Abstract:
A purpose of the invention is to provide a noise gate that can output an audio signal in which only a stationary noise is removed, without degrading an utterance voice of a speaking person. A sound collection device 1 includes an FFT processing unit 11, the noise gate 12, and an IFFT processing unit 13.The sound collection device 1 transforms a collected audio signal NET into a frequency spectrum NE′N by using the FFT processing unit 11. The noise gate 12 estimates a noise spectrum N′N of a stationary noise based on the frequency spectrum NE′N of the audio signal. The noise gate 12 decreases a signal level (a gain) of the audio signal in a case where a signal level ratio of the frequency spectrum NE′N of the audio signal to the noise spectrum N′N is less than a threshold value, and outputs the audio signal. The sound collection device 1 outputs an audio signal CO′T which is generated in such a manner that the IFFT processing unit 13 inversely transforms a frequency spectrum CO′N after removing the stationary noise N′N.
Abstract:
Provided is a sound emission and collection device capable of estimating the azimuth of a sound source (such as a main utterer) precisely without any processing load. The sound emission and collection device (1) is connected with another sound emission and collection device via a network or the like. The sound emission and collection device (1) receives a sound signal from another sound emission and collection device, as a sound emission signal (FE), and emits the same from a speaker (SP). The sound emission and collection device (1) collects the sound at microphones (MIC1 to MIC3), and produces sound collection beam signals (NE1 to NE3) of different azimuths. The sound emission and collection device down-samples the individual sound collection beam signals (NE1 to NE3), and filters out the echoes of the down-sampled sound collection beam signals (DNE1 to DNE3). The sound emission and collection device selects the sound collection beam signal (DNE1′) of the highest signal level from the echo-filtered sound collection beam signals (DNE1′ to DNE3′). The sound emission and collection device filters out the echoes of a sound collection beam signal (NE1) from the sound collection azimuth (D1) of the sound collection beam signal (DNE1′), and transmits the same to another sound emission and collection device.
Abstract:
A position detecting system is provided, which is capable of effectively preventing erroneous detection of audio to be measured. The position detecting system includes a terminal device that inputs an audio signal from an audio device and a microphone. The audio device sequentially inputs measurement audio signals that have been formed by two or more audio signals of different frequencies to a speaker and receives a notification signal, wherein the report signal indicates that the audio of the measurement audio signal has been collected from the terminal device. The audio device clocks a time t1 and a time t2, namely clocks after the audio of the measurement audio signal is output from the speakers SP1-SP2 until the notification signals of the measurement audio signals are received by the signal receiving unit. The audio device calculates the position of the microphone by using the times t1 and t2. For each frequency component of the measurement audio signal, when an audio signal exceeding a predetermined level is inputted from the microphone, the terminal device detects it as a component of the measurement audio signal and transmits a notification signal upon detection of the measurement audio signal.
Abstract:
An acoustic apparatus without increasing noise etc. even when plural directional microphones collect sounds from a place of the same distances is provided. Sound signals output from the microphone arrays are subjected to phase shift by phase shift circuits 211A to 211H, and the sound signals are combined by an adder 212, The phase shift circuits 211A to 211H performs phase shifts according to installation positions of the microphone arrays. The phase shift circuit 211A makes the shift 0 degree, the phase shift circuit 211B makes the shift 45 degrees, the phase shift circuit 211C makes the shift 90 degrees, and sequentially to the phase shift circuit 211H, the shifts are made according to rotational angles.
Abstract:
To provide an audio conference apparatus and an audio conference system which can smoothly proceed with the audio conference by removing a recursion sound of the conference voice is achieved. An audio conference apparatus 1 outputs ring tones from corresponding channels before a communication control unit 12 outputs audio signals from the unused channels (S1 to S3). Speakers SP1 to SP16 emits the ring tone from predetermined sound source positions corresponding to the respective channels. Microphones MIC1A to MIC16A and microphones MIC1B to MIC16B collect audio signals including a recursion sound of the ring tone. The echo cancel unit 20 generates a pseudo-recursion sound signal on the basis of an input signal, and subtracts the pseudo-recursion sound signal from the collected audio signals. An audio conference system is configured to connect a plurality of the audio conference apparatuses to each other.
Abstract:
A control unit gives acoustic environment instruction data to a picked up sound directionality control unit and an adaptive filter. According to this, the picked up sound directionality control unit generates a picked up sound signal constituted by a predetermined picked up sound directionality. The adaptive filter detects the picked up sound directionality from the acoustic environment instruction data, and reads out the filter parameter corresponding to this picked up sound directionality, from a memory. The adaptive filter sets a delay coefficient and a filter coefficient of an FIR filter, and generates a pseudo echo signal by an impulse response with respect to the received sound signal. Based on an error signal obtained by subtracting the pseudo echo signal from the picked up sound signal by an adder, the adaptive filter sets a more optimum filter parameter, and generates the next pseudo echo signal.
Abstract:
Disclosed herein is a liquid crystal display device formed by laminating at least two first and second liquid crystal panels, the liquid crystal panels being each formed by disposing a liquid crystal layer between two transparent substrates arranged so as to be opposed to each other and two-dimensionally arranging pixels in a form of a matrix on one of the two substrates, and disposing a backlight on a side of the first liquid crystal panel. The liquid crystal display device includes: a first driver configured to drive the first liquid crystal panel on a side of the backlight by n-time speed driving in which one frame period is divided into n fields; and a second driver configured to drive the second liquid crystal panel on a display surface side by normal driving in which one frame period is not divided.
Abstract:
The invention provides a waterproof slide fastener and a manufacturing apparatus thereof, wherein each fastener element has a coupling head, a neck portion, a body portion, and leg portions which are continuously fused and integrated, a terminal end is projected from a gap between upper/lower half portions of the coupling head, upper/lower flanges of a slider make a sliding contact with a surface of each of upper/lower half portions of each leg portion, each half portion of the leg portion extending from the body portion via a crotch portion and guiding the flanges, two or more leg portions are disposed with a predetermined interval in a sliding direction, an entire shoulder width of a shoulder portion is larger than a body width of the body portion if it is provided, thereby intensifying fixing strength between a fastener tape and fastener elements and waterproofness, and allowing a smooth sliding of a slider.
Abstract:
A signal processing unit calculates a first difference in time of arrival of sound from a sound source to a first and to a second microphone comprising a microphone array and calculates a second difference in time of arrival, which is the difference between the first difference in time of arrival and an actual time, of arrival, and determines the position of the sound source based on the sum of the first difference in time of arrival and the second difference in time of arrival.
Abstract:
An image processing apparatus writes a virtual space image obtained by imaging a virtual space in which objects are arranged from a virtual camera to an output area. When a pointer image representing a positional relationship between a referential position and an arrangement position of the object is depicted on the virtual space image stored in the output area, the pointer image to be depicted is changed in correspondence with conditions, such as the height of the virtual camera and the attribute of the object.