Techniques for presence-detection devices to detect movement of a person in an environment by emitting ultrasonic signals using a loudspeaker that is concurrently outputting audible sound. To detect movement by the person, the devices characterize the change in the frequency, or the Doppler shift, of the reflections of the ultrasonic signals off the person caused by the movement of the person. However, when a loudspeaker plays audible sound while emitting the ultrasonic signal, audio signals generated by microphones of the devices include distortions caused by the loudspeaker. These distortions can be interpreted by the presence-detection devices as indicating movement of a person when there is no movement, or as indicating lack of movement when a user is moving. The techniques include processing audio signals to remove distortions to more accurately identify changes in the frequency of the reflections of the ultrasonic signals caused by the movement of the person.

A method for controlling a frequency response having a bell-shaped amplitude characteristic of a filter that is configured to process an audio signal includes preparing a high-frequency parameter related to a high-frequency side characteristic on a high-frequency side of the frequency response, and a low-frequency parameter related to a low-frequency side characteristic on a low-frequency side of the frequency response, independently changing the high-frequency parameter or the low-frequency parameter in accordance with a first change instruction, changing both of the high-frequency parameter and the low-frequency parameter in conjunction with each other in accordance with a second change instruction, and controlling a slope on the high-frequency side and a slope on the low-frequency side of the bell-shaped amplitude characteristic of the frequency response, by using the high-frequency parameter and the low-frequency parameter.

A method for controlling a frequency response having a bell-shaped amplitude characteristic of a filter that is configured to process an audio signal includes preparing a high-frequency parameter related to a high-frequency side characteristic on a high-frequency side of the frequency response, and a low-frequency parameter related to a low-frequency side characteristic on a low-frequency side of the frequency response, independently changing the high-frequency parameter or the low-frequency parameter in accordance with a first change instruction, changing both of the high-frequency parameter and the low-frequency parameter in conjunction with each other in accordance with a second change instruction, and controlling a slope on the high-frequency side and a slope on the low-frequency side of the bell-shaped amplitude characteristic of the frequency response, by using the high-frequency parameter and the low-frequency parameter.

An electronic device is provided. The electronic device includes a microphone configured to receive an audio signal including a voice of a user, a sensor configured to detect a vibration signal generated by the user, at least one processor, and a memory configured to store an instruction executable by the processor. The at least one processor may be configured to determine a noise level included in the audio signal, calculate a verification score based on the noise level, the audio signal, and the vibration signal, and perform speaker verification for the user based on the verification score.

An electronic device is provided. The electronic device includes a microphone configured to receive an audio signal including a voice of a user, a sensor configured to detect a vibration signal generated by the user, at least one processor, and a memory configured to store an instruction executable by the processor. The at least one processor may be configured to determine a noise level included in the audio signal, calculate a verification score based on the noise level, the audio signal, and the vibration signal, and perform speaker verification for the user based on the verification score.

The present invention discloses a method of noise reduction for an intelligent network communication, which includes the following steps: first, receiving a local sound message through a sound receiver of a communication device at the transmitting end. Next, a voice recognizer is used to identify the voice characteristics of the speaker; then, it is determined from a voice database whether there is a corresponding or similar voice characteristic of the speaker recognized by the voice recognizer. Finally, filtering other signals other than the voice characteristic signal of the speaker through a sound filter to obtain the original sound emitted by the speaker.

The present invention discloses a method of noise reduction for an intelligent network communication, which includes the following steps: first, receiving a local sound message through a sound receiver of a communication device at the transmitting end. Next, a voice recognizer is used to identify the voice characteristics of the speaker; then, it is determined from a voice database whether there is a corresponding or similar voice characteristic of the speaker recognized by the voice recognizer. Finally, filtering other signals other than the voice characteristic signal of the speaker through a sound filter to obtain the original sound emitted by the speaker.

A hearing device component (6) comprises a sensor-unit (8) for receiving an audio-signal (AS), a separation device (9) for separating part-signals (PS.sub.i) from the audio-signal (AS), a classification device (10) for classifying the part-signals (PS.sub.i) separated from the audio-signals (AS), and a modulation device (11) for modulating the part-signals (PS.sub.i), wherein the classification device (10) is communicatively coupled to the modulation device (11) and wherein the modulation device (11) is designed to enable a concurrent modulation of different part-signals (PS.sub.i) with different modulation-functions depending on their classification.

Audio beamforming systems and methods that enable more precise control of lobes and nulls of an array microphone are provided. Optimized beamformer coefficients can be generated to result in beamformed signals associated with one or more lobes steered towards one or more desired sound locations and one or more nulls steered towards one or more undesired sound location. The performance of acoustic echo cancellation can be improved and enhanced.

Audio beamforming systems and methods that enable more precise control of lobes and nulls of an array microphone are provided. Optimized beamformer coefficients can be generated to result in beamformed signals associated with one or more lobes steered towards one or more desired sound locations and one or more nulls steered towards one or more undesired sound location. The performance of acoustic echo cancellation can be improved and enhanced.