摘要:
There is provided a signal processing apparatus, for suppressing a noise, which includes a first calculator to obtain a phase difference between two spectrum signals in a frequency domain transformed from sound signals received by at least two microphones to estimate a sound source by the phase difference, a second calculator to obtain a value representing a target signal likelihood and to determine a sound suppressing phase difference range at each frequency, in which a sound signal is suppressed, on the basis of the target signal likelihood, and a filter. The filter generate a synchronized spectrum signal by synchronizing each frequency component of one of the two spectrum signals to each frequency component of the other of the two spectrum signals for each frequency when the phase difference is within the sound suppressing phase difference range and to generate a filtered spectrum signal.
摘要:
A noise suppressing device receives sound signals through a plurality of sound-receiving units and suppresses noise components included in the input sound signals. The noise suppressing device includes a detecting unit which detects a usage pattern of the noise suppressing device from a plurality of usage patterns in which positional relationships of the plurality of sound-receiving units and/or positional relationships between the plurality of sound-receiving units and a target sound source are different from each other, a converting unit which converts using environment information used in a noise suppressing process to each of the sound signals inputted by the plurality of sound-receiving units into using environment information in accordance with a usage pattern detected by the detecting unit and a suppressing unit which performs the noise suppressing process using the using environment information converted by the converting unit to the sound signals.
摘要:
A sound signal correcting apparatus converts an acquired sound signal into a phase spectrum and an amplitude spectrum by an FFT process, compares the amplitude spectrum of the obtained sound signal with a noise model so that a correction coefficient used for correcting the amplitude spectrum of the sound signal is derived, smoothes waveform of the amplitude spectrum of the sound signal using the derived correction coefficient, and converts the sound signal into a sound signal where the amplitude spectrum is corrected by performing an inverse FFT process on the phase spectrum and the smoothed amplitude spectrum.
摘要:
A transfer function estimating device for estimating a transfer function of a sound, includes: a sound receiving module receiving a sound from a given sound source and converting the sound into a tone signal; a storage module storing first transfer functions of the sound propagating from the given sound source to the sound receiving module and transformation coefficients for converting the first transfer functions into given second transfer functions so as to associate with each other; a reference tone signal acquiring module acquiring a reference tone signal of the sound source; an acquiring module acquiring a transfer function of the sound received by the sound receiving module on the basis of the tone signal and the reference tone signal; a specifying module acquiring a cross-correlation value between the transfer function acquired by the acquiring module and each of the first transfer functions stored in the storage module.
摘要:
A sound processing apparatus is provided for estimating the power of background noise using a directional sound receiving technology using a plurality of sound receiving units, computing a gain control value on the basis of the estimated power of background noise and a predetermined power target value, and outputting the gain control value, so that a delay time of starting gain control can be reduced, and a slow response of a speech recognition application program or degradation of the speech quality of a voice communication program can be prevented.
摘要:
The present invention provides a sound signal processing function comprising a plurality of kinds of sound signal processing with the same arrangement of microphones that does not require replacement of the microphones or the sound signal processing part regardless of the application or the sound signal processing function.The present invention uses an apparatus having a signal processing function such as a personal computer as the platform. An array section includes a plurality of microphones arranged in the X and Y axis directions. A received sound signal from each direction is subjected to a delay process by a delay unit, a subtraction process by subtracters 121 and 122, so as to obtain a received sound signal with a unidirectional pattern to the direction of the front of the apparatus and a received sound signal with a bidirectional pattern to the directions orthogonal thereto. In the case where the sound source is not in the direction of the front, a correction process to direct the sound source to the front is performed by a delay unit, a subtracter and adjustment of the gain amount. The directional sound signal calculating part, the sound source direction detecting part, and the noise suppressing part have a logic necessary to implement various functions using the uni/bidirectivity pattern signal as the input.
摘要:
A microphone array apparatus includes a microphone array including microphones, one of the microphones being a reference microphone, filters receiving output signals of the microphones, and a filter coefficient calculator which receives the output signals of the microphones, a noise and a residual signal obtained by subtracting filtered output signals of the microphones other than the reference microphone from a filtered output signal of the reference microphone and which obtain filter coefficients of the filters in accordance with an evaluation function based on the residual signal.
摘要:
The present invention provides a microphone array including a small number of real microphone that can realize the same characteristics as a microphone array including a large number of real microphones. The microphone array of the present invention includes a plurality of real microphones, at least one virtual microphone, and an estimator for estimating a sound signal to be received by the virtual microphone based on the sound signals received by the real microphones.
摘要:
Each of processor units 31-1-31-n comprises a CPU, memory, and interface function. A high-speed communications network 32 interconnects the processor units for inter-processor communications. Each of input/output units 33-1-33-n stores multimedia data and transmits the multimedia data through the network under the control of the corresponding processor unit. A processor control unit 34 controls the process of each of the processor units 31-1-31-n according to the process request received through the network. A switch 35 connects each of the input/output units 33-1-33-n to the network and exchanges data through the network. A communications line 36 transfers an instruction, etc. from the processor control unit 34 to each of the processor units 31-1-31-n.
摘要:
A sound processing apparatus includes a first calculator that calculates first power based on a first signal received by a first microphone that is among the first microphone and a second microphone; a second calculator that calculates second power based on a second signal received by the second microphone; a gain calculator that calculates a gain on the basis of the ratio of the first power to the second power; and a multiplier that processes the second signal using the gain calculated by the gain calculator.