A system configured to perform microphone occlusion event detection. When a device detects a microphone occlusion event, the device will modify audio processing performed prior to speech processing, such as by disabling spatial processing and only processing audio data from a single microphone. The device detects the microphone occlusion event by determining inter-level difference (ILD) values between two microphone signals and using the ILD values as input features to a classifier. For example, when a far-end reference signal is inactive, the classifier may process a first ILD value within a high frequency band. However, when the far-end reference signal is active, the classifier may process the first ILD value and a second ILD value within a low frequency band.

Embodiments include a processing device communicatively coupled to a plurality of audio devices comprising at least one microphone and at least one speaker, and to a digital signal processing (DSP) component having a plurality of audio input channels for receiving audio signals captured by the at least one microphone, the processing device being configured to identify one or more of the audio devices based on a unique identifier associated with each of said one or more audio devices; obtain device information from each identified audio device; and adjust one or more settings of the DSP component based on the device information. A computer-implemented method of automatically configuring an audio conferencing system, comprising a digital signal processing (DSP) component and a plurality of audio devices including at least one speaker and at least one microphone, is also provided.

Aspects of a multi-orientation playback device including at least one microphone array are discussed. A method may include determining an orientation of the playback device which includes at least one microphone array and determining at least one microphone training response for the playback device from a plurality of microphone training responses based on the orientation of the playback device. The at least one microphone array can detect a sound input, and the location information of a source of the sound input can be determined based on the at least one microphone training response and the detected sound input. Based on the location information of the source, the directional focus of the at least one microphone array can be adjusted, and the sound input can be captured based on the adjusted directional focus.

A wireless microphone system comprises system equipment (for example, rack-mounted equipment including receivers/transceivers, distribution amplifier), one or more transmission line accessories, and a transmission line network connecting the accessories with the system equipment. The transmission line accessory compensates for downlink RF losses on transmission lines between accessories and between an accessory and system equipment. Compensation parameters for the transmission line accessory is determined by the system equipment by generating an uplink RF test signal by an RF source at the system equipment. The RF source may be varied over a plurality of frequencies to determine the compensation parameters over the plurality of frequencies. The system equipment subsequently instructs the transmission line accessory to configure an adjustable RF gain circuit (and also possibly a compensation filter) accordingly. The wireless microphone system may also discover accessories on the transmission line network to facilitate installation and maintenance.

Audio signals from microphones of a mobile device are received. Each audio signal is generated by a respective microphone of the microphones. First microphones are selected from among the microphones to generate a front audio signal. Second microphones are selected from among the microphones to generate a back audio signal. A first audio signal portion, which is determined based at least in part on the back audio signal, is removed from the front audio signal to generate a modified front audio signal. A second audio signal portion is removed from the modified front audio signal to generate a left-front audio signal. A third audio signal portion is removed from the modified front audio signal to generate a right-front audio signal.

An audio capture device selects between multiple microphones to generate an output audio signal depending on detected conditions. The audio capture device determines whether one or more microphones are wet or dry and selects one or more audio signals from the one or more microphones depending on their respective conditions. The audio capture device generates a mono audio output signal or a stereo output signal depending on the respective conditions of the one or more microphones.

A test device for testing a microphone has at least one test loudspeaker for generating at least one test tone into at least one test cavity. The test device has a compartment for accommodating the microphone to be tested in acoustic communication with the test cavity. The test device has at least one reference microphone for ascertaining a reference signal of the test tone emitted from the test loudspeaker. The test device has a reference cavity separated from the test cavity and acoustically coupled with the reference microphone and the test cavity. The test loudspeaker is arranged between the reference microphone and the test loudspeaker.

Augmented reality visual display of microphone pick-up patterns are disclosed. An example method includes capturing, via a camera of a computing device, an image of a microphone, and displaying the image on a display of the computing device. The method also includes determining, by the computing device, a location and orientation of the microphone relative to the camera, determining one or more parameters of a pick-up pattern of the microphone, determining a visual representation of the pick-up pattern based on the one or more parameters, and displaying the visual representation of the pick-up pattern overlaid on the image of the microphone.

Audio device and method for operating an audio device is disclosed, the audio device comprising an interface, memory, and a processor, wherein the processor is configured to: obtain a first microphone input signal and a second microphone input signal; process the first microphone input signal and the second microphone input signal for provision of an output signal; and output the output signal. To obtain the first microphone input signal and the second microphone input signal comprises to: obtain a first microphone signal and a second microphone signal; determine a gain compensation scheme based on the first microphone signal and the second microphone signal; and compensate a gain of one or both of the first microphone signal and the second microphone signal in accordance with the gain compensation scheme for provision of the first microphone input signal and the second microphone input signal; wherein to determine the gain compensation scheme comprises to: apply a plurality of test compensation schemes to the first microphone signal and the second microphone signal; determine a performance parameter for each of the test compensation schemes; and select the gain compensation scheme based on the performance parameters.

Provided are a voice call method, an apparatus, a terminal, and a storage medium, relating to the technical field of the terminal. The method includes: obtaining, in response to a voice call instruction, a display screen state of the flexible display screen, the display screen state comprising at least one of a roll-up state and a spread-out state; controlling, in response to the flexible display screen being in the roll-up state, the flexible display screen to spread out, wherein the first housing and the second housing move relative to each other while spreading out the flexible display screen, and wherein a distance between the sound receiving hole and a sound source after the spreading out is smaller than a distance between the sound receiving hole and the sound source before the spreading out; and collecting call voice through the radio microphone.