Disclosed is a signal processing apparatus including a surrounding sound signal acquisition unit, a NC (Noise Canceling) signal generation part, a cooped-up feeling elimination signal generation part, and an addition part. The surrounding sound signal acquisition unit is configured to collect a surrounding sound to generate a surrounding sound signal. The NC signal generation part is configured to generate a noise canceling signal from the surrounding sound signal. The cooped-up feeling elimination signal generation part is configured to generate a cooped-up feeling elimination signal from the surrounding sound signal. The addition part is configured to add together the generated noise canceling signal and the cooped-up feeling elimination signal at a prescribed ratio.

Audio systems, methods, and processor instructions are provided that detect voice activity of a user and provide an output voice signal. The systems, methods, and instructions receive a plurality of microphone signals and combine the plurality of microphone signals according to a first combination and a second combination. The first combination produces a primary signal having enhanced response in the direction of the user's mouth, and the second combination produces a reference signal having reduced response in the direction of the user's mouth. The primary signal and the reference signal are added and subtracted to produce a voice-enhanced signal and a voice-reduced signal, respectively. The voice-enhanced signal and the voice-reduced signal are compares and an output voice signal is provided based upon the comparison.

An image capture device, having: a housing, a lens snout, a front microphone, a top microphone, and an audio processor. The housing has a top and front housing surface. The lens snout protrudes from the front housing surface. The front microphone mounted within or on the front housing surface and below the lens snout. The top microphone mounted within or on a top housing surface in a position biased toward the front housing surface. The audio processor comprises a memory that is configured to store instructions that when executed cause the audio processor to generate an output audio signal. The top microphone is located at a position to receive direct freestream air flow when the housing is positioned in a pitched forward orientation at a pitched forward angle relative to a vertical axis. The front microphone receives turbulent air flow from the lens snout when the housing is positioned in the pitched forward orientation.

The disclosure relates to a method and system for adjusting audio signals by a digital signal processor (DSP). The method comprises receiving an audio signal, adjusting the audio signal and transferring the adjusted audio signal to be output by one or more speakers. The method further comprises receiving sensor data indicative of an air environment around the one or more speakers and adjusting the audio signal based on the sensor data. The method and system may be employed in a vehicle.

The specification and drawings present a use of multiple microphones for increasing acoustic sensing capabilities by processing acoustic signals from the multiple microphones in outdoor luminaire mounted surveillance/sensor systems. For example, various embodiments presented herein describe signal processing means to utilize stereo/multiple microphones in a luminaire (such as an outdoor roadway luminaire) to provide enhanced information regarding the surroundings of the luminaire. The multiple microphone luminaire sensor processing system can provide a more environmentally robust and sensitive approach which can be, for example, resistant to environmental noise such as a wind noise, as well as capable of isolating specific sounds from the surroundings, e.g., in specific directions.

Presented herein are techniques for dynamically setting, in real-time, a ratio of acoustical stimulation signals to electrical stimulation signals delivered by a hearing prosthesis. The ratio of the acoustical stimulation signals to the electrical stimulation signals is set based on one or more characteristics or attributes of the input sound signals that are received and processed by the hearing prosthesis in order to generate the acoustical and electrical stimulation signals.

A sound processing apparatus includes a first microphone which acquires environmental sound, a second microphone which acquires sound of a noise source, and a CPU which causes the sound processing apparatus to function as a noise detection unit configured to generate a noise signal of the noise source according to a sound signal from the second microphone, the noise detection unit reducing sound other than noise of the noise source from the sound signal from the second microphone and generating the noise signal, and a noise reducing unit configured to reduce the noise of the noise included in a sound signal from the first microphone using the sound signal from the noise detection unit.

The invention provides a loudspeaker nonlinear compensation method. The method includes steps: obtaining system parameter of the loudspeaker, the No. i time-domain excitation voltage signal and the No. i state vector of the loudspeaker; compensating the No. i time-domain excitation voltage signal according to the system parameter and the No. i state vector and obtain i compensation voltage signal; obtaining the No. i+1 state vector according to the calculation of the system parameter and the No. i compensation voltage signal; outputting the No. i compensation voltage signal and record the quantity of the compensation voltage signal; judging whether the quantity of the compensation voltage signal is equal to the preset number value.

A system for masking audio signals includes a microphone for generating an ambient audio signal representing ambient noise, a speaker for rendering masking audio, and a processor in communication with the microphone and the speaker. The processor performs spectral analysis on the ambient audio signal from the microphone to determine a spectral envelope of the ambient noise, adjusts a frequency response of an optimizing filter based on the spectral envelope, applies the optimizing filter to a baseline masking waveform, producing an output waveform with relative spectral distribution matching the ambient noise, and provides the output waveform to the speaker.

An improved implantable auditory stimulation system includes two or more implanted microphones for transcutaneous detection of acoustic signals. Each of the implanted microphones provides an output signal. The microphone output signals may be combinatively utilized by an implanted processor to generate a signal for driving an implanted auditory stimulation device. The implanted microphones may be located at offset subcutaneous locations and/or may be provided with different design sensitivities, wherein combinative processing of the microphone output signals may yield an improved drive signal. In one embodiment, the microphone signal may be processed for beamforming and/or directionality purposes.