Mitsubishi Denki Kabushiki Kaisha

Mitsubishi Electric Corporation

Bunkei Matsuoka

Masaru Kimura

Shinya Takahashi

Hirohisa Tasaki

Atsuyoshi Yano

Ikuo Kajiyama

Takashi Yamazaki

Satoru Furuta

Tomoharu Awano

Asako Omote

A sound signal control device in a device including an input transducer to detect sound in a space in a vehicle, a sensor to image an occupant in the space, and an output transducer to emit sound into the space includes: a controller to generate information indicating a position of a head of the occupant and information indicating a state of the occupant, based on an image imaged by the sensor; a processor to generate a cancellation signal for cancelling noise at the position based on a sound signal representing sound detected by the input transducer and the information, and generate a control signal for controlling sound at the position based on the cancellation signal; and a converter to cause the output transducer to emit a sound corresponding to the control signal. The processor adjusts a degree to which the noise is cancelled, based on the information indicating the state.

Sound signal control device and method, and recording medium

An audio signal processing deviceincludes a period detecting unitfor detecting the fundamental period of an input audio signal; a square wave generating unitfor generating, according to the fundamental period the period detecting unitdetects, a square wavewhose period is an integer multiple of the fundamental period; an amplitude correction coefficient generating unitfor calculating an amplitude correction coefficientapproximately equal and proportional to the intensity of the input audio signal; a first multiplierfor generating an amplitude-corrected square waveby multiplying the square waveby the amplitude correction coefficient; and an adderfor adding the amplitude-corrected square waveto the input audio signal

Audio signal processing device

An echo canceler includes a signal-to-echo ratio calculating unitand a residual echo suppressing unit. The signal-to-echo ratio calculating unitcomputes a signal-to-echo ratio SE(n) indicating a ratio of an echo component to a received signal x(n) from a first residual signal and a second residual signal. The first residual signal is obtained using a filter coefficient sequence of an update filter, which is obtained up to the previous operation. The second residual signal is obtained using an updated filter coefficient sequence that undergoes a coefficient update which is performed, using an arbitrary update step size &#x3bc;(n), on the filter coefficient sequence of the update filter, which is obtained up to the previous operation. The residual echo suppressing unitsuppresses the echo component contained in the microphone input signal in accordance with the signal-to-echo ratio the signal-to-echo ratio calculating unitcomputes.

Echo canceler and echo detector

A signal processing device is provided, including a prediction error calculating unit that calculates an error signal between a left signal l(n) and a prediction signal of the left signal l (n) predicted from a right signal r(n), a gain adjusting unit that makes a gain adjustment and outputs an error signal, a first adder that adds the left signal l(n) and the error signal and outputs, and a second adder that adds the right signal r(n) and the error signal in opposite phase and outputs.

Signal processing device for processing stereo signals

A signal processing sectionis set up which provides an anti-phase component signal S extracted by an anti-phase component extracting sectionwith a transfer characteristic (H+H)/(H&#x2212;H); an adderadds the phase-inverted signal of the anti-phase component signal S provided with the transfer characteristic by the signal processing sectionand an in-phase component signal M extracted by an in-phase component extracting section; and an adderadds the anti-phase component signal S provided with the transfer characteristic by the signal processing sectionand the in-phase component signal M extracted by the in-phase component extracting section

Audio device

To date, there has been no speaker-characteristic compensation method including a processing step of reducing crosstalk between speakers incorporated in a mobile terminal. Accordingly, it has been impossible to reduce crosstalk that, in a mobile terminal, occurs between speakers. A speaker-characteristic compensation method, for a mobile terminal device having at least two speakers in a case, according to the present invention includes steps of reducing mutual crosstalk between the speakers, thereby reducing crosstalk between the speakers.

Speaker-characteristic method and speaker reproduction system

A mobile phone terminal includes a transmitting side speech signal path switch and receiving side speech signal path switch, and a general purpose USB port. The signal path switches are interposed between a speech coder/decoder and an A/D converter and D/A converter to enable a terminal acoustic evaluation signal to be input and output through a path different from that in a normal operation mode of the mobile phone terminal. The general purpose USB port is used for connecting the signal path switches with an external terminal unit so that the control signals of the signal path switches and a terminal acoustic test signal are transferred across these signal path switches and the general purpose USB port. The mobile phone terminal can mount the external connection port for the acoustic evaluation test more efficiently.

Mobile phone terminal, and peripheral unit for acoustic test of mobile phone terminal

An echo canceller has a high-pass filtera speakeran A/D converterand an echo canceller. The high-pass filtersuppresses a low-frequency component in a received signal. The speakeroutputs an acoustic sound of a low-frequency suppressed received signal passed through the high-pass filterThe A/D converterconverts an acoustic echo from a microphoneinto a transmission signal of a digital form. The echo cancellergenerates a pseudo echo signal based on the low-frequency suppressed received signal passed through the high-pass filterand eliminates the acoustic echo to be inputted to the microphonefrom the speakerby subtracting the pseudo echo signal from the digital signal converted by the A/D converter

Echo processing apparatus

A speech decoding unit estimates coding parameters of a speech pause by carrying out smoothing algorithm of the coding parameters by using a coding parameter x constituting far-end talker background noise information extracted by a parameter extracting circuit , and a coding parameter x used for synthesizing the previous background noise.

Speech decoding unit and speech decoding method

A first pseudo-echo signal approximating an echo of a reception signal superposed on a transmission signal is produced from a delayed reception signal corresponding to a past time according to filer coefficients, and a first residual signal is produced from the first pseudo-echo signal and a delayed transmission signal. Also, the filer coefficients are renewed to reduce the first residual signal. Therefore, when a single-talk state based on a far-end caller's voice of the reception signal is changed to a double-talk state by adding a near-end caller' voice to the transmission signal, the first filter coefficients deteriorate. Also, a second pseudo-echo signal is produced from a reception signal currently received according to the first filter coefficients renewed. Because a time difference exists between the signals corresponding to the renewal of the first filter coefficients and the production of the second pseudo-echo signal, a deterioration factor is indicated in a delayed second residual signal produced from the second pseudo-echo signal, so that the double-talk state can be detected according to a difference between the first residual signal and the delayed second residual signal.

Double-talk detecting apparatus, echo canceller using the double-talk detecting apparatus and echo suppressor using the double-talk detecting apparatus

In a variable rate speech coding method for a CELP speech coding system, an adaptive sound source vector and a first noise source vector are selected from a sound source code book and a noise source code book so that a first synthesized speech signal is obtained which has a minimum distortion relative to an input speech signal. A virtual reference speech signal is generated using a sound source signal which is produced using the adaptive sound source vector. A second noise source vector corresponding to the adaptive sound source vector is selected so that a second synthesized speech signal is obtained which has a minimum distortion relative to the virtual reference speech signal. The sending of a noise source code book index corresponding to the first noise source vector is suspended according to the quality of the second synthesized speech signal.

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Tokyo, Japan

Bunkei Matsuoka