A voice analyzer analyzes whether a voice signal input into a voice input unit includes a specific characteristic component. A voice recognizer recognizes a voice represented by the voice signal input into the voice input unit. A response instruction unit instructs a response to a response operation unit that operates in response to the voice recognized by the voice recognizer. A controller controls the voice recognizer not to execute voice recognition processing by the voice recognizer or controls the response instruction unit not to instruct the response operation unit about an instruction content by the voice recognized by the voice recognizer, when the voice analyzer analyzes that the voice signal includes the specific characteristic component.

A voice analyzer analyzes whether a voice signal input into a voice input unit includes a specific characteristic component. A voice recognizer recognizes a voice represented by the voice signal input into the voice input unit. A response instruction unit instructs a response to a response operation unit that operates in response to the voice recognized by the voice recognizer. A controller controls the voice recognizer not to execute voice recognition processing by the voice recognizer or controls the response instruction unit not to instruct the response operation unit about an instruction content by the voice recognized by the voice recognizer, when the voice analyzer analyzes that the voice signal includes the specific characteristic component.

A hearing device adapted for being worn at or in an ear of a user, comprises a) an input unit comprising at last two input transducers each for converting sound around said hearing device to an electric input signal representing said sound, thereby providing at least two electric input signals; b) a beamformer filter comprising a minimum processing beamformer defined by optimized beamformer weights, the beamformer filter being configured to provide a filtered signal in dependence of said at least two electric input signals and said optimized beamformer weights; c) a reference signal representing sound around said hearing device; d) a performance criterion for said minimum processing beamformer. The minimum processing beamformer is a beamformer that provides the filtered signal with as little modification as possible in terms of a selected distance measure compared to said reference signal, while still fulfilling said performance criterion. The optimized beamformer weights are adaptively determined in dependence of said at least two electric input signals, said reference signal, said distance measure, and said performance criterion. A method of operating a hearing device is further disclosed. The invention may e.g. be used in hearing aids or headsets.

A data processing method and a related apparatus are disclosed, to improve data recovery effects in various packet loss scenarios. The method includes: determining a packet loss scenario type corresponding to a first data packet when detecting that the first data packet is lost in a data packet transmission process (201); generating a second data packet based on a data packet adjacent to the first data packet if the packet loss scenario type corresponding to the first data packet meets a data packet compensation condition (202); and finally adding the second data packet to a corresponding target location at which the first data packet is located before the first data packet is lost, that is, using the second data packet to compensate for the lost first data packet (203).

A server includes a communication interface, a memory, and a processor. The communication interface communicates with a first terminal device that transmits voice data generated from an input voice and a second terminal device that outputs a voice based on the voice data received from the first terminal device. The memory stores voice recognition results by the first and second terminal devices for the input voice input to the first terminal device and the second terminal device respectively. The processor determines a difference between the input voice input to the first terminal device and the voice output based on the voice data of the input voice received by the second terminal device from the first terminal device based on a comparison between the voice recognition result by the first terminal device and the voice recognition result by the second terminal device.

The disclosed system provides a facility for improving user call quality at a mobile device. The system includes an over-the-top (OTT) client that may be installed on the mobile device for allowing a user to initiate call quality tests. The system performs a call quality test to generate a call quality score or metric from the obtained audio sample via a Call Quality Algorithm. If the call quality score or metric falls below a predetermined threshold, the system may suggest (via the OTT client or via a push notification) that the user of the mobile device switch from the first communication interface (e.g., a radio network such as 4G) to a second communication interface (e.g., Wi-Fi) on the mobile device. The disclosed system tracks the call quality score or metric of multiple mobile devices operating within a telecommunications network, identifies service outages within the network, and notifies impacted mobile devices.

Methods and systems for more efficient analyses of and response to voice commands and queries are provided. The system may be configured to receive one or more of audio files corresponding to a voice query and determine, for each of the audio files, whether the audio file is a first type of audio file capable of being processed based on a characteristic of the audio file or a second type of audio file that cannot, and may require further processing in order to recognize the voice query associated with the audio file. The system may process each of the first type of audio files and respond to the associated voice queries. The system may also determine a priority for each of the second type of audio files for further processing of the second type of audio files.

Methods and systems for more efficient analyses of and response to voice commands and queries are provided. The system may be configured to receive one or more of audio files corresponding to a voice query and determine, for each of the audio files, whether the audio file is a first type of audio file capable of being processed based on a characteristic of the audio file or a second type of audio file that cannot, and may require further processing in order to recognize the voice query associated with the audio file. The system may process each of the first type of audio files and respond to the associated voice queries. The system may also determine a priority for each of the second type of audio files for further processing of the second type of audio files.

An audio analyzer configured to obtain spectral domain representations of two or more input audio signals. Additionally the audio analyzer is configured to obtain directional information associated with spectral bands of the spectral domain representations and to obtain loudness information associated with different directions as an analysis result. Contributions to the loudness information are determined in dependence on the directional information.

Systems and methods for communications are disclosed. The systems and methods can monitor a cloud-based voice over internet protocol (VoIP) calling system to determine an active call. The systems and methods can also analyze the active call to determine an indication of call quality, the analyzing occurring during the active call. Additionally, the systems and methods can compare the indication of call quality to a quality threshold. The compare can occur during the active call to determine when the active call has a poor call quality. The systems and methods can also report the poor call quality based on the comparing the indication of call quality to the quality threshold.