A terminal control method, a terminal and a non-transitory computer-readable storage medium are provided. The terminal control method includes: receiving, by a microphone, a detection audio signal emitted from a speaker and having a frequency within a pre-set detection frequency range; acquiring actual audio parameters of the detection audio signal when being received by the microphone, and original audio parameters of the detection audio signal when being emitted from the speaker; determining a relative state between the microphone and the speaker according to the actual audio parameters and the original audio parameters; determining a terminal control operation to be performed, according to the relative state and a pre-set correspondence between relative states and terminal control operations; and performing the determined terminal control operation on a terminal where the microphone is located.

A plurality of transducers positioned at respective specified locations in a vehicle are actuated to generate a plurality of respective tones. A plurality of respective time differences are determined between times that each respective tone is generated by the respective transducer and the tone is detected by a portable device. A location of the portable device is determined based at least in part on the time differences.

An electronic device is provided. The electronic device includes a first housing structure including a first microphone, a second housing structure including a second microphone and foldably coupled to the first housing structure, a sensor module disposed in the first housing structure or the second housing structure, and a processor in the first housing structure or the second housing structure, operationally connected with the first microphone, the second microphone, and the sensor module, the processor configured to receive a folding event related to the first housing structure and the second housing structure through the sensor module while a specific voice recording mode is performed, identify folding state information in response to receiving the folding event, identify a recording function configuration related to the first microphone and the second microphone, based on the folding state information and the specific voice recording mode, and perform recording by applying the recording function configuration.

A distributed microphones signal server includes a network interface circuit that communicates with mobile terminals and a plurality of microphones that are geographically distributed. The server receives an audio request message containing a location of the mobile terminal, and selects a microphone from among the plurality of microphones responsive to the location of the mobile terminal. The server obtains an audio stream from the selected microphone, and communicates the audio stream to the mobile terminal. Related methods by a distributed microphones signal server are disclosed. Related computer program products and methods by a mobile terminal are disclosed.

A device may process, after obtaining an image that includes image data concerning a product, first audio data obtained concerning a first utterance of a user of a user device, and first video data obtained concerning a first eye gaze direction of the user, to determine a first reaction of the user to the image. The device may process, after causing display of the image and an overlay superimposed on the image by the user device, second audio data obtained concerning a second utterance of the user, and second video data obtained concerning a second eye gaze direction of the user, to determine a second reaction of the user to the image and the overlay. The device may retrain a product recommendation model using the first reaction and the second reaction, and may generate and cause display of a product recommendation based on the product recommendation model.

Even when a speech is small, the speech is allowed to be detected and an echo is allowed to be appropriately suppressed. Whenever a sample point of a reception signal transmitted through a receiving signal path that transmits a signal to a speaker is acquired, an optimal mask is sequentially generated or selected from base masks as one or a plurality of masks generated based on a learning signal based on a reception signal acquired within a predetermined period before a time point at which the sample point was acquired. Whenever the optimal mask is selected, whether a double-talk state is present is sequentially detected based on a result of comparing an input signal with the optimal mask. When detecting that a speech is not input to a microphone and the reception signal includes a speech, a process of suppressing an echo is sequentially performed on the input signal.

The embodiments of the present application provide an earphone abnormality processing method, an earphone, a system, and a storage medium. In the embodiments of the present application, an abnormal state self-detection function is added to an earphone of an existing dual-microphone, and when the earphone is in a self-detection state, acquiring a sound signal picked up from a specified sound source by a primary microphone and a secondary microphone; and determining the type of abnormal state of the earphone according to a frequency response curve of the sound signal picked up by the primary microphone and the secondary microphone, and further performing abnormality processing on the earphone by using a processing manner adapted to the abnormal state of the earphone, thereby solving the difficulty in the prior art of being unable to process the sound pickup abnormality of an earphone, and improving the usage performance of an earphone and also facilitating the prolonging of the service life of the earphone.

The overall performance of an ANC system may be improved by configuring the ANC system to perform adaption in the frequency domain. The ANC systems may be configured to update an algorithm of an adaptive filter based, at least in part, on the first input signal, the second input signal, and a feedback signal that is based on an output of the adaptive filter. Updating may include changing parameters of the algorithm based on a SDR based, at least in part, on the first input signal. Updating may also include normalizing a step size and processing at least full band information for the input signal in a frequency domain to generate coefficient values for the algorithm. Updating may also include applying a frequency domain magnitude constraint on adaptive filter coefficients.

An apparatus for sound detection, sound localization and beam forming comprises a display and a plurality of microphone stacks, wherein the display surrounds each microphone stack in lateral directions. The apparatus further comprises a plurality of elastic connectors, wherein each elastic connector surrounds one respective microphone stack in lateral direction and mechanically connects the respective microphone stack with the display. Each microphone stack further comprises a microelectromechanical transducer array, the transducer array comprising a plurality of membranes, in particular nano-membranes, and corresponding integrated back-volumes, the back-volumes being arranged under the membranes. An optical reading device is configured to separately detect the displacement of each membrane.

The present disclosure generally relates to managing phone calls. In some embodiments, an electronic device with a touch screen displays a phone call user interface which includes informational items associated with phone calls, such as an add call affordance. In some embodiments, when the electronic device receives a request to initiate another phone and displays an additional user interface that includes a merge call affordance that replaces display of the add call affordance.