

A larger number, L', of delta vectors .DELTA..sub.i (i=0, 1, 2, . . . , L'-1) than the required number L are each multiplied by a matrix of a linear predictive synthesis filter (3), their power (A.DELTA..sub.i).sup.T (A.DELTA..sub.i) is evaluated (42), and the delta vectors are reordered in decreasing order of power (43); then, L delta vectors are selected in decreasing order of power, the largest power first, to construct a tree-structure data code book (41), using which A-b-S vector quantization is performed (48). This provides increased freedom for the space formed by the delta vectors and improves quantization characteristics. Further, variable rate encoding is achieved by taking advantage of the structure of the tree-structure data code book.

Speech encoding method and apparatus using tree-structure delta code book

Speech coding uses a periodic excitation signal source, a non-periodic signal excitation source, a synthesizing unit, a filter, and an error evaluating unit. The synthesizing unit synthesizes a periodic excitation signal and a non-periodic excitation signal output from the above-mentioned sources to generate an excitation sound source signal. The filter regenerates an input speech signal from the excitation sound signal. The error evaluating unit controls the periodic excitation signal source and the non-periodic excitation signal source so that the above-mentioned units output periodic and non-periodic excitation signals which minimize a difference of the regenerated signal from an input speech signal. When the error is minimized, the above-mentioned sound source is supplied to a feedback excitation signal modifying unit. The feedback excitation modifying unit reduces a non-periodic component in the above-mentioned supplied signal so that the non-periodic component is reduced more when the relative amount of the periodic component is greater. The modified signal is fed back to the periodic excitation signal source unit to modify the content thereof.

Speech signal coding system wherein non-periodic component feedback to

An information signal detection device and an information signal detection method are disclosed with which it is possible to transmit information by means of a DTMF signal using a channel that is prone to distortion, such as a high-efficiency channel. The information signal detection device comprises a filter bank component and a logic component. The filter bank component receives via a channel an information signal in which are superposed sine waves of two different frequencies selected from a specific number of frequencies sent by the sender, and generates binary information patterns showing in binary information whether the signal components of the various frequencies of this information signal are present. The logic component recognizes the end of the signal region when the binary information pattern generated in the binary information pattern generation step is containing no binary information showing that a signal component is present continuously for a specific period of time. The logic component also integrates the binary information patterns which indicate that at least one type of frequency signal component is contained and which are output from the filter bank component until the end of the signal period is recognized. One the end of the signal period has been recognized, the logic component decides the contents of the DTMF signal sent from the sender based on these integration results.

Information signal detection device and information signal detection

A speech signal coding apparatus inputs a pitch period generated by coding a speech signal, and outputs information on a range for said pitch period, together with the pitch period. A speech signal decoding apparatus inputs the pitch period and the above information on the range, and determines whether or not the pitch period is within the range. When the pitch period is determined to be within the range, the speech signal decoding apparatus outputs the above pitch period. When the pitch period is determined not to be within the range, the speech signal decoding apparatus outputs as a pitch period a predetermined value within the range.

Speech signal coding and decoding system transmitting allowance range

Speech coding system having codebook storing differential vectors

A CELP type speech coding system is provided with an arithmetic processing unit which transforms a perceptual weighted input speech signal vector AX to a vector .sup.t AAX, a sparse adaptive codebook which stores a plurality of pitch prediction residual vectors P sparsed by a sparse unit, and a multiplying unit which multiplies the successively read out vectors P and the output .sup.t AAX from the arithmetic processing unit. In addition, the CELP type speech coding system includes a filter operation unit which performs a filter operation on the vectors P, and an evaluation unit which finds the optimum vector P based on the output from the filter operation unit, so as to enable reduction of the amount of arithmetic operations.

Speech coding and decoding system

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

Fiber reinforced cement is continuously prepared by forming a conveyor by connecting a plurality of suction boxes each having a perforated plate, contacting the perforated plates of the suction boxes with a filter cloth and moving the suction boxes with the filter cloth at a synchronous speed, feeding a cementitious slurry and a fiber on the filter cloth, removing excess water by suction from the cementitious slurry through a suction trough connected to a suction device, and releasing the suction on the suction boxes to separate the filter cloth from the perforated plates of the suction boxes.

Process for continuously preparing fiber reinforced cement

An apparatus for continuous liquid-solid separation has suction boxes which respectively have a suction hole on a rear surface and a perforated surface at a front surface and said suction boxes being connected in series to form a conveyor and said suction holes being connected to a suction trough so as to give the sucking condition of the suction boxes in the zone of the suction trough wherein a filter cloth is contacted with the perforated surfaces of the suction boxes and is held by side plates which can be closed and opened at both sides of the suction boxes and the filter cloth and the suction boxes are moved at the synchronized velocity under suction.

Apparatus for continuous liquid-solid separation

A process for preparing glass fiber-reinforced cement sheets by dewatering a slurry wherein at least one member selected from asbestos, cotton, flax, hemp, wool, or silk fibers is added to the aqueous cement-glass fiber slurry. The sheets so produced have increased bending strengths and improved peeling resistance.

Growing community of inventors

Kawasaki, Japan

Hideaki Kurihara