The present technology relates to a signal processing apparatus and method, and a program 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.

Techniques are described for reducing sensitivity to non-acoustic stimuli. In some embodiments, differential beamforming is applied to microphone signals generated based on responses of microphones to an acoustic stimulus and a non-acoustic stimulus. Compensated signals can be generated based on the microphone signals such that the compensated signals are in phase with respect to the acoustic stimulus. The non-acoustic stimulus is detectable by comparing a first signal to a second signal to determine that one signal has a greater instantaneous magnitude. The first signal can be a beamformed signal or signal derived therefrom, and the second signal can be an average of the compensated signals or signal derived therefrom. An output audio signal can be generated by switching or cross fading between the beamformed signal and a noise-reduced signal such that a contribution of the noise-reduced signal is increased and a contribution of the beamformed signal is decreased.

Techniques are described for detecting and correcting mismatched microphone sensitivities in a microphone array without knowledge of the acoustic excitation source(s). Mismatch is detectable based on time and/or frequency domain analysis of each microphone's long term exposure to a real-world sound field that includes an acoustic source and a non-acoustic source, and is corrected by adjusting the amount of amplification applied to at least one microphone signal. In the time domain, sensitivity matching can be performed by using an average of all microphone signals as a reference signal. In some embodiments, the reference signal is the root mean square of the average. Alternatively, a single microphone can be selected as a reference. In some embodiments, sensitivity mismatch is detected and corrected at specific frequencies based on comparing frequency components of amplified microphone signals. Sensitivity matching can be repeated to ensure that the microphones remain sensitivity-matched over time.

An electronic device or method for adjusting a gain on a voice operated control system can include one or more processors and a memory having computer instructions. The instructions, when executed by the one or more processors causes the one or more processors to perform the operations of receiving a first microphone signal, receiving a second microphone signal, updating a slow time weighted ratio of the filtered first and second signals, and updating a fast time weighted ratio of the filtered first and second signals. The one or more processors can further perform the operations of calculating an absolute difference between the fast time weighted ratio and the slow time weighted ratio, comparing the absolute difference with a threshold, and increasing the gain when the absolute difference is greater than the threshold. Other embodiments are disclosed.

Certain aspects of the present disclosure provide an apparatus. The apparatus comprises a support structure comprising at least one microphone sensor, and a first material layer disposed adjacent to the support structure, wherein a first layer of air is formed between the first material layer and the support structure, the first layer of air being adjacent to the microphone sensor. In certain aspects, multiple material layers may be used, each of the material layers forming a layer of air. For instance, the apparatus may also include a second material layer disposed adjacent to the first material layer, wherein a second layer of air is formed between the first material layer and the second material layer.

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.

An electronic device or method for adjusting a gain on a voice operated control system can include one or more processors and a memory having computer instructions. The instructions, when executed by the one or more processors causes the one or more processors to perform the operations of receiving a first microphone signal, receiving a second microphone signal, updating a slow time weighted ratio of the filtered first and second signals, and updating a fast time weighted ratio of the filtered first and second signals. The one or more processors can further perform the operations of calculating an absolute difference between the fast time weighted ratio and the slow time weighted ratio, comparing the absolute difference with a threshold, and increasing the gain when the absolute difference is greater than the threshold. Other embodiments are disclosed.

A hearing device, e.g. a hearing aid, comprises first and second separate, interconnectable parts comprising first and second input transducers, respectively, for providing first and second electric input signals, respectively, representative of sound in an environment of the user, and a beamformer filtering unit configured to provide a spatially filtered signal based thereon, and a memory comprising a previously determined own voice transfer function corresponding to a target sound source located at said user's mouth. The hearing device is configured to determine an updated own voice transfer function according to activation of a predefined trigger, when the user's own voice is present, and to store an updated own voice transfer function in said memory. The hearing device further comprises at least one combination unit configured to apply a first multiplication factor to at least one of the first and second electric input signals, and a control unit configured to determine the first multiplication factor so as to decrease, e.g. minimize a difference measure representative of a difference between the previously determined own voice transfer function and the updated own voice transfer function.

An electronic device according to an embodiment may include: a first sound input device configured to obtain external sound and produce a first signal and a processor operatively connected to the first sound input device. The processor may be configured to: receive the first signal from the first sound input device; produce a first high-frequency signal by passing the first signal through a high-pass filter to; determine a first energy value of the first high-frequency signal; determine a second energy value of the first signal; compare a product of the second energy value of the first signal and the first energy value of the first high-frequency signal with a first threshold value to produce a first result; and determine whether the first sound input device is blocked based on the first result. In addition, various other embodiments may be provided.

A computer implemented method includes receiving audio signals representative of speech via multiple audio channels transmitted from corresponding multiple distributed devices, designating one of the audio channels as a reference channel, and for each of the remaining audio channels, determine a difference in time from the reference channel, and correcting each remaining audio channel by compensating for the corresponding difference in time from the reference channel.