Described embodiments generally relate to a signal processing device for on ear detection for an earbud. The device comprises a first microphone input for receiving a microphone signal from a first microphone, the first microphone being configured to be positioned within an ear of a user when the earbud is being worn; a second microphone input for receiving a microphone signal from a second microphone, the second microphone being configured to be positioned outside the ear of the user when the earbud is being worn; a signal generator configured to generate a signal for acoustic playback from a speaker configured to be positioned within the earbud; and a processor. The processor is configured to receive at least one first microphone signal from each of the first microphone input and the second microphone input, and compare the first microphone signals to determine the on ear status of the earbud; determine that the on ear status of the earbud cannot be sufficiently determined, generate a signal for acoustic playback from the speaker, receive a second microphone signal from the first microphone input, and compare the second microphone signal to the generated signal to determine the on ear status of the earbud.

The present technology relates to a signal processing apparatus and method that are capable of reproducing sound at an optional listening position with a high sense of reality. The signal processing apparatus includes a rendering unit that generates reproduction data of sound at an optional listening position in a target space on the basis of recording signals of microphones attached to a plurality of moving bodies in the target space. The present technology can be applied to a reproduction apparatus.

Embodiments described herein provide a combined multi-source time difference of arrival (TDOA) tracking and voice activity detection (VAD) mechanism that is applicable for generic array geometries, e.g., a microphone array that lies on a plane. The combined multi-source TDOA tracking and VAD mechanism scans the azimuth and elevation angles of the microphone array in microphone pairs, based on which a planar locus of physically admissible TDOAs can be formed in the multi-dimensional TDOA space of multiple microphone pairs. In this way, the multi-dimensional TDOA tracking reduces the number of calculations that was usually involved in traditional TDOA by performing the TDOA search for each dimension separately.

Systems and methods for optimizing voice detection via a network microphone device are disclosed herein. In one example, individual microphones of a network microphone device detect sound. The sound data is captured in a first buffer and analyzed to detect a trigger event. Metadata associated with the sound data is captured in a second buffer and provided to at least one network device to determine at least one characteristic of the detected sound based on the metadata. The network device provides a response that includes an instruction, based on the determined characteristic, to modify at least one performance parameter of the NMD. The NMD then modifies the at least one performance parameter based on the instruction.

The phase responses of a first and a second microphone are adjusted with first and second filters that filter their microphone signals. The first filter corresponds to a first contribution to a phase shift between the microphones and includes a first adaptation parameter. The second filter corresponds to a second contribution to the phase shift and has a second adaptation parameter. A global filter, which is formed with the first and second filters, represents the first and second contributions to the phase shift and includes the first and second adaptation parameters. The global filter determines a first value for the first adaptation parameter and a second value for the second adaptation parameter via a multidimensional optimization. The phase responses are adjusted by applying the first filter and the second filter to at least one of the microphone signals with the adaptation parameters set to the first and second values, respectively.

An implantable medical device is configured to detect signals with first and second implantable sensors configured to be implanted in a recipient. The implantable medical device is configured to adaptively equalize a response of the first implantable sensor to a response of a similar external sensor, wherein adaption control of the equalization is based on a coherence between the signals detected by first implantable sensor and the signals detected by the external microphone indicating the presence of acoustic signals. In addition, the implantable medical device is configured to adaptively filter vibration signals, including body noises, from the implantable sound signals, wherein adaption control of the filter is based on a coherence between the signals detected by first implantable sensor and the signals detected by the second implantable sensor indicating the presence of vibration.

Focusing sound signals in a shared 3D space uses an array of physical microphones, preferably disposed evenly across a room to provide even sound coverage throughout the room. At least one processor coupled to the physical microphones does not form beams, but instead preferably forms 1000's of virtual microphone bubbles within the room. By determining the processing gains of the sound signals sourced at each of the bubbles, the location(s) of the sound source(s) in the room can be determined. This system provides not only sound improvement by focusing on the sound source(s), but with the advantage that a desired sound source can be focused on more effectively (rather than steered to) while un-focusing undesired sound sources (like reverb and noise) instead of rejecting out of beam signals. This provides a full three dimensional location and a more natural presentation of each sound within the room.

Systems and methods for microphone degradation detection and compensation are disclosed. For example, microphones of an electronic device may capture audio and generate corresponding audio data, such as during a period of time where only ambient noise is present. Sound intensity level value differences between audio data from the various microphones may be determined and when one or more of the sound intensity level value differences satisfies a threshold amount, the microphone associated with the variant sound intensity level value may be determined to be degraded. The sound intensity level value difference may be compensated for, such as by utilizing sound boosting techniques and/or modifying parameters of a beamforming component.

Described embodiments generally relate to a signal processing device for on ear detection for an earbud. The device comprises a first microphone input for receiving a microphone signal from a first microphone, the first microphone being configured to be positioned within an ear of a user when the earbud is being worn; a second microphone input for receiving a microphone signal from a second microphone, the second microphone being configured to be positioned outside the ear of the user when the earbud is being worn; a signal generator configured to generate a signal for acoustic playback from a speaker configured to be positioned within the earbud; and a processor. The processor is configured to receive at least one first microphone signal from each of the first microphone input and the second microphone input, and compare the first microphone signals to determine the on ear status of the earbud; determine that the on ear status of the earbud cannot be sufficiently determined, generate a signal for acoustic playback from the speaker, receive a second microphone signal from the first microphone input, and compare the second microphone signal to the generated signal to determine the on ear status of the earbud.

A sound pickup device incudes a plurality of microphone elements, a sensitivity correcting unit that corrects a difference in sensitivity among the microphone elements by multiplying an output signal of each of the microphone elements by a gain, a target sound detecting unit that detects a voice of a speaker as a target sound, a sensitivity correction control unit that controls the gain based on a result of detecting the target sound, and a directivity synthesizing unit that picks up the target sound in a boosted manner using the output signals from the microphone elements of which difference in sensitivity is corrected, the target sound coming along a predetermined direction. The sensitivity correction control unit updates the gain based on the output signals from the microphone elements if the voice of the speaker is detected and does not update the gain if the voice of the speaker is not detected.